Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

149 files changed, 12616 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
new file mode 100644
index 0000000000..77da78ba7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant.h"
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant_loop.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+                                   and idxVec, uses state_first as
+                                   input) */
+    int16_t *in,     /* (i) vector to encode */
+    int16_t *weightDenum   /* (i) denominator of synthesis filter */
+                            ) {
+  int16_t *syntOut;
+  size_t quantLen[2];
+
+  /* Stack based */
+  int16_t syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
+  int16_t in_weightedVec[STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t *in_weighted = &in_weightedVec[LPC_FILTERORDER];
+
+  /* Initialize the buffers */
+  WebRtcSpl_MemSetW16(syntOutBuf, 0, LPC_FILTERORDER+STATE_SHORT_LEN_30MS);
+  syntOut = &syntOutBuf[LPC_FILTERORDER];
+  /* Start with zero state */
+  WebRtcSpl_MemSetW16(in_weightedVec, 0, LPC_FILTERORDER);
+
+  /* Perform the quantization loop in two sections of length quantLen[i],
+     where the perceptual weighting filter is updated at the subframe
+     border */
+
+  if (iLBC_encbits->state_first) {
+    quantLen[0]=SUBL;
+    quantLen[1]=iLBCenc_inst->state_short_len-SUBL;
+  } else {
+    quantLen[0]=iLBCenc_inst->state_short_len-SUBL;
+    quantLen[1]=SUBL;
+  }
+
+  /* Calculate the weighted residual, switch perceptual weighting
+     filter at the subframe border */
+  WebRtcSpl_FilterARFastQ12(
+      in, in_weighted,
+      weightDenum, LPC_FILTERORDER+1, quantLen[0]);
+  WebRtcSpl_FilterARFastQ12(
+      &in[quantLen[0]], &in_weighted[quantLen[0]],
+      &weightDenum[LPC_FILTERORDER+1], LPC_FILTERORDER+1, quantLen[1]);
+
+  WebRtcIlbcfix_AbsQuantLoop(
+      syntOut,
+      in_weighted,
+      weightDenum,
+      quantLen,
+      iLBC_encbits->idxVec);
+
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
new file mode 100644
index 0000000000..c72e29cf29
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits* iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+                                   and idxVec, uses state_first as
+                                   input) */
+    int16_t* in,             /* (i) vector to encode */
+    int16_t* weightDenum     /* (i) denominator of synthesis filter */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
new file mode 100644
index 0000000000..cf9266299d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant_loop.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/sort_sq.h"
+
+void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
+                                int16_t *weightDenumIN, size_t *quantLenIN,
+                                int16_t *idxVecIN ) {
+  size_t k1, k2;
+  int16_t index;
+  int32_t toQW32;
+  int32_t toQ32;
+  int16_t tmp16a;
+  int16_t xq;
+
+  int16_t *syntOut   = syntOutIN;
+  int16_t *in_weighted  = in_weightedIN;
+  int16_t *weightDenum  = weightDenumIN;
+  size_t *quantLen  = quantLenIN;
+  int16_t *idxVec   = idxVecIN;
+
+  for(k1=0;k1<2;k1++) {
+    for(k2=0;k2<quantLen[k1];k2++){
+
+      /* Filter to get the predicted value */
+      WebRtcSpl_FilterARFastQ12(
+          syntOut, syntOut,
+          weightDenum, LPC_FILTERORDER+1, 1);
+
+      /* the quantizer */
+      toQW32 = (int32_t)(*in_weighted) - (int32_t)(*syntOut);
+
+      toQ32 = (((int32_t)toQW32)<<2);
+
+      if (toQ32 > 32767) {
+        toQ32 = (int32_t) 32767;
+      } else if (toQ32 < -32768) {
+        toQ32 = (int32_t) -32768;
+      }
+
+      /* Quantize the state */
+      if (toQW32<(-7577)) {
+        /* To prevent negative overflow */
+        index=0;
+      } else if (toQW32>8151) {
+        /* To prevent positive overflow */
+        index=7;
+      } else {
+        /* Find the best quantization index
+           (state_sq3Tbl is in Q13 and toQ is in Q11)
+        */
+        WebRtcIlbcfix_SortSq(&xq, &index,
+                             (int16_t)toQ32,
+                             WebRtcIlbcfix_kStateSq3, 8);
+      }
+
+      /* Store selected index */
+      (*idxVec++) = index;
+
+      /* Compute decoded sample and update of the prediction filter */
+      tmp16a = ((WebRtcIlbcfix_kStateSq3[index] + 2 ) >> 2);
+
+      *syntOut     = (int16_t) (tmp16a + (int32_t)(*in_weighted) - toQW32);
+
+      syntOut++; in_weighted++;
+    }
+    /* Update perceptual weighting filter at subframe border */
+    weightDenum += 11;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
new file mode 100644
index 0000000000..841d73b9fb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuantLoop(int16_t* syntOutIN,
+                                int16_t* in_weightedIN,
+                                int16_t* weightDenumIN,
+                                size_t* quantLenIN,
+                                int16_t* idxVecIN);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
new file mode 100644
index 0000000000..57b5abbe23
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+AudioDecoderIlbcImpl::AudioDecoderIlbcImpl() {
+  WebRtcIlbcfix_DecoderCreate(&dec_state_);
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+AudioDecoderIlbcImpl::~AudioDecoderIlbcImpl() {
+  WebRtcIlbcfix_DecoderFree(dec_state_);
+}
+
+bool AudioDecoderIlbcImpl::HasDecodePlc() const {
+  return true;
+}
+
+int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+size_t AudioDecoderIlbcImpl::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
+}
+
+void AudioDecoderIlbcImpl::Reset() {
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+  size_t bytes_per_frame;
+  int timestamps_per_frame;
+  if (payload.size() >= 950) {
+    RTC_LOG(LS_WARNING)
+        << "AudioDecoderIlbcImpl::ParsePayload: Payload too large";
+    return results;
+  }
+  if (payload.size() % 38 == 0) {
+    // 20 ms frames.
+    bytes_per_frame = 38;
+    timestamps_per_frame = 160;
+  } else if (payload.size() % 50 == 0) {
+    // 30 ms frames.
+    bytes_per_frame = 50;
+    timestamps_per_frame = 240;
+  } else {
+    RTC_LOG(LS_WARNING)
+        << "AudioDecoderIlbcImpl::ParsePayload: Invalid payload";
+    return results;
+  }
+
+  RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame);
+  if (payload.size() == bytes_per_frame) {
+    std::unique_ptr<EncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(this, std::move(payload)));
+    results.emplace_back(timestamp, 0, std::move(frame));
+  } else {
+    size_t byte_offset;
+    uint32_t timestamp_offset;
+    for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
+         byte_offset += bytes_per_frame,
+        timestamp_offset += timestamps_per_frame) {
+      std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
+          this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
+      results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
+    }
+  }
+
+  return results;
+}
+
+int AudioDecoderIlbcImpl::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderIlbcImpl::Channels() const {
+  return 1;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
new file mode 100644
index 0000000000..46ba755148
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+
+typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
+
+namespace webrtc {
+
+class AudioDecoderIlbcImpl final : public AudioDecoder {
+ public:
+  AudioDecoderIlbcImpl();
+  ~AudioDecoderIlbcImpl() override;
+
+  AudioDecoderIlbcImpl(const AudioDecoderIlbcImpl&) = delete;
+  AudioDecoderIlbcImpl& operator=(const AudioDecoderIlbcImpl&) = delete;
+
+  bool HasDecodePlc() const override;
+  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  IlbcDecoderInstance* dec_state_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
new file mode 100644
index 0000000000..9fbf42ceeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+
+#include <algorithm>
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kSampleRateHz = 8000;
+
+int GetIlbcBitrate(int ptime) {
+  switch (ptime) {
+    case 20:
+    case 40:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 30:
+    case 60:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
+}  // namespace
+
+AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
+                                           int payload_type)
+    : frame_size_ms_(config.frame_size_ms),
+      payload_type_(payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      encoder_(nullptr) {
+  RTC_CHECK(config.IsOk());
+  Reset();
+}
+
+AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+}
+
+int AudioEncoderIlbcImpl::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+size_t AudioEncoderIlbcImpl::NumChannels() const {
+  return 1;
+}
+
+size_t AudioEncoderIlbcImpl::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderIlbcImpl::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderIlbcImpl::GetTargetBitrate() const {
+  return GetIlbcBitrate(rtc::dchecked_cast<int>(num_10ms_frames_per_packet_) *
+                        10);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  // Save timestamp if starting a new packet.
+  if (num_10ms_frames_buffered_ == 0)
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  // Buffer input.
+  std::copy(audio.cbegin(), audio.cend(),
+            input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
+
+  // If we don't yet have enough buffered input for a whole packet, we're done
+  // for now.
+  if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+    return EncodedInfo();
+  }
+
+  // Encode buffered input.
+  RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  num_10ms_frames_buffered_ = 0;
+  size_t encoded_bytes = encoded->AppendData(
+      RequiredOutputSizeBytes(), [&](rtc::ArrayView<uint8_t> encoded) {
+        const int r = WebRtcIlbcfix_Encode(
+            encoder_, input_buffer_,
+            kSampleRateHz / 100 * num_10ms_frames_per_packet_, encoded.data());
+        RTC_CHECK_GE(r, 0);
+
+        return static_cast<size_t>(r);
+      });
+
+  RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
+
+  EncodedInfo info;
+  info.encoded_bytes = encoded_bytes;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kIlbc;
+  return info;
+}
+
+void AudioEncoderIlbcImpl::Reset() {
+  if (encoder_)
+    RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
+  const int encoder_frame_size_ms =
+      frame_size_ms_ > 30 ? frame_size_ms_ / 2 : frame_size_ms_;
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+  num_10ms_frames_buffered_ = 0;
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderIlbcImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
+  switch (num_10ms_frames_per_packet_) {
+    case 2:
+      return 38;
+    case 3:
+      return 50;
+    case 4:
+      return 2 * 38;
+    case 6:
+      return 2 * 50;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
new file mode 100644
index 0000000000..c8dfa2ca6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+namespace webrtc {
+
+class AudioEncoderIlbcImpl final : public AudioEncoder {
+ public:
+  AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
+  ~AudioEncoderIlbcImpl() override;
+
+  AudioEncoderIlbcImpl(const AudioEncoderIlbcImpl&) = delete;
+  AudioEncoderIlbcImpl& operator=(const AudioEncoderIlbcImpl&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ private:
+  size_t RequiredOutputSizeBytes() const;
+
+  static constexpr size_t kMaxSamplesPerPacket = 480;
+  const int frame_size_ms_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
+  uint32_t first_timestamp_in_buffer_;
+  int16_t input_buffer_[kMaxSamplesPerPacket];
+  IlbcEncoderInstance* encoder_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
new file mode 100644
index 0000000000..c915a2f9f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/augmented_cb_corr.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+    int16_t *target,   /* (i) Target vector */
+    int16_t *buffer,   /* (i) Memory buffer */
+    int16_t *interpSamples, /* (i) buffer with
+                                     interpolated samples */
+    int32_t *crossDot,  /* (o) The cross correlation between
+                                 the target and the Augmented
+                                 vector */
+    size_t low,    /* (i) Lag to start from (typically
+                             20) */
+    size_t high,   /* (i) Lag to end at (typically 39) */
+    int scale)   /* (i) Scale factor to use for
+                              the crossDot */
+{
+  size_t lagcount;
+  size_t ilow;
+  int16_t *targetPtr;
+  int32_t *crossDotPtr;
+  int16_t *iSPtr=interpSamples;
+
+  /* Calculate the correlation between the target and the
+     interpolated codebook. The correlation is calculated in
+     3 sections with the interpolated part in the middle */
+  crossDotPtr=crossDot;
+  for (lagcount=low; lagcount<=high; lagcount++) {
+
+    ilow = lagcount - 4;
+
+    /* Compute dot product for the first (lagcount-4) samples */
+    (*crossDotPtr) = WebRtcSpl_DotProductWithScale(target, buffer-lagcount, ilow, scale);
+
+    /* Compute dot product on the interpolated samples */
+    (*crossDotPtr) += WebRtcSpl_DotProductWithScale(target+ilow, iSPtr, 4, scale);
+    targetPtr = target + lagcount;
+    iSPtr += lagcount-ilow;
+
+    /* Compute dot product for the remaining samples */
+    (*crossDotPtr) += WebRtcSpl_DotProductWithScale(targetPtr, buffer-lagcount, SUBL-lagcount, scale);
+    crossDotPtr++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
new file mode 100644
index 0000000000..2e9612e51a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Calculate correlation between target and Augmented codebooks
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+    int16_t* target,        /* (i) Target vector */
+    int16_t* buffer,        /* (i) Memory buffer */
+    int16_t* interpSamples, /* (i) buffer with
+                                           interpolated samples */
+    int32_t* crossDot,      /* (o) The cross correlation between
+                                               the target and the Augmented
+                                               vector */
+    size_t low,             /* (i) Lag to start from (typically
+                                                            20) */
+    size_t high,            /* (i) Lag to end at (typically 39 */
+    int scale);             /* (i) Scale factor to use for the crossDot */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c
new file mode 100644
index 0000000000..1a9b882adf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+/* The output is in the same domain as the input */
+void WebRtcIlbcfix_BwExpand(
+    int16_t *out, /* (o) the bandwidth expanded lpc coefficients */
+    int16_t *in,  /* (i) the lpc coefficients before bandwidth
+                                   expansion */
+    int16_t *coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
+                            ) {
+  int i;
+
+  out[0] = in[0];
+  for (i = 1; i < length; i++) {
+    /* out[i] = coef[i] * in[i] with rounding.
+       in[] and out[] are in Q12 and coef[] is in Q15
+    */
+    out[i] = (int16_t)((coef[i] * in[i] + 16384) >> 15);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.h
new file mode 100644
index 0000000000..ff9b0b302e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_BwExpand(
+    int16_t* out,  /* (o) the bandwidth expanded lpc coefficients */
+    int16_t* in,   /* (i) the lpc coefficients before bandwidth
+                                    expansion */
+    int16_t* coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
new file mode 100644
index 0000000000..1e9a7040c7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/gain_dequant.h"
+#include "modules/audio_coding/codecs/ilbc/get_cd_vec.h"
+#include "rtc_base/sanitizer.h"
+
+// An arithmetic operation that is allowed to overflow. (It's still undefined
+// behavior, so not a good idea; this just makes UBSan ignore the violation, so
+// that our old code can continue to do what it's always been doing.)
+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
+    OverflowingAddS32S32ToS32(int32_t a, int32_t b) {
+  return a + b;
+}
+
+/*----------------------------------------------------------------*
+ *  Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_CbConstruct(
+    int16_t* decvector,        /* (o) Decoded vector */
+    const int16_t* index,      /* (i) Codebook indices */
+    const int16_t* gain_index, /* (i) Gain quantization indices */
+    int16_t* mem,              /* (i) Buffer for codevector construction */
+    size_t lMem,               /* (i) Length of buffer */
+    size_t veclen) {           /* (i) Length of vector */
+  size_t j;
+  int16_t gain[CB_NSTAGES];
+  /* Stack based */
+  int16_t cbvec0[SUBL];
+  int16_t cbvec1[SUBL];
+  int16_t cbvec2[SUBL];
+  int32_t a32;
+  int16_t *gainPtr;
+
+  /* gain de-quantization */
+
+  gain[0] = WebRtcIlbcfix_GainDequant(gain_index[0], 16384, 0);
+  gain[1] = WebRtcIlbcfix_GainDequant(gain_index[1], gain[0], 1);
+  gain[2] = WebRtcIlbcfix_GainDequant(gain_index[2], gain[1], 2);
+
+  /* codebook vector construction and construction of total vector */
+
+  /* Stack based */
+  if (!WebRtcIlbcfix_GetCbVec(cbvec0, mem, (size_t)index[0], lMem, veclen))
+    return false;  // Failure.
+  if (!WebRtcIlbcfix_GetCbVec(cbvec1, mem, (size_t)index[1], lMem, veclen))
+    return false;  // Failure.
+  if (!WebRtcIlbcfix_GetCbVec(cbvec2, mem, (size_t)index[2], lMem, veclen))
+    return false;  // Failure.
+
+  gainPtr = &gain[0];
+  for (j=0;j<veclen;j++) {
+    a32 = (*gainPtr++) * cbvec0[j];
+    a32 += (*gainPtr++) * cbvec1[j];
+    a32 = OverflowingAddS32S32ToS32(a32, (*gainPtr) * cbvec2[j]);
+    gainPtr -= 2;
+    decvector[j] = (int16_t)((a32 + 8192) >> 14);
+  }
+
+  return true;  // Success.
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
new file mode 100644
index 0000000000..8f7c663164
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+
+#include <stdbool.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+// Returns true on success, false on failure.
+ABSL_MUST_USE_RESULT
+bool WebRtcIlbcfix_CbConstruct(
+    int16_t* decvector,        /* (o) Decoded vector */
+    const int16_t* index,      /* (i) Codebook indices */
+    const int16_t* gain_index, /* (i) Gain quantization indices */
+    int16_t* mem,              /* (i) Buffer for codevector construction */
+    size_t lMem,               /* (i) Length of buffer */
+    size_t veclen              /* (i) Length of vector */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
new file mode 100644
index 0000000000..21e4197607
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergy.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy.h"
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Function WebRtcIlbcfix_CbMemEnergy computes the energy of all
+ * the vectors in the codebook memory that will be used in the
+ * following search for the best match.
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbMemEnergy(
+    size_t range,
+    int16_t *CB,   /* (i) The CB memory (1:st section) */
+    int16_t *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
+    size_t lMem,   /* (i) Length of the CB memory */
+    size_t lTarget,   /* (i) Length of the target vector */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
+                               ) {
+  int16_t *ppi, *ppo, *pp;
+  int32_t energy, tmp32;
+
+  /* Compute the energy and store it in a vector. Also the
+   * corresponding shift values are stored. The energy values
+   * are reused in all three stages. */
+
+  /* Calculate the energy in the first block of 'lTarget' sampels. */
+  ppi = CB+lMem-lTarget-1;
+  ppo = CB+lMem-1;
+
+  pp=CB+lMem-lTarget;
+  energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+  /* Normalize the energy and store the number of shifts */
+  energyShifts[0] = (int16_t)WebRtcSpl_NormW32(energy);
+  tmp32 = energy << energyShifts[0];
+  energyW16[0] = (int16_t)(tmp32 >> 16);
+
+  /* Compute the energy of the rest of the cb memory
+   * by step wise adding and subtracting the next
+   * sample and the last sample respectively. */
+  WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, 0);
+
+  /* Next, precompute the energy values for the filtered cb section */
+  energy=0;
+  pp=filteredCB+lMem-lTarget;
+
+  energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+  /* Normalize the energy and store the number of shifts */
+  energyShifts[base_size] = (int16_t)WebRtcSpl_NormW32(energy);
+  tmp32 = energy << energyShifts[base_size];
+  energyW16[base_size] = (int16_t)(tmp32 >> 16);
+
+  ppi = filteredCB + lMem - 1 - lTarget;
+  ppo = filteredCB + lMem - 1;
+
+  WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, base_size);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
new file mode 100644
index 0000000000..17ec337dc6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergy.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbMemEnergy(
+    size_t range,
+    int16_t* CB,           /* (i) The CB memory (1:st section) */
+    int16_t* filteredCB,   /* (i) The filtered CB memory (2:nd section) */
+    size_t lMem,           /* (i) Length of the CB memory */
+    size_t lTarget,        /* (i) Length of the target vector */
+    int16_t* energyW16,    /* (o) Energy in the CB vectors */
+    int16_t* energyShifts, /* (o) Shift value of the energy */
+    int scale,             /* (i) The scaling of all energy values */
+    size_t base_size /* (i) Index to where energy values should be stored */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
new file mode 100644
index 0000000000..0619bbe422
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyAugmentation.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbMemEnergyAugmentation(
+    int16_t *interpSamples, /* (i) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size,  /* (i) Index to where energy values should be stored */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts /* (o) Shift value of the energy */
+                                           ){
+  int32_t energy, tmp32;
+  int16_t *ppe, *pp, *interpSamplesPtr;
+  int16_t *CBmemPtr;
+  size_t lagcount;
+  int16_t *enPtr=&energyW16[base_size-20];
+  int16_t *enShPtr=&energyShifts[base_size-20];
+  int32_t nrjRecursive;
+
+  CBmemPtr = CBmem+147;
+  interpSamplesPtr = interpSamples;
+
+  /* Compute the energy for the first (low-5) noninterpolated samples */
+  nrjRecursive = WebRtcSpl_DotProductWithScale( CBmemPtr-19, CBmemPtr-19, 15, scale);
+  ppe = CBmemPtr - 20;
+
+  for (lagcount=20; lagcount<=39; lagcount++) {
+
+    /* Update the energy recursively to save complexity */
+    nrjRecursive += (*ppe * *ppe) >> scale;
+    ppe--;
+    energy = nrjRecursive;
+
+    /* interpolation */
+    energy += WebRtcSpl_DotProductWithScale(interpSamplesPtr, interpSamplesPtr, 4, scale);
+    interpSamplesPtr += 4;
+
+    /* Compute energy for the remaining samples */
+    pp = CBmemPtr - lagcount;
+    energy += WebRtcSpl_DotProductWithScale(pp, pp, SUBL-lagcount, scale);
+
+    /* Normalize the energy and store the number of shifts */
+    (*enShPtr) = (int16_t)WebRtcSpl_NormW32(energy);
+    tmp32 = energy << *enShPtr;
+    *enPtr = (int16_t)(tmp32 >> 16);
+    enShPtr++;
+    enPtr++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
new file mode 100644
index 0000000000..d7b7a0d97e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyAugmentation.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbMemEnergyAugmentation(
+    int16_t* interpSamples, /* (i) The interpolated samples */
+    int16_t* CBmem,         /* (i) The CB memory */
+    int scale,              /* (i) The scaling of all energy values */
+    size_t base_size,   /* (i) Index to where energy values should be stored */
+    int16_t* energyW16, /* (o) Energy in the CB vectors */
+    int16_t* energyShifts /* (o) Shift value of the energy */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
new file mode 100644
index 0000000000..58c0c5fe6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyCalc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* Compute the energy of the rest of the cb memory
+ * by step wise adding and subtracting the next
+ * sample and the last sample respectively */
+void WebRtcIlbcfix_CbMemEnergyCalc(
+    int32_t energy,   /* (i) input start energy */
+    size_t range,   /* (i) number of iterations */
+    int16_t *ppi,   /* (i) input pointer 1 */
+    int16_t *ppo,   /* (i) input pointer 2 */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
+                                   )
+{
+  size_t j;
+  int16_t shft;
+  int32_t tmp;
+  int16_t *eSh_ptr;
+  int16_t *eW16_ptr;
+
+
+  eSh_ptr  = &energyShifts[1+base_size];
+  eW16_ptr = &energyW16[1+base_size];
+
+  for (j = 0; j + 1 < range; j++) {
+
+    /* Calculate next energy by a +/-
+       operation on the edge samples */
+    tmp = (*ppi) * (*ppi) - (*ppo) * (*ppo);
+    energy += tmp >> scale;
+    energy = WEBRTC_SPL_MAX(energy, 0);
+
+    ppi--;
+    ppo--;
+
+    /* Normalize the energy into a int16_t and store
+       the number of shifts */
+
+    shft = (int16_t)WebRtcSpl_NormW32(energy);
+    *eSh_ptr++ = shft;
+
+    tmp = energy << shft;
+    *eW16_ptr++ = (int16_t)(tmp >> 16);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
new file mode 100644
index 0000000000..1d1e8d62b9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyCalc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbMemEnergyCalc(
+    int32_t energy,        /* (i) input start energy */
+    size_t range,          /* (i) number of iterations */
+    int16_t* ppi,          /* (i) input pointer 1 */
+    int16_t* ppo,          /* (i) input pointer 2 */
+    int16_t* energyW16,    /* (o) Energy in the CB vectors */
+    int16_t* energyShifts, /* (o) Shift value of the energy */
+    int scale,             /* (i) The scaling of all energy values */
+    size_t base_size /* (i) Index to where energy values should be stored */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c
new file mode 100644
index 0000000000..24b5292354
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -0,0 +1,405 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearch.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_search.h"
+
+#include "modules/audio_coding/codecs/ilbc/augmented_cb_corr.h"
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy.h"
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h"
+#include "modules/audio_coding/codecs/ilbc/cb_search_core.h"
+#include "modules/audio_coding/codecs/ilbc/cb_update_best_index.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/energy_inverse.h"
+#include "modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h"
+#include "modules/audio_coding/codecs/ilbc/gain_quant.h"
+#include "modules/audio_coding/codecs/ilbc/interpolate_samples.h"
+
+/*----------------------------------------------------------------*
+ *  Search routine for codebook encoding and gain quantization.
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbSearch(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) the encoder state structure */
+    int16_t *index,  /* (o) Codebook indices */
+    int16_t *gain_index, /* (o) Gain quantization indices */
+    int16_t *intarget, /* (i) Target vector for encoding */
+    int16_t *decResidual,/* (i) Decoded residual for codebook construction */
+    size_t lMem,  /* (i) Length of buffer */
+    size_t lTarget,  /* (i) Length of vector */
+    int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */
+    size_t block  /* (i) the subblock number */
+                            ) {
+  size_t i, range;
+  int16_t ii, j, stage;
+  int16_t *pp;
+  int16_t tmp;
+  int scale;
+  int16_t bits, temp1, temp2;
+  size_t base_size;
+  int32_t codedEner, targetEner;
+  int16_t gains[CB_NSTAGES+1];
+  int16_t *cb_vecPtr;
+  size_t indexOffset, sInd, eInd;
+  int32_t CritMax=0;
+  int16_t shTotMax=WEBRTC_SPL_WORD16_MIN;
+  size_t bestIndex=0;
+  int16_t bestGain=0;
+  size_t indexNew;
+  int16_t CritNewSh;
+  int32_t CritNew;
+  int32_t *cDotPtr;
+  size_t noOfZeros;
+  int16_t *gainPtr;
+  int32_t t32, tmpW32;
+  int16_t *WebRtcIlbcfix_kGainSq5_ptr;
+  /* Stack based */
+  int16_t CBbuf[CB_MEML+LPC_FILTERORDER+CB_HALFFILTERLEN];
+  int32_t cDot[128];
+  int32_t Crit[128];
+  int16_t targetVec[SUBL+LPC_FILTERORDER];
+  int16_t cbvectors[CB_MEML + 1];  /* Adding one extra position for
+                                            Coverity warnings. */
+  int16_t codedVec[SUBL];
+  int16_t interpSamples[20*4];
+  int16_t interpSamplesFilt[20*4];
+  int16_t energyW16[CB_EXPAND*128];
+  int16_t energyShifts[CB_EXPAND*128];
+  int16_t *inverseEnergy=energyW16;   /* Reuse memory */
+  int16_t *inverseEnergyShifts=energyShifts; /* Reuse memory */
+  int16_t *buf = &CBbuf[LPC_FILTERORDER];
+  int16_t *target = &targetVec[LPC_FILTERORDER];
+  int16_t *aug_vec = (int16_t*)cDot;   /* length [SUBL], reuse memory */
+
+  /* Determine size of codebook sections */
+
+  base_size=lMem-lTarget+1;
+  if (lTarget==SUBL) {
+    base_size=lMem-19;
+  }
+
+  /* weighting of the CB memory */
+  noOfZeros=lMem-WebRtcIlbcfix_kFilterRange[block];
+  WebRtcSpl_MemSetW16(&buf[-LPC_FILTERORDER], 0, noOfZeros+LPC_FILTERORDER);
+  WebRtcSpl_FilterARFastQ12(
+      decResidual+noOfZeros, buf+noOfZeros,
+      weightDenum, LPC_FILTERORDER+1, WebRtcIlbcfix_kFilterRange[block]);
+
+  /* weighting of the target vector */
+  WEBRTC_SPL_MEMCPY_W16(&target[-LPC_FILTERORDER], buf+noOfZeros+WebRtcIlbcfix_kFilterRange[block]-LPC_FILTERORDER, LPC_FILTERORDER);
+  WebRtcSpl_FilterARFastQ12(
+      intarget, target,
+      weightDenum, LPC_FILTERORDER+1, lTarget);
+
+  /* Store target, towards the end codedVec is calculated as
+     the initial target minus the remaining target */
+  WEBRTC_SPL_MEMCPY_W16(codedVec, target, lTarget);
+
+  /* Find the highest absolute value to calculate proper
+     vector scale factor (so that it uses 12 bits) */
+  temp1 = WebRtcSpl_MaxAbsValueW16(buf, lMem);
+  temp2 = WebRtcSpl_MaxAbsValueW16(target, lTarget);
+
+  if ((temp1>0)&&(temp2>0)) {
+    temp1 = WEBRTC_SPL_MAX(temp1, temp2);
+    scale = WebRtcSpl_GetSizeInBits((uint32_t)(temp1 * temp1));
+  } else {
+    /* temp1 or temp2 is negative (maximum was -32768) */
+    scale = 30;
+  }
+
+  /* Scale to so that a mul-add 40 times does not overflow */
+  scale = scale - 25;
+  scale = WEBRTC_SPL_MAX(0, scale);
+
+  /* Compute energy of the original target */
+  targetEner = WebRtcSpl_DotProductWithScale(target, target, lTarget, scale);
+
+  /* Prepare search over one more codebook section. This section
+     is created by filtering the original buffer with a filter. */
+  WebRtcIlbcfix_FilteredCbVecs(cbvectors, buf, lMem, WebRtcIlbcfix_kFilterRange[block]);
+
+  range = WebRtcIlbcfix_kSearchRange[block][0];
+
+  if(lTarget == SUBL) {
+    /* Create the interpolated samples and store them for use in all stages */
+
+    /* First section, non-filtered half of the cb */
+    WebRtcIlbcfix_InterpolateSamples(interpSamples, buf, lMem);
+
+    /* Second section, filtered half of the cb */
+    WebRtcIlbcfix_InterpolateSamples(interpSamplesFilt, cbvectors, lMem);
+
+    /* Compute the CB vectors' energies for the first cb section (non-filtered) */
+    WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamples, buf,
+                                          scale, 20, energyW16, energyShifts);
+
+    /* Compute the CB vectors' energies for the second cb section (filtered cb) */
+    WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamplesFilt, cbvectors, scale,
+                                          base_size + 20, energyW16,
+                                          energyShifts);
+
+    /* Compute the CB vectors' energies and store them in the vector
+     * energyW16. Also the corresponding shift values are stored. The
+     * energy values are used in all three stages. */
+    WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
+                              lTarget, energyW16+20, energyShifts+20, scale, base_size);
+
+  } else {
+    /* Compute the CB vectors' energies and store them in the vector
+     * energyW16. Also the corresponding shift values are stored. The
+     * energy values are used in all three stages. */
+    WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
+                              lTarget, energyW16, energyShifts, scale, base_size);
+
+    /* Set the energy positions 58-63 and 122-127 to zero
+       (otherwise they are uninitialized) */
+    WebRtcSpl_MemSetW16(energyW16+range, 0, (base_size-range));
+    WebRtcSpl_MemSetW16(energyW16+range+base_size, 0, (base_size-range));
+  }
+
+  /* Calculate Inverse Energy (energyW16 is already normalized
+     and will contain the inverse energy in Q29 after this call */
+  WebRtcIlbcfix_EnergyInverse(energyW16, base_size*CB_EXPAND);
+
+  /* The gain value computed in the previous stage is used
+   * as an upper limit to what the next stage gain value
+   * is allowed to be. In stage 0, 16384 (1.0 in Q14) is used as
+   * the upper limit. */
+  gains[0] = 16384;
+
+  for (stage=0; stage<CB_NSTAGES; stage++) {
+
+    /* Set up memories */
+    range = WebRtcIlbcfix_kSearchRange[block][stage];
+
+    /* initialize search measures */
+    CritMax=0;
+    shTotMax=-100;
+    bestIndex=0;
+    bestGain=0;
+
+    /* loop over lags 40+ in the first codebook section, full search */
+    cb_vecPtr = buf+lMem-lTarget;
+
+    /* Calculate all the cross correlations (augmented part of CB) */
+    if (lTarget==SUBL) {
+      WebRtcIlbcfix_AugmentedCbCorr(target, buf+lMem,
+                                    interpSamples, cDot,
+                                    20, 39, scale);
+      cDotPtr=&cDot[20];
+    } else {
+      cDotPtr=cDot;
+    }
+    /* Calculate all the cross correlations (main part of CB) */
+    WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, range, scale, -1);
+
+    /* Adjust the search range for the augmented vectors */
+    if (lTarget==SUBL) {
+      range=WebRtcIlbcfix_kSearchRange[block][stage]+20;
+    } else {
+      range=WebRtcIlbcfix_kSearchRange[block][stage];
+    }
+
+    indexOffset=0;
+
+    /* Search for best index in this part of the vector */
+    WebRtcIlbcfix_CbSearchCore(
+        cDot, range, stage, inverseEnergy,
+        inverseEnergyShifts, Crit,
+        &indexNew, &CritNew, &CritNewSh);
+
+    /* Update the global best index and the corresponding gain */
+    WebRtcIlbcfix_CbUpdateBestIndex(
+        CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew+indexOffset],
+        inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
+        &CritMax, &shTotMax, &bestIndex, &bestGain);
+
+    sInd = ((CB_RESRANGE >> 1) > bestIndex) ?
+        0 : (bestIndex - (CB_RESRANGE >> 1));
+    eInd=sInd+CB_RESRANGE;
+    if (eInd>=range) {
+      eInd=range-1;
+      sInd=eInd-CB_RESRANGE;
+    }
+
+    range = WebRtcIlbcfix_kSearchRange[block][stage];
+
+    if (lTarget==SUBL) {
+      i=sInd;
+      if (sInd<20) {
+        WebRtcIlbcfix_AugmentedCbCorr(target, cbvectors + lMem,
+                                      interpSamplesFilt, cDot, sInd + 20,
+                                      WEBRTC_SPL_MIN(39, (eInd + 20)), scale);
+        i=20;
+        cDotPtr = &cDot[20 - sInd];
+      } else {
+        cDotPtr = cDot;
+      }
+
+      cb_vecPtr = cbvectors+lMem-20-i;
+
+      /* Calculate the cross correlations (main part of the filtered CB) */
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget,
+                                 eInd - i + 1, scale, -1);
+
+    } else {
+      cDotPtr = cDot;
+      cb_vecPtr = cbvectors+lMem-lTarget-sInd;
+
+      /* Calculate the cross correlations (main part of the filtered CB) */
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget,
+                                 eInd - sInd + 1, scale, -1);
+
+    }
+
+    /* Adjust the search range for the augmented vectors */
+    indexOffset=base_size+sInd;
+
+    /* Search for best index in this part of the vector */
+    WebRtcIlbcfix_CbSearchCore(
+        cDot, eInd-sInd+1, stage, inverseEnergy+indexOffset,
+        inverseEnergyShifts+indexOffset, Crit,
+        &indexNew, &CritNew, &CritNewSh);
+
+    /* Update the global best index and the corresponding gain */
+    WebRtcIlbcfix_CbUpdateBestIndex(
+        CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew],
+        inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
+        &CritMax, &shTotMax, &bestIndex, &bestGain);
+
+    index[stage] = (int16_t)bestIndex;
+
+
+    bestGain = WebRtcIlbcfix_GainQuant(bestGain,
+                                       (int16_t)WEBRTC_SPL_ABS_W16(gains[stage]), stage, &gain_index[stage]);
+
+    /* Extract the best (according to measure) codebook vector
+       Also adjust the index, so that the augmented vectors are last.
+       Above these vectors were first...
+    */
+
+    if(lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) {
+
+      if((size_t)index[stage]<base_size) {
+        pp=buf+lMem-lTarget-index[stage];
+      } else {
+        pp=cbvectors+lMem-lTarget-
+            index[stage]+base_size;
+      }
+
+    } else {
+
+      if ((size_t)index[stage]<base_size) {
+        if (index[stage]>=20) {
+          /* Adjust index and extract vector */
+          index[stage]-=20;
+          pp=buf+lMem-lTarget-index[stage];
+        } else {
+          /* Adjust index and extract vector */
+          index[stage]+=(int16_t)(base_size-20);
+
+          WebRtcIlbcfix_CreateAugmentedVec(index[stage]-base_size+40,
+                                           buf+lMem, aug_vec);
+          pp = aug_vec;
+
+        }
+      } else {
+
+        if ((index[stage] - base_size) >= 20) {
+          /* Adjust index and extract vector */
+          index[stage]-=20;
+          pp=cbvectors+lMem-lTarget-
+              index[stage]+base_size;
+        } else {
+          /* Adjust index and extract vector */
+          index[stage]+=(int16_t)(base_size-20);
+          WebRtcIlbcfix_CreateAugmentedVec(index[stage]-2*base_size+40,
+                                           cbvectors+lMem, aug_vec);
+          pp = aug_vec;
+        }
+      }
+    }
+
+    /* Subtract the best codebook vector, according
+       to measure, from the target vector */
+
+    WebRtcSpl_AddAffineVectorToVector(target, pp, (int16_t)(-bestGain),
+                                      (int32_t)8192, (int16_t)14, lTarget);
+
+    /* record quantized gain */
+    gains[stage+1] = bestGain;
+
+  } /* end of Main Loop. for (stage=0;... */
+
+  /* Calculte the coded vector (original target - what's left) */
+  for (i=0;i<lTarget;i++) {
+    codedVec[i]-=target[i];
+  }
+
+  /* Gain adjustment for energy matching */
+  codedEner = WebRtcSpl_DotProductWithScale(codedVec, codedVec, lTarget, scale);
+
+  j=gain_index[0];
+
+  temp1 = (int16_t)WebRtcSpl_NormW32(codedEner);
+  temp2 = (int16_t)WebRtcSpl_NormW32(targetEner);
+
+  if(temp1 < temp2) {
+    bits = 16 - temp1;
+  } else {
+    bits = 16 - temp2;
+  }
+
+  tmp = (int16_t)((gains[1] * gains[1]) >> 14);
+
+  targetEner = (int16_t)WEBRTC_SPL_SHIFT_W32(targetEner, -bits) * tmp;
+
+  tmpW32 = ((int32_t)(gains[1]-1))<<1;
+
+  /* Pointer to the table that contains
+     gain_sq5TblFIX * gain_sq5TblFIX in Q14 */
+  gainPtr=(int16_t*)WebRtcIlbcfix_kGainSq5Sq+gain_index[0];
+  temp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(codedEner, -bits);
+
+  WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[j];
+
+  /* targetEner and codedEner are in Q(-2*scale) */
+  for (ii=gain_index[0];ii<32;ii++) {
+
+    /* Change the index if
+       (codedEnergy*gainTbl[i]*gainTbl[i])<(targetEn*gain[0]*gain[0]) AND
+       gainTbl[i] < 2*gain[0]
+    */
+
+    t32 = temp1 * *gainPtr;
+    t32 = t32 - targetEner;
+    if (t32 < 0) {
+      if ((*WebRtcIlbcfix_kGainSq5_ptr) < tmpW32) {
+        j=ii;
+        WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[ii];
+      }
+    }
+    gainPtr++;
+  }
+  gain_index[0]=j;
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.h
new file mode 100644
index 0000000000..84a52c7868
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearch.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbSearch(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) the encoder state structure */
+    int16_t* index,       /* (o) Codebook indices */
+    int16_t* gain_index,  /* (o) Gain quantization indices */
+    int16_t* intarget,    /* (i) Target vector for encoding */
+    int16_t* decResidual, /* (i) Decoded residual for codebook construction */
+    size_t lMem,          /* (i) Length of buffer */
+    size_t lTarget,       /* (i) Length of vector */
+    int16_t* weightDenum, /* (i) weighting filter coefficients in Q12 */
+    size_t block          /* (i) the subblock number */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
new file mode 100644
index 0000000000..a75e5b0ab8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearchCore.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_search_core.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbSearchCore(
+    int32_t *cDot,    /* (i) Cross Correlation */
+    size_t range,    /* (i) Search range */
+    int16_t stage,    /* (i) Stage of this search */
+    int16_t *inverseEnergy,  /* (i) Inversed energy */
+    int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy
+                                           with the offset 2*16-29 */
+    int32_t *Crit,    /* (o) The criteria */
+    size_t *bestIndex,   /* (o) Index that corresponds to
+                                                   maximum criteria (in this
+                                                   vector) */
+    int32_t *bestCrit,   /* (o) Value of critera for the
+                                                   chosen index */
+    int16_t *bestCritSh)   /* (o) The domain of the chosen
+                                                   criteria */
+{
+  int32_t maxW32, tmp32;
+  int16_t max, sh, tmp16;
+  size_t i;
+  int32_t *cDotPtr;
+  int16_t cDotSqW16;
+  int16_t *inverseEnergyPtr;
+  int32_t *critPtr;
+  int16_t *inverseEnergyShiftPtr;
+
+  /* Don't allow negative values for stage 0 */
+  if (stage==0) {
+    cDotPtr=cDot;
+    for (i=0;i<range;i++) {
+      *cDotPtr=WEBRTC_SPL_MAX(0, (*cDotPtr));
+      cDotPtr++;
+    }
+  }
+
+  /* Normalize cDot to int16_t, calculate the square of cDot and store the upper int16_t */
+  maxW32 = WebRtcSpl_MaxAbsValueW32(cDot, range);
+
+  sh = (int16_t)WebRtcSpl_NormW32(maxW32);
+  cDotPtr = cDot;
+  inverseEnergyPtr = inverseEnergy;
+  critPtr = Crit;
+  inverseEnergyShiftPtr=inverseEnergyShift;
+  max=WEBRTC_SPL_WORD16_MIN;
+
+  for (i=0;i<range;i++) {
+    /* Calculate cDot*cDot and put the result in a int16_t */
+    tmp32 = *cDotPtr << sh;
+    tmp16 = (int16_t)(tmp32 >> 16);
+    cDotSqW16 = (int16_t)(((int32_t)(tmp16)*(tmp16))>>16);
+
+    /* Calculate the criteria (cDot*cDot/energy) */
+    *critPtr = cDotSqW16 * *inverseEnergyPtr;
+
+    /* Extract the maximum shift value under the constraint
+       that the criteria is not zero */
+    if ((*critPtr)!=0) {
+      max = WEBRTC_SPL_MAX((*inverseEnergyShiftPtr), max);
+    }
+
+    inverseEnergyPtr++;
+    inverseEnergyShiftPtr++;
+    critPtr++;
+    cDotPtr++;
+  }
+
+  /* If no max shifts still at initialization value, set shift to zero */
+  if (max==WEBRTC_SPL_WORD16_MIN) {
+    max = 0;
+  }
+
+  /* Modify the criterias, so that all of them use the same Q domain */
+  critPtr=Crit;
+  inverseEnergyShiftPtr=inverseEnergyShift;
+  for (i=0;i<range;i++) {
+    /* Guarantee that the shift value is less than 16
+       in order to simplify for DSP's (and guard against >31) */
+    tmp16 = WEBRTC_SPL_MIN(16, max-(*inverseEnergyShiftPtr));
+
+    (*critPtr)=WEBRTC_SPL_SHIFT_W32((*critPtr),-tmp16);
+    critPtr++;
+    inverseEnergyShiftPtr++;
+  }
+
+  /* Find the index of the best value */
+  *bestIndex = WebRtcSpl_MaxIndexW32(Crit, range);
+  *bestCrit = Crit[*bestIndex];
+
+  /* Calculate total shifts of this criteria */
+  *bestCritSh = 32 - 2*sh + max;
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
new file mode 100644
index 0000000000..5da70e0988
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearchCore.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbSearchCore(
+    int32_t* cDot,               /* (i) Cross Correlation */
+    size_t range,                /* (i) Search range */
+    int16_t stage,               /* (i) Stage of this search */
+    int16_t* inverseEnergy,      /* (i) Inversed energy */
+    int16_t* inverseEnergyShift, /* (i) Shifts of inversed energy
+                                          with the offset 2*16-29 */
+    int32_t* Crit,               /* (o) The criteria */
+    size_t* bestIndex,           /* (o) Index that corresponds to
+                                           maximum criteria (in this
+                                           vector) */
+    int32_t* bestCrit, /* (o) Value of critera for the
+                                chosen index */
+    int16_t* bestCritSh); /* (o) The domain of the chosen
+                                   criteria */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
new file mode 100644
index 0000000000..d6fa4d93d4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbUpdateBestIndex.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_update_best_index.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbUpdateBestIndex(
+    int32_t CritNew,    /* (i) New Potentially best Criteria */
+    int16_t CritNewSh,   /* (i) Shift value of above Criteria */
+    size_t IndexNew,   /* (i) Index of new Criteria */
+    int32_t cDotNew,    /* (i) Cross dot of new index */
+    int16_t invEnergyNew,  /* (i) Inversed energy new index */
+    int16_t energyShiftNew,  /* (i) Energy shifts of new index */
+    int32_t *CritMax,   /* (i/o) Maximum Criteria (so far) */
+    int16_t *shTotMax,   /* (i/o) Shifts of maximum criteria */
+    size_t *bestIndex,   /* (i/o) Index that corresponds to
+                                                   maximum criteria */
+    int16_t *bestGain)   /* (i/o) Gain in Q14 that corresponds
+                                                   to maximum criteria */
+{
+  int16_t shOld, shNew, tmp16;
+  int16_t scaleTmp;
+  int32_t gainW32;
+
+  /* Normalize the new and old Criteria to the same domain */
+  if (CritNewSh>(*shTotMax)) {
+    shOld=WEBRTC_SPL_MIN(31,CritNewSh-(*shTotMax));
+    shNew=0;
+  } else {
+    shOld=0;
+    shNew=WEBRTC_SPL_MIN(31,(*shTotMax)-CritNewSh);
+  }
+
+  /* Compare the two criterias. If the new one is better,
+     calculate the gain and store this index as the new best one
+  */
+
+  if ((CritNew >> shNew) > (*CritMax >> shOld)) {
+
+    tmp16 = (int16_t)WebRtcSpl_NormW32(cDotNew);
+    tmp16 = 16 - tmp16;
+
+    /* Calculate the gain in Q14
+       Compensate for inverseEnergyshift in Q29 and that the energy
+       value was stored in a int16_t (shifted down 16 steps)
+       => 29-14+16 = 31 */
+
+    scaleTmp = -energyShiftNew-tmp16+31;
+    scaleTmp = WEBRTC_SPL_MIN(31, scaleTmp);
+
+    gainW32 = ((int16_t)WEBRTC_SPL_SHIFT_W32(cDotNew, -tmp16) * invEnergyNew) >>
+        scaleTmp;
+
+    /* Check if criteria satisfies Gain criteria (max 1.3)
+       if it is larger set the gain to 1.3
+       (slightly different from FLP version)
+    */
+    if (gainW32>21299) {
+      *bestGain=21299;
+    } else if (gainW32<-21299) {
+      *bestGain=-21299;
+    } else {
+      *bestGain=(int16_t)gainW32;
+    }
+
+    *CritMax=CritNew;
+    *shTotMax=CritNewSh;
+    *bestIndex = IndexNew;
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
new file mode 100644
index 0000000000..1a95d531e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbUpdateBestIndex.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbUpdateBestIndex(
+    int32_t CritNew,        /* (i) New Potentially best Criteria */
+    int16_t CritNewSh,      /* (i) Shift value of above Criteria */
+    size_t IndexNew,        /* (i) Index of new Criteria */
+    int32_t cDotNew,        /* (i) Cross dot of new index */
+    int16_t invEnergyNew,   /* (i) Inversed energy new index */
+    int16_t energyShiftNew, /* (i) Energy shifts of new index */
+    int32_t* CritMax,       /* (i/o) Maximum Criteria (so far) */
+    int16_t* shTotMax,      /* (i/o) Shifts of maximum criteria */
+    size_t* bestIndex,      /* (i/o) Index that corresponds to
+                                      maximum criteria */
+    int16_t* bestGain);     /* (i/o) Gain in Q14 that corresponds
+                                     to maximum criteria */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c
new file mode 100644
index 0000000000..b4eee66219
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Chebyshev.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/chebyshev.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*------------------------------------------------------------------*
+ *  Calculate the Chevyshev polynomial series
+ *  F(w) = 2*exp(-j5w)*C(x)
+ *   C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
+ *   T_i(x) is the i:th order Chebyshev polynomial
+ *------------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_Chebyshev(
+    /* (o) Result of C(x) */
+    int16_t x,  /* (i) Value to the Chevyshev polynomial */
+    int16_t *f  /* (i) The coefficients in the polynomial */
+                                      ) {
+  int16_t b1_high, b1_low; /* Use the high, low format to increase the accuracy */
+  int32_t b2;
+  int32_t tmp1W32;
+  int32_t tmp2W32;
+  int i;
+
+  b2 = (int32_t)0x1000000; /* b2 = 1.0 (Q23) */
+  /* Calculate b1 = 2*x + f[1] */
+  tmp1W32 = (x << 10) + (f[1] << 14);
+
+  for (i = 2; i < 5; i++) {
+    tmp2W32 = tmp1W32;
+
+    /* Split b1 (in tmp1W32) into a high and low part */
+    b1_high = (int16_t)(tmp1W32 >> 16);
+    b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1);
+
+    /* Calculate 2*x*b1-b2+f[i] */
+    tmp1W32 = ((b1_high * x + ((b1_low * x) >> 15)) << 2) - b2 + (f[i] << 14);
+
+    /* Update b2 for next round */
+    b2 = tmp2W32;
+  }
+
+  /* Split b1 (in tmp1W32) into a high and low part */
+  b1_high = (int16_t)(tmp1W32 >> 16);
+  b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1);
+
+  /* tmp1W32 = x*b1 - b2 + f[i]/2 */
+  tmp1W32 = ((b1_high * x) << 1) + (((b1_low * x) >> 15) << 1) -
+      b2 + (f[i] << 13);
+
+  /* Handle overflows and set to maximum or minimum int16_t instead */
+  if (tmp1W32>((int32_t)33553408)) {
+    return(WEBRTC_SPL_WORD16_MAX);
+  } else if (tmp1W32<((int32_t)-33554432)) {
+    return(WEBRTC_SPL_WORD16_MIN);
+  } else {
+    return (int16_t)(tmp1W32 >> 10);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.h
new file mode 100644
index 0000000000..7e7742c5cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Chebyshev.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*------------------------------------------------------------------*
+ *  Calculate the Chevyshev polynomial series
+ *  F(w) = 2*exp(-j5w)*C(x)
+ *   C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
+ *   T_i(x) is the i:th order Chebyshev polynomial
+ *------------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_Chebyshev(
+    /* (o) Result of C(x) */
+    int16_t x, /* (i) Value to the Chevyshev polynomial */
+    int16_t* f /* (i) The coefficients in the polynomial */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
new file mode 100644
index 0000000000..452bc78e3b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CompCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/comp_corr.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Compute cross correlation and pitch gain for pitch prediction
+ *  of last subframe at given lag.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CompCorr(
+    int32_t *corr, /* (o) cross correlation */
+    int32_t *ener, /* (o) energy */
+    int16_t *buffer, /* (i) signal buffer */
+    size_t lag,  /* (i) pitch lag */
+    size_t bLen, /* (i) length of buffer */
+    size_t sRange, /* (i) correlation search length */
+    int16_t scale /* (i) number of rightshifts to use */
+                            ){
+  int16_t *w16ptr;
+
+  w16ptr=&buffer[bLen-sRange-lag];
+
+  /* Calculate correlation and energy */
+  (*corr)=WebRtcSpl_DotProductWithScale(&buffer[bLen-sRange], w16ptr, sRange, scale);
+  (*ener)=WebRtcSpl_DotProductWithScale(w16ptr, w16ptr, sRange, scale);
+
+  /* For zero energy set the energy to 0 in order to avoid potential
+     problems for coming divisions */
+  if (*ener == 0) {
+    *corr = 0;
+    *ener = 1;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
new file mode 100644
index 0000000000..010c6a1ce5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CompCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Compute cross correlation and pitch gain for pitch prediction
+ *  of last subframe at given lag.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CompCorr(int32_t* corr,   /* (o) cross correlation */
+                            int32_t* ener,   /* (o) energy */
+                            int16_t* buffer, /* (i) signal buffer */
+                            size_t lag,      /* (i) pitch lag */
+                            size_t bLen,     /* (i) length of buffer */
+                            size_t sRange,   /* (i) correlation search length */
+                            int16_t scale /* (i) number of rightshifts to use */
+                            );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m
new file mode 100644
index 0000000000..4bda83622f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m
@@ -0,0 +1,57 @@
+% % Copyright(c) 2011 The WebRTC project authors.All Rights Reserved.%
+        % Use of this source code is governed by a BSD
+    -
+    style license % that can be found in the LICENSE file in the root of the source
+        % tree.An additional intellectual property rights grant can be found
+        % in the file PATENTS.All contributing project authors may
+        % be found in the AUTHORS file in the root of the source tree.%
+
+        clear;
+pack;
+%
+% Enter the path to YOUR executable and remember to define the perprocessor
+% variable PRINT_MIPS te get the instructions printed to the screen.
+%
+command = '!iLBCtest.exe 30 speechAndBGnoise.pcm out1.bit out1.pcm tlm10_30ms.dat';
+cout=' > st.txt';   %saves to matlab variable 'st'
+eval(strcat(command,cout));
+if(length(cout)>3)
+    load st.txt
+else
+    disp('No cout file to load')
+end
+
+% initialize vector to zero
+index = find(st(1:end,1)==-1);
+indexnonzero = find(st(1:end,1)>0);
+frames = length(index)-indexnonzero(1)+1;
+start = indexnonzero(1) - 1;
+functionOrder=max(st(:,2));
+new=zeros(frames,functionOrder);
+
+for i = 1:frames,
+    for j = index(start-1+i)+1:(index(start+i)-1),
+        new(i,st(j,2)) = new(i,st(j,2)) + st(j,1);
+    end
+end
+
+result=zeros(functionOrder,3);
+for i=1:functionOrder
+    nonzeroelements = find(new(1:end,i)>0);
+    result(i,1)=i;
+    
+    % Compute each function's mean complexity
+    % result(i,2)=(sum(new(nonzeroelements,i))/(length(nonzeroelements)*0.03))/1000000;
+    
+    % Compute each function's maximum complexity in encoding
+    % and decoding respectively and then add it together:
+    % result(i,3)=(max(new(1:end,i))/0.03)/1000000;
+    result(i,3)=(max(new(1:size(new,1)/2,i))/0.03)/1000000 + (max(new(size(new,1)/2+1:end,i))/0.03)/1000000;
+end
+
+result
+
+% Compute maximum complexity for a single frame (enc/dec separately and together)
+maxEncComplexityInAFrame = (max(sum(new(1:size(new,1)/2,:),2))/0.03)/1000000
+maxDecComplexityInAFrame = (max(sum(new(size(new,1)/2+1:end,:),2))/0.03)/1000000
+totalComplexity = maxEncComplexityInAFrame + maxDecComplexityInAFrame
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c
new file mode 100644
index 0000000000..22f2acb330
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c
@@ -0,0 +1,667 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* HP Filters {b[0] b[1] b[2] -a[1] -a[2]} */
+
+const int16_t WebRtcIlbcfix_kHpInCoefs[5] = {3798, -7596, 3798, 7807, -3733};
+const int16_t WebRtcIlbcfix_kHpOutCoefs[5] = {3849, -7699, 3849, 7918, -3833};
+
+/* Window in Q11 to window the energies of the 5 choises (3 for 20ms) in the choise for
+   the 80 sample start state
+*/
+const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[NSUB_MAX-1]= {
+  1638, 1843, 2048, 1843, 1638
+};
+
+/* LP Filter coeffs used for downsampling */
+const int16_t WebRtcIlbcfix_kLpFiltCoefs[FILTERORDER_DS_PLUS1]= {
+  -273, 512, 1297, 1696, 1297, 512, -273
+};
+
+/* Constants used in the LPC calculations */
+
+/* Hanning LPC window (in Q15) */
+const int16_t WebRtcIlbcfix_kLpcWin[BLOCKL_MAX] = {
+  6, 22, 50, 89, 139, 200, 272, 355, 449, 554, 669, 795,
+  932, 1079, 1237, 1405, 1583, 1771, 1969, 2177, 2395, 2622, 2858, 3104,
+  3359, 3622, 3894, 4175, 4464, 4761, 5066, 5379, 5699, 6026, 6361, 6702,
+  7050, 7404, 7764, 8130, 8502, 8879, 9262, 9649, 10040, 10436, 10836, 11240,
+  11647, 12058, 12471, 12887, 13306, 13726, 14148, 14572, 14997, 15423, 15850, 16277,
+  16704, 17131, 17558, 17983, 18408, 18831, 19252, 19672, 20089, 20504, 20916, 21325,
+  21730, 22132, 22530, 22924, 23314, 23698, 24078, 24452, 24821, 25185, 25542, 25893,
+  26238, 26575, 26906, 27230, 27547, 27855, 28156, 28450, 28734, 29011, 29279, 29538,
+  29788, 30029, 30261, 30483, 30696, 30899, 31092, 31275, 31448, 31611, 31764, 31906,
+  32037, 32158, 32268, 32367, 32456, 32533, 32600, 32655, 32700, 32733, 32755, 32767,
+  32767, 32755, 32733, 32700, 32655, 32600, 32533, 32456, 32367, 32268, 32158, 32037,
+  31906, 31764, 31611, 31448, 31275, 31092, 30899, 30696, 30483, 30261, 30029, 29788,
+  29538, 29279, 29011, 28734, 28450, 28156, 27855, 27547, 27230, 26906, 26575, 26238,
+  25893, 25542, 25185, 24821, 24452, 24078, 23698, 23314, 22924, 22530, 22132, 21730,
+  21325, 20916, 20504, 20089, 19672, 19252, 18831, 18408, 17983, 17558, 17131, 16704,
+  16277, 15850, 15423, 14997, 14572, 14148, 13726, 13306, 12887, 12471, 12058, 11647,
+  11240, 10836, 10436, 10040, 9649, 9262, 8879, 8502, 8130, 7764, 7404, 7050,
+  6702, 6361, 6026, 5699, 5379, 5066, 4761, 4464, 4175, 3894, 3622, 3359,
+  3104, 2858, 2622, 2395, 2177, 1969, 1771, 1583, 1405, 1237, 1079, 932,
+  795, 669, 554, 449, 355, 272, 200, 139, 89, 50, 22, 6
+};
+
+/* Asymmetric LPC window (in Q15)*/
+const int16_t WebRtcIlbcfix_kLpcAsymWin[BLOCKL_MAX] = {
+  2, 7, 15, 27, 42, 60, 81, 106, 135, 166, 201, 239,
+  280, 325, 373, 424, 478, 536, 597, 661, 728, 798, 872, 949,
+  1028, 1111, 1197, 1287, 1379, 1474, 1572, 1674, 1778, 1885, 1995, 2108,
+  2224, 2343, 2465, 2589, 2717, 2847, 2980, 3115, 3254, 3395, 3538, 3684,
+  3833, 3984, 4138, 4295, 4453, 4615, 4778, 4944, 5112, 5283, 5456, 5631,
+  5808, 5987, 6169, 6352, 6538, 6725, 6915, 7106, 7300, 7495, 7692, 7891,
+  8091, 8293, 8497, 8702, 8909, 9118, 9328, 9539, 9752, 9966, 10182, 10398,
+  10616, 10835, 11055, 11277, 11499, 11722, 11947, 12172, 12398, 12625, 12852, 13080,
+  13309, 13539, 13769, 14000, 14231, 14463, 14695, 14927, 15160, 15393, 15626, 15859,
+  16092, 16326, 16559, 16792, 17026, 17259, 17492, 17725, 17957, 18189, 18421, 18653,
+  18884, 19114, 19344, 19573, 19802, 20030, 20257, 20483, 20709, 20934, 21157, 21380,
+  21602, 21823, 22042, 22261, 22478, 22694, 22909, 23123, 23335, 23545, 23755, 23962,
+  24168, 24373, 24576, 24777, 24977, 25175, 25371, 25565, 25758, 25948, 26137, 26323,
+  26508, 26690, 26871, 27049, 27225, 27399, 27571, 27740, 27907, 28072, 28234, 28394,
+  28552, 28707, 28860, 29010, 29157, 29302, 29444, 29584, 29721, 29855, 29987, 30115,
+  30241, 30364, 30485, 30602, 30717, 30828, 30937, 31043, 31145, 31245, 31342, 31436,
+  31526, 31614, 31699, 31780, 31858, 31933, 32005, 32074, 32140, 32202, 32261, 32317,
+  32370, 32420, 32466, 32509, 32549, 32585, 32618, 32648, 32675, 32698, 32718, 32734,
+  32748, 32758, 32764, 32767, 32767, 32667, 32365, 31863, 31164, 30274, 29197, 27939,
+  26510, 24917, 23170, 21281, 19261, 17121, 14876, 12540, 10126, 7650, 5126, 2571
+};
+
+/* Lag window for LPC (Q31) */
+const int32_t WebRtcIlbcfix_kLpcLagWin[LPC_FILTERORDER + 1]={
+  2147483647,   2144885453,   2137754373,   2125918626,   2109459810,
+  2088483140,   2063130336,   2033564590,   1999977009,   1962580174,
+  1921610283};
+
+/* WebRtcIlbcfix_kLpcChirpSyntDenum vector in Q15 corresponding
+ * floating point vector {1 0.9025 0.9025^2 0.9025^3 ...}
+ */
+const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[LPC_FILTERORDER + 1] = {
+  32767, 29573, 26690, 24087,
+  21739, 19619, 17707, 15980,
+  14422, 13016, 11747};
+
+/* WebRtcIlbcfix_kLpcChirpWeightDenum in Q15 corresponding to
+ * floating point vector {1 0.4222 0.4222^2... }
+ */
+const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[LPC_FILTERORDER + 1] = {
+  32767, 13835, 5841, 2466, 1041, 440,
+  186, 78,  33,  14,  6};
+
+/* LSF quantization Q13 domain */
+const int16_t WebRtcIlbcfix_kLsfCb[64 * 3 + 128 * 3 + 128 * 4] = {
+  1273,       2238,       3696,
+  3199,       5309,       8209,
+  3606,       5671,       7829,
+  2815,       5262,       8778,
+  2608,       4027,       5493,
+  1582,       3076,       5945,
+  2983,       4181,       5396,
+  2437,       4322,       6902,
+  1861,       2998,       4613,
+  2007,       3250,       5214,
+  1388,       2459,       4262,
+  2563,       3805,       5269,
+  2036,       3522,       5129,
+  1935,       4025,       6694,
+  2744,       5121,       7338,
+  2810,       4248,       5723,
+  3054,       5405,       7745,
+  1449,       2593,       4763,
+  3411,       5128,       6596,
+  2484,       4659,       7496,
+  1668,       2879,       4818,
+  1812,       3072,       5036,
+  1638,       2649,       3900,
+  2464,       3550,       4644,
+  1853,       2900,       4158,
+  2458,       4163,       5830,
+  2556,       4036,       6254,
+  2703,       4432,       6519,
+  3062,       4953,       7609,
+  1725,       3703,       6187,
+  2221,       3877,       5427,
+  2339,       3579,       5197,
+  2021,       4633,       7037,
+  2216,       3328,       4535,
+  2961,       4739,       6667,
+  2807,       3955,       5099,
+  2788,       4501,       6088,
+  1642,       2755,       4431,
+  3341,       5282,       7333,
+  2414,       3726,       5727,
+  1582,       2822,       5269,
+  2259,       3447,       4905,
+  3117,       4986,       7054,
+  1825,       3491,       5542,
+  3338,       5736,       8627,
+  1789,       3090,       5488,
+  2566,       3720,       4923,
+  2846,       4682,       7161,
+  1950,       3321,       5976,
+  1834,       3383,       6734,
+  3238,       4769,       6094,
+  2031,       3978,       5903,
+  1877,       4068,       7436,
+  2131,       4644,       8296,
+  2764,       5010,       8013,
+  2194,       3667,       6302,
+  2053,       3127,       4342,
+  3523,       6595,      10010,
+  3134,       4457,       5748,
+  3142,       5819,       9414,
+  2223,       4334,       6353,
+  2022,       3224,       4822,
+  2186,       3458,       5544,
+  2552,       4757,       6870,
+  10905,      12917,      14578,
+  9503,      11485,      14485,
+  9518,      12494,      14052,
+  6222,       7487,       9174,
+  7759,       9186,      10506,
+  8315,      12755,      14786,
+  9609,      11486,      13866,
+  8909,      12077,      13643,
+  7369,       9054,      11520,
+  9408,      12163,      14715,
+  6436,       9911,      12843,
+  7109,       9556,      11884,
+  7557,      10075,      11640,
+  6482,       9202,      11547,
+  6463,       7914,      10980,
+  8611,      10427,      12752,
+  7101,       9676,      12606,
+  7428,      11252,      13172,
+  10197,      12955,      15842,
+  7487,      10955,      12613,
+  5575,       7858,      13621,
+  7268,      11719,      14752,
+  7476,      11744,      13795,
+  7049,       8686,      11922,
+  8234,      11314,      13983,
+  6560,      11173,      14984,
+  6405,       9211,      12337,
+  8222,      12054,      13801,
+  8039,      10728,      13255,
+  10066,      12733,      14389,
+  6016,       7338,      10040,
+  6896,       8648,      10234,
+  7538,       9170,      12175,
+  7327,      12608,      14983,
+  10516,      12643,      15223,
+  5538,       7644,      12213,
+  6728,      12221,      14253,
+  7563,       9377,      12948,
+  8661,      11023,      13401,
+  7280,       8806,      11085,
+  7723,       9793,      12333,
+  12225,      14648,      16709,
+  8768,      13389,      15245,
+  10267,      12197,      13812,
+  5301,       7078,      11484,
+  7100,      10280,      11906,
+  8716,      12555,      14183,
+  9567,      12464,      15434,
+  7832,      12305,      14300,
+  7608,      10556,      12121,
+  8913,      11311,      12868,
+  7414,       9722,      11239,
+  8666,      11641,      13250,
+  9079,      10752,      12300,
+  8024,      11608,      13306,
+  10453,      13607,      16449,
+  8135,       9573,      10909,
+  6375,       7741,      10125,
+  10025,      12217,      14874,
+  6985,      11063,      14109,
+  9296,      13051,      14642,
+  8613,      10975,      12542,
+  6583,      10414,      13534,
+  6191,       9368,      13430,
+  5742,       6859,       9260,
+  7723,       9813,      13679,
+  8137,      11291,      12833,
+  6562,       8973,      10641,
+  6062,       8462,      11335,
+  6928,       8784,      12647,
+  7501,       8784,      10031,
+  8372,      10045,      12135,
+  8191,       9864,      12746,
+  5917,       7487,      10979,
+  5516,       6848,      10318,
+  6819,       9899,      11421,
+  7882,      12912,      15670,
+  9558,      11230,      12753,
+  7752,       9327,      11472,
+  8479,       9980,      11358,
+  11418,      14072,      16386,
+  7968,      10330,      14423,
+  8423,      10555,      12162,
+  6337,      10306,      14391,
+  8850,      10879,      14276,
+  6750,      11885,      15710,
+  7037,       8328,       9764,
+  6914,       9266,      13476,
+  9746,      13949,      15519,
+  11032,      14444,      16925,
+  8032,      10271,      11810,
+  10962,      13451,      15833,
+  10021,      11667,      13324,
+  6273,       8226,      12936,
+  8543,      10397,      13496,
+  7936,      10302,      12745,
+  6769,       8138,      10446,
+  6081,       7786,      11719,
+  8637,      11795,      14975,
+  8790,      10336,      11812,
+  7040,       8490,      10771,
+  7338,      10381,      13153,
+  6598,       7888,       9358,
+  6518,       8237,      12030,
+  9055,      10763,      12983,
+  6490,      10009,      12007,
+  9589,      12023,      13632,
+  6867,       9447,      10995,
+  7930,       9816,      11397,
+  10241,      13300,      14939,
+  5830,       8670,      12387,
+  9870,      11915,      14247,
+  9318,      11647,      13272,
+  6721,      10836,      12929,
+  6543,       8233,       9944,
+  8034,      10854,      12394,
+  9112,      11787,      14218,
+  9302,      11114,      13400,
+  9022,      11366,      13816,
+  6962,      10461,      12480,
+  11288,      13333,      15222,
+  7249,       8974,      10547,
+  10566,      12336,      14390,
+  6697,      11339,      13521,
+  11851,      13944,      15826,
+  6847,       8381,      11349,
+  7509,       9331,      10939,
+  8029,       9618,      11909,
+  13973,      17644,      19647,      22474,
+  14722,      16522,      20035,      22134,
+  16305,      18179,      21106,      23048,
+  15150,      17948,      21394,      23225,
+  13582,      15191,      17687,      22333,
+  11778,      15546,      18458,      21753,
+  16619,      18410,      20827,      23559,
+  14229,      15746,      17907,      22474,
+  12465,      15327,      20700,      22831,
+  15085,      16799,      20182,      23410,
+  13026,      16935,      19890,      22892,
+  14310,      16854,      19007,      22944,
+  14210,      15897,      18891,      23154,
+  14633,      18059,      20132,      22899,
+  15246,      17781,      19780,      22640,
+  16396,      18904,      20912,      23035,
+  14618,      17401,      19510,      21672,
+  15473,      17497,      19813,      23439,
+  18851,      20736,      22323,      23864,
+  15055,      16804,      18530,      20916,
+  16490,      18196,      19990,      21939,
+  11711,      15223,      21154,      23312,
+  13294,      15546,      19393,      21472,
+  12956,      16060,      20610,      22417,
+  11628,      15843,      19617,      22501,
+  14106,      16872,      19839,      22689,
+  15655,      18192,      20161,      22452,
+  12953,      15244,      20619,      23549,
+  15322,      17193,      19926,      21762,
+  16873,      18676,      20444,      22359,
+  14874,      17871,      20083,      21959,
+  11534,      14486,      19194,      21857,
+  17766,      19617,      21338,      23178,
+  13404,      15284,      19080,      23136,
+  15392,      17527,      19470,      21953,
+  14462,      16153,      17985,      21192,
+  17734,      19750,      21903,      23783,
+  16973,      19096,      21675,      23815,
+  16597,      18936,      21257,      23461,
+  15966,      17865,      20602,      22920,
+  15416,      17456,      20301,      22972,
+  18335,      20093,      21732,      23497,
+  15548,      17217,      20679,      23594,
+  15208,      16995,      20816,      22870,
+  13890,      18015,      20531,      22468,
+  13211,      15377,      19951,      22388,
+  12852,      14635,      17978,      22680,
+  16002,      17732,      20373,      23544,
+  11373,      14134,      19534,      22707,
+  17329,      19151,      21241,      23462,
+  15612,      17296,      19362,      22850,
+  15422,      19104,      21285,      23164,
+  13792,      17111,      19349,      21370,
+  15352,      17876,      20776,      22667,
+  15253,      16961,      18921,      22123,
+  14108,      17264,      20294,      23246,
+  15785,      17897,      20010,      21822,
+  17399,      19147,      20915,      22753,
+  13010,      15659,      18127,      20840,
+  16826,      19422,      22218,      24084,
+  18108,      20641,      22695,      24237,
+  18018,      20273,      22268,      23920,
+  16057,      17821,      21365,      23665,
+  16005,      17901,      19892,      23016,
+  13232,      16683,      21107,      23221,
+  13280,      16615,      19915,      21829,
+  14950,      18575,      20599,      22511,
+  16337,      18261,      20277,      23216,
+  14306,      16477,      21203,      23158,
+  12803,      17498,      20248,      22014,
+  14327,      17068,      20160,      22006,
+  14402,      17461,      21599,      23688,
+  16968,      18834,      20896,      23055,
+  15070,      17157,      20451,      22315,
+  15419,      17107,      21601,      23946,
+  16039,      17639,      19533,      21424,
+  16326,      19261,      21745,      23673,
+  16489,      18534,      21658,      23782,
+  16594,      18471,      20549,      22807,
+  18973,      21212,      22890,      24278,
+  14264,      18674,      21123,      23071,
+  15117,      16841,      19239,      23118,
+  13762,      15782,      20478,      23230,
+  14111,      15949,      20058,      22354,
+  14990,      16738,      21139,      23492,
+  13735,      16971,      19026,      22158,
+  14676,      17314,      20232,      22807,
+  16196,      18146,      20459,      22339,
+  14747,      17258,      19315,      22437,
+  14973,      17778,      20692,      23367,
+  15715,      17472,      20385,      22349,
+  15702,      18228,      20829,      23410,
+  14428,      16188,      20541,      23630,
+  16824,      19394,      21365,      23246,
+  13069,      16392,      18900,      21121,
+  12047,      16640,      19463,      21689,
+  14757,      17433,      19659,      23125,
+  15185,      16930,      19900,      22540,
+  16026,      17725,      19618,      22399,
+  16086,      18643,      21179,      23472,
+  15462,      17248,      19102,      21196,
+  17368,      20016,      22396,      24096,
+  12340,      14475,      19665,      23362,
+  13636,      16229,      19462,      22728,
+  14096,      16211,      19591,      21635,
+  12152,      14867,      19943,      22301,
+  14492,      17503,      21002,      22728,
+  14834,      16788,      19447,      21411,
+  14650,      16433,      19326,      22308,
+  14624,      16328,      19659,      23204,
+  13888,      16572,      20665,      22488,
+  12977,      16102,      18841,      22246,
+  15523,      18431,      21757,      23738,
+  14095,      16349,      18837,      20947,
+  13266,      17809,      21088,      22839,
+  15427,      18190,      20270,      23143,
+  11859,      16753,      20935,      22486,
+  12310,      17667,      21736,      23319,
+  14021,      15926,      18702,      22002,
+  12286,      15299,      19178,      21126,
+  15703,      17491,      21039,      23151,
+  12272,      14018,      18213,      22570,
+  14817,      16364,      18485,      22598,
+  17109,      19683,      21851,      23677,
+  12657,      14903,      19039,      22061,
+  14713,      16487,      20527,      22814,
+  14635,      16726,      18763,      21715,
+  15878,      18550,      20718,      22906
+};
+
+const int16_t WebRtcIlbcfix_kLsfDimCb[LSF_NSPLIT] = {3, 3, 4};
+const int16_t WebRtcIlbcfix_kLsfSizeCb[LSF_NSPLIT] = {64,128,128};
+
+const int16_t WebRtcIlbcfix_kLsfMean[LPC_FILTERORDER] = {
+  2308,       3652,       5434,       7885,
+  10255,      12559,      15160,      17513,
+  20328,      22752};
+
+const int16_t WebRtcIlbcfix_kLspMean[LPC_FILTERORDER] = {
+  31476, 29565, 25819, 18725, 10276,
+  1236, -9049, -17600, -25884, -30618
+};
+
+/* Q14 */
+const int16_t WebRtcIlbcfix_kLsfWeight20ms[4] = {12288, 8192, 4096, 0};
+const int16_t WebRtcIlbcfix_kLsfWeight30ms[6] = {8192, 16384, 10923, 5461, 0, 0};
+
+/*
+   cos(x) in Q15
+   WebRtcIlbcfix_kCos[i] = cos(pi*i/64.0)
+   used in WebRtcIlbcfix_Lsp2Lsf()
+*/
+
+const int16_t WebRtcIlbcfix_kCos[64] = {
+  32767,  32729,  32610,  32413,  32138,  31786,  31357,  30853,
+  30274,  29622,  28899,  28106,  27246,  26320,  25330,  24279,
+  23170,  22006,  20788,  19520,  18205,  16846,  15447,  14010,
+  12540,  11039,   9512,   7962,   6393,   4808,   3212,   1608,
+  0,  -1608,  -3212,  -4808,  -6393,  -7962,  -9512, -11039,
+  -12540, -14010, -15447, -16846, -18205, -19520, -20788, -22006,
+  -23170, -24279, -25330, -26320, -27246, -28106, -28899, -29622,
+  -30274, -30853, -31357, -31786, -32138, -32413, -32610, -32729
+};
+
+/*
+   Derivative in Q19, used to interpolate between the
+   WebRtcIlbcfix_kCos[] values to get a more exact y = cos(x)
+*/
+const int16_t WebRtcIlbcfix_kCosDerivative[64] = {
+  -632,  -1893,  -3150,  -4399,  -5638,  -6863,  -8072,  -9261,
+  -10428, -11570, -12684, -13767, -14817, -15832, -16808, -17744,
+  -18637, -19486, -20287, -21039, -21741, -22390, -22986, -23526,
+  -24009, -24435, -24801, -25108, -25354, -25540, -25664, -25726,
+  -25726, -25664, -25540, -25354, -25108, -24801, -24435, -24009,
+  -23526, -22986, -22390, -21741, -21039, -20287, -19486, -18637,
+  -17744, -16808, -15832, -14817, -13767, -12684, -11570, -10428,
+  -9261,  -8072,  -6863,  -5638,  -4399,  -3150,  -1893,   -632};
+
+/*
+  Table in Q15, used for a2lsf conversion
+  WebRtcIlbcfix_kCosGrid[i] = cos((2*pi*i)/(float)(2*COS_GRID_POINTS));
+*/
+
+const int16_t WebRtcIlbcfix_kCosGrid[COS_GRID_POINTS + 1] = {
+  32760, 32723, 32588, 32364, 32051, 31651, 31164, 30591,
+  29935, 29196, 28377, 27481, 26509, 25465, 24351, 23170,
+  21926, 20621, 19260, 17846, 16384, 14876, 13327, 11743,
+  10125, 8480, 6812, 5126, 3425, 1714, 0, -1714, -3425,
+  -5126, -6812, -8480, -10125, -11743, -13327, -14876,
+  -16384, -17846, -19260, -20621, -21926, -23170, -24351,
+  -25465, -26509, -27481, -28377, -29196, -29935, -30591,
+  -31164, -31651, -32051, -32364, -32588, -32723, -32760
+};
+
+/*
+   Derivative of y = acos(x) in Q12
+   used in WebRtcIlbcfix_Lsp2Lsf()
+*/
+
+const int16_t WebRtcIlbcfix_kAcosDerivative[64] = {
+  -26887, -8812, -5323, -3813, -2979, -2444, -2081, -1811,
+  -1608, -1450, -1322, -1219, -1132, -1059, -998, -946,
+  -901, -861, -827, -797, -772, -750, -730, -713,
+  -699, -687, -677, -668, -662, -657, -654, -652,
+  -652, -654, -657, -662, -668, -677, -687, -699,
+  -713, -730, -750, -772, -797, -827, -861, -901,
+  -946, -998, -1059, -1132, -1219, -1322, -1450, -1608,
+  -1811, -2081, -2444, -2979, -3813, -5323, -8812, -26887
+};
+
+
+/* Tables for quantization of start state */
+
+/* State quantization tables */
+const int16_t WebRtcIlbcfix_kStateSq3[8] = { /* Values in Q13 */
+  -30473, -17838, -9257, -2537,
+  3639, 10893, 19958, 32636
+};
+
+/* This table defines the limits for the selection of the freqg
+   less or equal than value 0 => index = 0
+   less or equal than value k => index = k
+*/
+const int32_t WebRtcIlbcfix_kChooseFrgQuant[64] = {
+  118, 163, 222, 305, 425, 604,
+  851, 1174, 1617, 2222, 3080, 4191,
+  5525, 7215, 9193, 11540, 14397, 17604,
+  21204, 25209, 29863, 35720, 42531, 50375,
+  59162, 68845, 80108, 93754, 110326, 129488,
+  150654, 174328, 201962, 233195, 267843, 308239,
+  354503, 405988, 464251, 531550, 608652, 697516,
+  802526, 928793, 1080145, 1258120, 1481106, 1760881,
+  2111111, 2546619, 3078825, 3748642, 4563142, 5573115,
+  6887601, 8582108, 10797296, 14014513, 18625760, 25529599,
+  37302935, 58819185, 109782723, WEBRTC_SPL_WORD32_MAX
+};
+
+const int16_t WebRtcIlbcfix_kScale[64] = {
+  /* Values in Q16 */
+  29485, 25003, 21345, 18316, 15578, 13128, 10973, 9310, 7955,
+  6762, 5789, 4877, 4255, 3699, 3258, 2904, 2595, 2328,
+  2123, 1932, 1785, 1631, 1493, 1370, 1260, 1167, 1083,
+  /* Values in Q21 */
+  32081, 29611, 27262, 25229, 23432, 21803, 20226, 18883, 17609,
+  16408, 15311, 14327, 13390, 12513, 11693, 10919, 10163, 9435,
+  8739, 8100, 7424, 6813, 6192, 5648, 5122, 4639, 4207, 3798,
+  3404, 3048, 2706, 2348, 2036, 1713, 1393, 1087, 747
+};
+
+/*frgq in fixpoint, but already computed like this:
+  for(i=0; i<64; i++){
+  a = (pow(10,frgq[i])/4.5);
+  WebRtcIlbcfix_kFrgQuantMod[i] = round(a);
+  }
+
+  Value 0 :36 in Q8
+  37:58 in Q5
+  59:63 in Q3
+*/
+const int16_t WebRtcIlbcfix_kFrgQuantMod[64] = {
+  /* First 37 values in Q8 */
+  569, 671, 786, 916, 1077, 1278,
+  1529, 1802, 2109, 2481, 2898, 3440,
+  3943, 4535, 5149, 5778, 6464, 7208,
+  7904, 8682, 9397, 10285, 11240, 12246,
+  13313, 14382, 15492, 16735, 18131, 19693,
+  21280, 22912, 24624, 26544, 28432, 30488,
+  32720,
+  /* 22 values in Q5 */
+  4383, 4684, 5012, 5363, 5739, 6146,
+  6603, 7113, 7679, 8285, 9040, 9850,
+  10838, 11882, 13103, 14467, 15950, 17669,
+  19712, 22016, 24800, 28576,
+  /* 5 values in Q3 */
+  8240, 9792, 12040, 15440, 22472
+};
+
+/* Constants for codebook search and creation */
+
+/* Expansion filter to get additional cb section.
+ * Q12 and reversed compared to flp
+ */
+const int16_t WebRtcIlbcfix_kCbFiltersRev[CB_FILTERLEN]={
+  -140, 446, -755, 3302, 2922, -590, 343, -138};
+
+/* Weighting coefficients for short lags.
+ * [0.2 0.4 0.6 0.8] in Q15 */
+const int16_t WebRtcIlbcfix_kAlpha[4]={
+  6554, 13107, 19661, 26214};
+
+/* Ranges for search and filters at different subframes */
+
+const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={
+  {58,58,58}, {108,44,44}, {108,108,108}, {108,108,108}, {108,108,108}};
+
+const size_t WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147};
+
+/* Gain Quantization for the codebook gains of the 3 stages */
+
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq3[9]={
+  -16384, -10813, -5407, 0, 4096, 8192,
+  12288, 16384, 32767};
+
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq4[17]={
+  -17203, -14746, -12288, -9830, -7373, -4915,
+  -2458, 0, 2458, 4915, 7373, 9830,
+  12288, 14746, 17203, 19661, 32767};
+
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq5[33]={
+  614,        1229,        1843,        2458,        3072,       3686,
+  4301,        4915,        5530,        6144,        6758,        7373,
+  7987,        8602,        9216,        9830,       10445,       11059,
+  11674,       12288,       12902,       13517,       14131,       14746,
+  15360,       15974,       16589,       17203,       17818,       18432,
+  19046,       19661,    32767};
+
+/* Q14 gain_sq5Tbl squared in Q14 */
+const int16_t WebRtcIlbcfix_kGainSq5Sq[32] = {
+  23,   92,    207,  368,  576,  829,
+  1129,  1474,   1866,  2304,  2787,  3317,
+  3893,  4516,   5184,  5897,  6658,  7464,
+  8318,  9216,   10160,  11151,  12187,  13271,
+  14400,  15574,   16796,  18062,  19377,  20736,
+  22140,  23593
+};
+
+const int16_t* const WebRtcIlbcfix_kGain[3] =
+{WebRtcIlbcfix_kGainSq5, WebRtcIlbcfix_kGainSq4, WebRtcIlbcfix_kGainSq3};
+
+
+/* Tables for the Enhancer, using upsamling factor 4 (ENH_UPS0 = 4) */
+
+const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1]={
+  {0,    0,    0, 4096,    0,  0,   0},
+  {64, -315, 1181, 3531, -436, 77, -64},
+  {97, -509, 2464, 2464, -509, 97, -97},
+  {77, -436, 3531, 1181, -315, 64, -77}
+};
+
+const int16_t WebRtcIlbcfix_kEnhWt[3] = {
+  4800, 16384, 27968 /* Q16 */
+};
+
+const size_t WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = {
+  160, 480, 800, 1120, 1440, 1760, 2080, 2400  /* Q(-2) */
+};
+
+/* PLC table */
+
+const int16_t WebRtcIlbcfix_kPlcPerSqr[6] = { /* Grid points for square of periodiciy in Q15 */
+  839, 1343, 2048, 2998, 4247, 5849
+};
+
+const int16_t WebRtcIlbcfix_kPlcPitchFact[6] = { /* Value of y=(x^4-0.4)/(0.7-0.4) in grid points in Q15 */
+  0, 5462, 10922, 16384, 21846, 27306
+};
+
+const int16_t WebRtcIlbcfix_kPlcPfSlope[6] = { /* Slope of y=(x^4-0.4)/(0.7-0.4) in Q11 */
+  26667, 18729, 13653, 10258, 7901, 6214
+};
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.h
new file mode 100644
index 0000000000..a8645c00db
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* high pass filters */
+
+extern const int16_t WebRtcIlbcfix_kHpInCoefs[];
+extern const int16_t WebRtcIlbcfix_kHpOutCoefs[];
+
+/* Window for start state decision */
+extern const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[];
+
+/* low pass filter used for downsampling */
+extern const int16_t WebRtcIlbcfix_kLpFiltCoefs[];
+
+/* LPC analysis and quantization */
+
+extern const int16_t WebRtcIlbcfix_kLpcWin[];
+extern const int16_t WebRtcIlbcfix_kLpcAsymWin[];
+extern const int32_t WebRtcIlbcfix_kLpcLagWin[];
+extern const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[];
+extern const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[];
+extern const int16_t WebRtcIlbcfix_kLsfDimCb[];
+extern const int16_t WebRtcIlbcfix_kLsfSizeCb[];
+extern const int16_t WebRtcIlbcfix_kLsfCb[];
+extern const int16_t WebRtcIlbcfix_kLsfWeight20ms[];
+extern const int16_t WebRtcIlbcfix_kLsfWeight30ms[];
+extern const int16_t WebRtcIlbcfix_kLsfMean[];
+extern const int16_t WebRtcIlbcfix_kLspMean[];
+extern const int16_t WebRtcIlbcfix_kCos[];
+extern const int16_t WebRtcIlbcfix_kCosDerivative[];
+extern const int16_t WebRtcIlbcfix_kCosGrid[];
+extern const int16_t WebRtcIlbcfix_kAcosDerivative[];
+
+/* state quantization tables */
+
+extern const int16_t WebRtcIlbcfix_kStateSq3[];
+extern const int32_t WebRtcIlbcfix_kChooseFrgQuant[];
+extern const int16_t WebRtcIlbcfix_kScale[];
+extern const int16_t WebRtcIlbcfix_kFrgQuantMod[];
+
+/* Ranges for search and filters at different subframes */
+
+extern const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES];
+extern const size_t WebRtcIlbcfix_kFilterRange[];
+
+/* gain quantization tables */
+
+extern const int16_t WebRtcIlbcfix_kGainSq3[];
+extern const int16_t WebRtcIlbcfix_kGainSq4[];
+extern const int16_t WebRtcIlbcfix_kGainSq5[];
+extern const int16_t WebRtcIlbcfix_kGainSq5Sq[];
+extern const int16_t* const WebRtcIlbcfix_kGain[];
+
+/* adaptive codebook definitions */
+
+extern const int16_t WebRtcIlbcfix_kCbFiltersRev[];
+extern const int16_t WebRtcIlbcfix_kAlpha[];
+
+/* enhancer definitions */
+
+extern const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0]
+                                                 [ENH_FLO_MULT2_PLUS1];
+extern const int16_t WebRtcIlbcfix_kEnhWt[];
+extern const size_t WebRtcIlbcfix_kEnhPlocs[];
+
+/* PLC tables */
+
+extern const int16_t WebRtcIlbcfix_kPlcPerSqr[];
+extern const int16_t WebRtcIlbcfix_kPlcPitchFact[];
+extern const int16_t WebRtcIlbcfix_kPlcPfSlope[];
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
new file mode 100644
index 0000000000..7e21faee6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CreateAugmentedVec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "rtc_base/sanitizer.h"
+
+/*----------------------------------------------------------------*
+ *  Recreate a specific codebook vector from the augmented part.
+ *
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CreateAugmentedVec(
+    size_t index,          /* (i) Index for the augmented vector to be
+                              created */
+    const int16_t* buffer, /* (i) Pointer to the end of the codebook memory
+                              that is used for creation of the augmented
+                              codebook */
+    int16_t* cbVec) {      /* (o) The constructed codebook vector */
+  size_t ilow;
+  const int16_t *ppo, *ppi;
+  int16_t cbVecTmp[4];
+  /* Interpolation starts 4 elements before cbVec+index, but must not start
+     outside `cbVec`; clamping interp_len to stay within `cbVec`.
+   */
+  size_t interp_len = WEBRTC_SPL_MIN(index, 4);
+
+  rtc_MsanCheckInitialized(buffer - index - interp_len, sizeof(buffer[0]),
+                           index + interp_len);
+
+  ilow = index - interp_len;
+
+  /* copy the first noninterpolated part */
+  ppo = buffer-index;
+  WEBRTC_SPL_MEMCPY_W16(cbVec, ppo, index);
+
+  /* interpolation */
+  ppo = buffer - interp_len;
+  ppi = buffer - index - interp_len;
+
+  /* perform cbVec[ilow+k] = ((ppi[k]*alphaTbl[k])>>15) +
+                             ((ppo[k]*alphaTbl[interp_len-1-k])>>15);
+     for k = 0..interp_len-1
+  */
+  WebRtcSpl_ElementwiseVectorMult(&cbVec[ilow], ppi, WebRtcIlbcfix_kAlpha,
+                                  interp_len, 15);
+  WebRtcSpl_ReverseOrderMultArrayElements(
+      cbVecTmp, ppo, &WebRtcIlbcfix_kAlpha[interp_len - 1], interp_len, 15);
+  WebRtcSpl_AddVectorsAndShift(&cbVec[ilow], &cbVec[ilow], cbVecTmp, interp_len,
+                               0);
+
+  /* copy the second noninterpolated part */
+  ppo = buffer - index;
+  /* `tempbuff2` is declared in WebRtcIlbcfix_GetCbVec and is SUBL+5 elements
+     long. `buffer` points one element past the end of that vector, i.e., at
+     tempbuff2+SUBL+5. Since ppo=buffer-index, we cannot read any more than
+     `index` elements from `ppo`.
+
+     `cbVec` is declared to be SUBL elements long in WebRtcIlbcfix_CbConstruct.
+     Therefore, we can only write SUBL-index elements to cbVec+index.
+
+     These two conditions limit the number of elements to copy.
+   */
+  WEBRTC_SPL_MEMCPY_W16(cbVec+index, ppo, WEBRTC_SPL_MIN(SUBL-index, index));
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
new file mode 100644
index 0000000000..d7e5be1c2f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CreateAugmentedVec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Recreate a specific codebook vector from the augmented part.
+ *
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CreateAugmentedVec(
+    size_t index, /* (i) Index for the augmented vector to be
+                     created */
+    const int16_t* buffer, /* (i) Pointer to the end of the codebook memory
+                              that is used for creation of the augmented
+                              codebook */
+    int16_t* cbVec);       /* (o) The construced codebook vector */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c
new file mode 100644
index 0000000000..d7621d5b65
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c
@@ -0,0 +1,261 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Decode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/decode_residual.h"
+#include "modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h"
+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+#include "rtc_base/system/arch.h"
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
+#endif
+
+/*----------------------------------------------------------------*
+ *  main decoder function
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_DecodeImpl(
+    int16_t *decblock,    /* (o) decoded signal block */
+    const uint16_t *bytes, /* (i) encoded signal bits */
+    IlbcDecoder *iLBCdec_inst, /* (i/o) the decoder state
+                                           structure */
+    int16_t mode      /* (i) 0: bad packet, PLC,
+                                                                   1: normal */
+                           ) {
+  const int old_mode = iLBCdec_inst->mode;
+  const int old_use_enhancer = iLBCdec_inst->use_enhancer;
+
+  size_t i;
+  int16_t order_plus_one;
+
+  int16_t last_bit;
+  int16_t *data;
+  /* Stack based */
+  int16_t decresidual[BLOCKL_MAX];
+  int16_t PLCresidual[BLOCKL_MAX + LPC_FILTERORDER];
+  int16_t syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
+  int16_t PLClpc[LPC_FILTERORDER + 1];
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+  uint16_t swapped[NO_OF_WORDS_30MS];
+#endif
+  iLBC_bits *iLBCbits_inst = (iLBC_bits*)PLCresidual;
+
+  /* Reuse some buffers that are non overlapping in order to save stack memory */
+  data = &PLCresidual[LPC_FILTERORDER];
+
+  if (mode) { /* the data are good */
+
+    /* decode data */
+
+    /* Unpacketize bits into parameters */
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+    WebRtcIlbcfix_SwapBytes(bytes, iLBCdec_inst->no_of_words, swapped);
+    last_bit = WebRtcIlbcfix_UnpackBits(swapped, iLBCbits_inst, iLBCdec_inst->mode);
+#else
+    last_bit = WebRtcIlbcfix_UnpackBits(bytes, iLBCbits_inst, iLBCdec_inst->mode);
+#endif
+
+    /* Check for bit errors */
+    if (iLBCbits_inst->startIdx<1)
+      mode = 0;
+    if ((iLBCdec_inst->mode==20) && (iLBCbits_inst->startIdx>3))
+      mode = 0;
+    if ((iLBCdec_inst->mode==30) && (iLBCbits_inst->startIdx>5))
+      mode = 0;
+    if (last_bit==1)
+      mode = 0;
+
+    if (mode) { /* No bit errors was detected, continue decoding */
+      /* Stack based */
+      int16_t lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
+      int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+
+      /* adjust index */
+      WebRtcIlbcfix_IndexConvDec(iLBCbits_inst->cb_index);
+
+      /* decode the lsf */
+      WebRtcIlbcfix_SimpleLsfDeQ(lsfdeq, (int16_t*)(iLBCbits_inst->lsf), iLBCdec_inst->lpc_n);
+      WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCdec_inst->lpc_n);
+      WebRtcIlbcfix_DecoderInterpolateLsp(syntdenum, weightdenum,
+                                          lsfdeq, LPC_FILTERORDER, iLBCdec_inst);
+
+      /* Decode the residual using the cb and gain indexes */
+      if (!WebRtcIlbcfix_DecodeResidual(iLBCdec_inst, iLBCbits_inst,
+                                        decresidual, syntdenum))
+        goto error;
+
+      /* preparing the plc for a future loss! */
+      WebRtcIlbcfix_DoThePlc(
+          PLCresidual, PLClpc, 0, decresidual,
+          syntdenum + (LPC_FILTERORDER + 1) * (iLBCdec_inst->nsub - 1),
+          iLBCdec_inst->last_lag, iLBCdec_inst);
+
+      /* Use the output from doThePLC */
+      WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
+    }
+
+  }
+
+  if (mode == 0) {
+    /* the data is bad (either a PLC call
+     * was made or a bit error was detected)
+     */
+
+    /* packet loss conceal */
+
+    WebRtcIlbcfix_DoThePlc(PLCresidual, PLClpc, 1, decresidual, syntdenum,
+                           iLBCdec_inst->last_lag, iLBCdec_inst);
+
+    WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
+
+    order_plus_one = LPC_FILTERORDER + 1;
+
+    for (i = 0; i < iLBCdec_inst->nsub; i++) {
+      WEBRTC_SPL_MEMCPY_W16(syntdenum+(i*order_plus_one),
+                            PLClpc, order_plus_one);
+    }
+  }
+
+  if ((*iLBCdec_inst).use_enhancer == 1) { /* Enhancer activated */
+
+    /* Update the filter and filter coefficients if there was a packet loss */
+    if (iLBCdec_inst->prev_enh_pl==2) {
+      for (i=0;i<iLBCdec_inst->nsub;i++) {
+        WEBRTC_SPL_MEMCPY_W16(&(iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)]),
+                              syntdenum, (LPC_FILTERORDER+1));
+      }
+    }
+
+    /* post filtering */
+    (*iLBCdec_inst).last_lag =
+        WebRtcIlbcfix_EnhancerInterface(data, decresidual, iLBCdec_inst);
+
+    /* synthesis filtering */
+
+    /* Set up the filter state */
+    WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
+
+    if (iLBCdec_inst->mode==20) {
+      /* Enhancer has 40 samples delay */
+      i=0;
+      WebRtcSpl_FilterARFastQ12(
+          data, data,
+          iLBCdec_inst->old_syntdenum + (i+iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1),
+          LPC_FILTERORDER+1, SUBL);
+
+      for (i=1; i < iLBCdec_inst->nsub; i++) {
+        WebRtcSpl_FilterARFastQ12(
+            data+i*SUBL, data+i*SUBL,
+            syntdenum+(i-1)*(LPC_FILTERORDER+1),
+            LPC_FILTERORDER+1, SUBL);
+      }
+
+    } else if (iLBCdec_inst->mode==30) {
+      /* Enhancer has 80 samples delay */
+      for (i=0; i < 2; i++) {
+        WebRtcSpl_FilterARFastQ12(
+            data+i*SUBL, data+i*SUBL,
+            iLBCdec_inst->old_syntdenum + (i+4)*(LPC_FILTERORDER+1),
+            LPC_FILTERORDER+1, SUBL);
+      }
+      for (i=2; i < iLBCdec_inst->nsub; i++) {
+        WebRtcSpl_FilterARFastQ12(
+            data+i*SUBL, data+i*SUBL,
+            syntdenum+(i-2)*(LPC_FILTERORDER+1),
+            LPC_FILTERORDER+1, SUBL);
+      }
+    }
+
+    /* Save the filter state */
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+
+  } else { /* Enhancer not activated */
+    size_t lag;
+
+    /* Find last lag (since the enhancer is not called to give this info) */
+    lag = 20;
+    if (iLBCdec_inst->mode==20) {
+      lag = WebRtcIlbcfix_XcorrCoef(
+          &decresidual[iLBCdec_inst->blockl-60],
+          &decresidual[iLBCdec_inst->blockl-60-lag],
+          60,
+          80, lag, -1);
+    } else {
+      lag = WebRtcIlbcfix_XcorrCoef(
+          &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL],
+          &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL-lag],
+          ENH_BLOCKL,
+          100, lag, -1);
+    }
+
+    /* Store lag (it is needed if next packet is lost) */
+    (*iLBCdec_inst).last_lag = lag;
+
+    /* copy data and run synthesis filter */
+    WEBRTC_SPL_MEMCPY_W16(data, decresidual, iLBCdec_inst->blockl);
+
+    /* Set up the filter state */
+    WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
+
+    for (i=0; i < iLBCdec_inst->nsub; i++) {
+      WebRtcSpl_FilterARFastQ12(
+          data+i*SUBL, data+i*SUBL,
+          syntdenum + i*(LPC_FILTERORDER+1),
+          LPC_FILTERORDER+1, SUBL);
+    }
+
+    /* Save the filter state */
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+  }
+
+  WEBRTC_SPL_MEMCPY_W16(decblock,data,iLBCdec_inst->blockl);
+
+  /* High pass filter the signal (with upscaling a factor 2 and saturation) */
+  WebRtcIlbcfix_HpOutput(decblock, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
+                         iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+                         iLBCdec_inst->blockl);
+
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->old_syntdenum,
+                        syntdenum, iLBCdec_inst->nsub*(LPC_FILTERORDER+1));
+
+  iLBCdec_inst->prev_enh_pl=0;
+
+  if (mode==0) { /* PLC was used */
+    iLBCdec_inst->prev_enh_pl=1;
+  }
+
+  return 0;  // Success.
+
+error:
+  // The decoder got sick from eating that data. Reset it and return.
+  WebRtcIlbcfix_InitDecode(iLBCdec_inst, old_mode, old_use_enhancer);
+  return -1;  // Error
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.h
new file mode 100644
index 0000000000..a7d2910115
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Decode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  main decoder function
+ *---------------------------------------------------------------*/
+
+// Returns 0 on success, -1 on error.
+ABSL_MUST_USE_RESULT
+int WebRtcIlbcfix_DecodeImpl(
+    int16_t* decblock,         /* (o) decoded signal block */
+    const uint16_t* bytes,     /* (i) encoded signal bits */
+    IlbcDecoder* iLBCdec_inst, /* (i/o) the decoder state
+                                           structure */
+    int16_t mode               /* (i) 0: bad packet, PLC,
+                                      1: normal */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
new file mode 100644
index 0000000000..a9668e2889
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecodeResidual.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/decode_residual.h"
+
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+
+/*----------------------------------------------------------------*
+ *  frame residual decoder function (subrutine to iLBC_decode)
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_DecodeResidual(
+    IlbcDecoder *iLBCdec_inst,
+    /* (i/o) the decoder state structure */
+    iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
+                                for the decoding  */
+    int16_t *decresidual,  /* (o) decoded residual frame */
+    int16_t *syntdenum   /* (i) the decoded synthesis filter
+                                  coefficients */
+                                  ) {
+  size_t meml_gotten, diff, start_pos;
+  size_t subcount, subframe;
+  int16_t *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */
+  int16_t *memVec = iLBCdec_inst->prevResidual;  /* Memory for codebook and filter state (reuse memory in state) */
+  int16_t *mem = &memVec[CB_HALFFILTERLEN];   /* Memory for codebook */
+
+  diff = STATE_LEN - iLBCdec_inst->state_short_len;
+
+  if (iLBC_encbits->state_first == 1) {
+    start_pos = (iLBC_encbits->startIdx-1)*SUBL;
+  } else {
+    start_pos = (iLBC_encbits->startIdx-1)*SUBL + diff;
+  }
+
+  /* decode scalar part of start state */
+
+  WebRtcIlbcfix_StateConstruct(iLBC_encbits->idxForMax,
+                               iLBC_encbits->idxVec, &syntdenum[(iLBC_encbits->startIdx-1)*(LPC_FILTERORDER+1)],
+                               &decresidual[start_pos], iLBCdec_inst->state_short_len
+                               );
+
+  if (iLBC_encbits->state_first) { /* put adaptive part in the end */
+
+    /* setup memory */
+
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCdec_inst->state_short_len);
+    WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCdec_inst->state_short_len, decresidual+start_pos,
+                          iLBCdec_inst->state_short_len);
+
+    /* construct decoded vector */
+
+    if (!WebRtcIlbcfix_CbConstruct(
+            &decresidual[start_pos + iLBCdec_inst->state_short_len],
+            iLBC_encbits->cb_index, iLBC_encbits->gain_index,
+            mem + CB_MEML - ST_MEM_L_TBL, ST_MEM_L_TBL, diff))
+      return false;  // Error.
+
+  }
+  else {/* put adaptive part in the beginning */
+
+    /* setup memory */
+
+    meml_gotten = iLBCdec_inst->state_short_len;
+    WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
+                                  decresidual+start_pos, meml_gotten);
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
+
+    /* construct decoded vector */
+
+    if (!WebRtcIlbcfix_CbConstruct(reverseDecresidual, iLBC_encbits->cb_index,
+                                   iLBC_encbits->gain_index,
+                                   mem + CB_MEML - ST_MEM_L_TBL, ST_MEM_L_TBL,
+                                   diff))
+      return false;  // Error.
+
+    /* get decoded residual from reversed vector */
+
+    WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1],
+                                  reverseDecresidual, diff);
+  }
+
+  /* counter for predicted subframes */
+
+  subcount=1;
+
+  /* forward prediction of subframes */
+
+  if (iLBCdec_inst->nsub > iLBC_encbits->startIdx + 1) {
+
+    /* setup memory */
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
+    WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
+                          decresidual+(iLBC_encbits->startIdx-1)*SUBL, STATE_LEN);
+
+    /* loop over subframes to encode */
+
+    size_t Nfor = iLBCdec_inst->nsub - iLBC_encbits->startIdx - 1;
+    for (subframe=0; subframe<Nfor; subframe++) {
+
+      /* construct decoded vector */
+      if (!WebRtcIlbcfix_CbConstruct(
+              &decresidual[(iLBC_encbits->startIdx + 1 + subframe) * SUBL],
+              iLBC_encbits->cb_index + subcount * CB_NSTAGES,
+              iLBC_encbits->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+              SUBL))
+        return false;  // Error;
+
+      /* update memory */
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &decresidual[(iLBC_encbits->startIdx+1+subframe)*SUBL], SUBL);
+
+      subcount++;
+    }
+
+  }
+
+  /* backward prediction of subframes */
+
+  if (iLBC_encbits->startIdx > 1) {
+
+    /* setup memory */
+
+    meml_gotten = SUBL*(iLBCdec_inst->nsub+1-iLBC_encbits->startIdx);
+    if( meml_gotten > CB_MEML ) {
+      meml_gotten=CB_MEML;
+    }
+
+    WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
+                                  decresidual+(iLBC_encbits->startIdx-1)*SUBL, meml_gotten);
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
+
+    /* loop over subframes to decode */
+
+    size_t Nback = iLBC_encbits->startIdx - 1;
+    for (subframe=0; subframe<Nback; subframe++) {
+
+      /* construct decoded vector */
+      if (!WebRtcIlbcfix_CbConstruct(
+              &reverseDecresidual[subframe * SUBL],
+              iLBC_encbits->cb_index + subcount * CB_NSTAGES,
+              iLBC_encbits->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+              SUBL))
+        return false;  // Error.
+
+      /* update memory */
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &reverseDecresidual[subframe*SUBL], SUBL);
+
+      subcount++;
+    }
+
+    /* get decoded residual from reversed vector */
+    WebRtcSpl_MemCpyReversedOrder(decresidual+SUBL*Nback-1,
+                                  reverseDecresidual, SUBL*Nback);
+  }
+
+  return true;  // Success.
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
new file mode 100644
index 0000000000..d079577661
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecodeResidual.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+
+#include <stdbool.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  frame residual decoder function (subrutine to iLBC_decode)
+ *---------------------------------------------------------------*/
+
+// Returns true on success, false on failure. In case of failure, the decoder
+// state may be corrupted and needs resetting.
+ABSL_MUST_USE_RESULT
+bool WebRtcIlbcfix_DecodeResidual(
+    IlbcDecoder* iLBCdec_inst, /* (i/o) the decoder state structure */
+    iLBC_bits* iLBC_encbits,   /* (i/o) Encoded bits, which are used
+                                        for the decoding  */
+    int16_t* decresidual,      /* (o) decoded residual frame */
+    int16_t* syntdenum         /* (i) the decoded synthesis filter
+                                                         coefficients */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
new file mode 100644
index 0000000000..d96bb9b2e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecoderInterpolateLsp.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h"
+
+/*----------------------------------------------------------------*
+ *  obtain synthesis and weighting filters form lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecoderInterpolateLsp(
+    int16_t *syntdenum,  /* (o) synthesis filter coefficients */
+    int16_t *weightdenum, /* (o) weighting denumerator
+                                   coefficients */
+    int16_t *lsfdeq,   /* (i) dequantized lsf coefficients */
+    int16_t length,   /* (i) length of lsf coefficient vector */
+    IlbcDecoder *iLBCdec_inst
+    /* (i) the decoder state structure */
+                                          ){
+  size_t i;
+  int pos, lp_length;
+  int16_t  lp[LPC_FILTERORDER + 1], *lsfdeq2;
+
+  lsfdeq2 = lsfdeq + length;
+  lp_length = length + 1;
+
+  if (iLBCdec_inst->mode==30) {
+    /* subframe 1: Interpolation between old and first LSF */
+
+    WebRtcIlbcfix_LspInterpolate2PolyDec(lp, (*iLBCdec_inst).lsfdeqold, lsfdeq,
+                                         WebRtcIlbcfix_kLsfWeight30ms[0], length);
+    WEBRTC_SPL_MEMCPY_W16(syntdenum,lp,lp_length);
+    WebRtcIlbcfix_BwExpand(weightdenum, lp, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
+
+    /* subframes 2 to 6: interpolation between first and last LSF */
+
+    pos = lp_length;
+    for (i = 1; i < 6; i++) {
+      WebRtcIlbcfix_LspInterpolate2PolyDec(lp, lsfdeq, lsfdeq2,
+                                           WebRtcIlbcfix_kLsfWeight30ms[i], length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum + pos,lp,lp_length);
+      WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
+      pos += lp_length;
+    }
+  } else { /* iLBCdec_inst->mode=20 */
+    /* subframes 1 to 4: interpolation between old and new LSF */
+    pos = 0;
+    for (i = 0; i < iLBCdec_inst->nsub; i++) {
+      WebRtcIlbcfix_LspInterpolate2PolyDec(lp, iLBCdec_inst->lsfdeqold, lsfdeq,
+                                           WebRtcIlbcfix_kLsfWeight20ms[i], length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum+pos,lp,lp_length);
+      WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
+      pos += lp_length;
+    }
+  }
+
+  /* update memory */
+
+  if (iLBCdec_inst->mode==30) {
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq2, length);
+  } else {
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq, length);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
new file mode 100644
index 0000000000..8b08114467
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecoderInterpolateLsp.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  obtain synthesis and weighting filters form lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecoderInterpolateLsp(
+    int16_t* syntdenum, /* (o) synthesis filter coefficients */
+    int16_t* weightdenum, /* (o) weighting denumerator
+                                   coefficients */
+    int16_t* lsfdeq, /* (i) dequantized lsf coefficients */
+    int16_t length,  /* (i) length of lsf coefficient vector */
+    IlbcDecoder* iLBCdec_inst
+    /* (i) the decoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/defines.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/defines.h
new file mode 100644
index 0000000000..64135c4887
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/defines.h
@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ define.h
+
+******************************************************************/
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+
+#include <stdint.h>
+#include <string.h>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+/* general codec settings */
+
+#define FS 8000
+#define BLOCKL_20MS 160
+#define BLOCKL_30MS 240
+#define BLOCKL_MAX 240
+#define NSUB_20MS 4
+#define NSUB_30MS 6
+#define NSUB_MAX 6
+#define NASUB_20MS 2
+#define NASUB_30MS 4
+#define NASUB_MAX 4
+#define SUBL 40
+#define STATE_LEN 80
+#define STATE_SHORT_LEN_30MS 58
+#define STATE_SHORT_LEN_20MS 57
+
+/* LPC settings */
+
+#define LPC_FILTERORDER 10
+#define LPC_LOOKBACK 60
+#define LPC_N_20MS 1
+#define LPC_N_30MS 2
+#define LPC_N_MAX 2
+#define LPC_ASYMDIFF 20
+#define LSF_NSPLIT 3
+#define LSF_NUMBER_OF_STEPS 4
+#define LPC_HALFORDER 5
+#define COS_GRID_POINTS 60
+
+/* cb settings */
+
+#define CB_NSTAGES 3
+#define CB_EXPAND 2
+#define CB_MEML 147
+#define CB_FILTERLEN (2 * 4)
+#define CB_HALFFILTERLEN 4
+#define CB_RESRANGE 34
+#define CB_MAXGAIN_FIXQ6 83 /* error = -0.24% */
+#define CB_MAXGAIN_FIXQ14 21299
+
+/* enhancer */
+
+#define ENH_BLOCKL 80 /* block length */
+#define ENH_BLOCKL_HALF (ENH_BLOCKL / 2)
+#define ENH_HL                                                         \
+  3 /* 2*ENH_HL+1 is number blocks                                     \
+                                                        in said second \
+       sequence */
+#define ENH_SLOP                    \
+  2 /* max difference estimated and \
+                                                       correct pitch period */
+#define ENH_PLOCSL                                                          \
+  8 /* pitch-estimates and                                                  \
+                                                     pitch-locations buffer \
+       length */
+#define ENH_OVERHANG 2
+#define ENH_UPS0 4 /* upsampling rate */
+#define ENH_FL0 3  /* 2*FLO+1 is the length of each filter */
+#define ENH_FLO_MULT2_PLUS1 7
+#define ENH_VECTL (ENH_BLOCKL + 2 * ENH_FL0)
+#define ENH_CORRDIM (2 * ENH_SLOP + 1)
+#define ENH_NBLOCKS (BLOCKL / ENH_BLOCKL)
+#define ENH_NBLOCKS_EXTRA 5
+#define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS+ENH_NBLOCKS_EXTRA */
+#define ENH_BUFL (ENH_NBLOCKS_TOT) * ENH_BLOCKL
+#define ENH_BUFL_FILTEROVERHEAD 3
+#define ENH_A0 819                      /* Q14 */
+#define ENH_A0_MINUS_A0A0DIV4 848256041 /* Q34 */
+#define ENH_A0DIV2 26843546             /* Q30 */
+
+/* PLC */
+
+/* Down sampling */
+
+#define FILTERORDER_DS_PLUS1 7
+#define DELAY_DS 3
+#define FACTOR_DS 2
+
+/* bit stream defs */
+
+#define NO_OF_BYTES_20MS 38
+#define NO_OF_BYTES_30MS 50
+#define NO_OF_WORDS_20MS 19
+#define NO_OF_WORDS_30MS 25
+#define STATE_BITS 3
+#define BYTE_LEN 8
+#define ULP_CLASSES 3
+
+/* help parameters */
+
+#define TWO_PI_FIX 25736 /* Q12 */
+
+/* Constants for codebook search and creation */
+
+#define ST_MEM_L_TBL 85
+#define MEM_LF_TBL 147
+
+/* Struct for the bits */
+typedef struct iLBC_bits_t_ {
+  int16_t lsf[LSF_NSPLIT * LPC_N_MAX];
+  int16_t cb_index[CB_NSTAGES * (NASUB_MAX + 1)];   /* First CB_NSTAGES values
+                                                       contains extra CB index */
+  int16_t gain_index[CB_NSTAGES * (NASUB_MAX + 1)]; /* First CB_NSTAGES values
+                                                       contains extra CB gain */
+  size_t idxForMax;
+  int16_t state_first;
+  int16_t idxVec[STATE_SHORT_LEN_30MS];
+  int16_t firstbits;
+  size_t startIdx;
+} iLBC_bits;
+
+/* type definition encoder instance */
+typedef struct IlbcEncoder_ {
+  /* flag for frame size mode */
+  int16_t mode;
+
+  /* basic parameters for different frame sizes */
+  size_t blockl;
+  size_t nsub;
+  int16_t nasub;
+  size_t no_of_bytes, no_of_words;
+  int16_t lpc_n;
+  size_t state_short_len;
+
+  /* analysis filter state */
+  int16_t anaMem[LPC_FILTERORDER];
+
+  /* Fix-point old lsf parameters for interpolation */
+  int16_t lsfold[LPC_FILTERORDER];
+  int16_t lsfdeqold[LPC_FILTERORDER];
+
+  /* signal buffer for LP analysis */
+  int16_t lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
+
+  /* state of input HP filter */
+  int16_t hpimemx[2];
+  int16_t hpimemy[4];
+
+#ifdef SPLIT_10MS
+  int16_t weightdenumbuf[66];
+  int16_t past_samples[160];
+  uint16_t bytes[25];
+  int16_t section;
+  int16_t Nfor_flag;
+  int16_t Nback_flag;
+  int16_t start_pos;
+  size_t diff;
+#endif
+
+} IlbcEncoder;
+
+/* type definition decoder instance */
+typedef struct IlbcDecoder_ {
+  /* flag for frame size mode */
+  int16_t mode;
+
+  /* basic parameters for different frame sizes */
+  size_t blockl;
+  size_t nsub;
+  int16_t nasub;
+  size_t no_of_bytes, no_of_words;
+  int16_t lpc_n;
+  size_t state_short_len;
+
+  /* synthesis filter state */
+  int16_t syntMem[LPC_FILTERORDER];
+
+  /* old LSF for interpolation */
+  int16_t lsfdeqold[LPC_FILTERORDER];
+
+  /* pitch lag estimated in enhancer and used in PLC */
+  size_t last_lag;
+
+  /* PLC state information */
+  int consPLICount, prev_enh_pl;
+  int16_t perSquare;
+
+  int16_t prevScale, prevPLI;
+  size_t prevLag;
+  int16_t prevLpc[LPC_FILTERORDER + 1];
+  int16_t prevResidual[NSUB_MAX * SUBL];
+  int16_t seed;
+
+  /* previous synthesis filter parameters */
+
+  int16_t old_syntdenum[(LPC_FILTERORDER + 1) * NSUB_MAX];
+
+  /* state of output HP filter */
+  int16_t hpimemx[2];
+  int16_t hpimemy[4];
+
+  /* enhancer state information */
+  int use_enhancer;
+  int16_t enh_buf[ENH_BUFL + ENH_BUFL_FILTEROVERHEAD];
+  size_t enh_period[ENH_NBLOCKS_TOT];
+
+} IlbcDecoder;
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c
new file mode 100644
index 0000000000..9ca6ca48e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c
@@ -0,0 +1,309 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DoThePlc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/comp_corr.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Packet loss concealment routine. Conceals a residual signal
+ *  and LP parameters. If no packet loss, update state.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DoThePlc(
+    int16_t *PLCresidual,  /* (o) concealed residual */
+    int16_t *PLClpc,    /* (o) concealed LP parameters */
+    int16_t PLI,     /* (i) packet loss indicator
+                                                           0 - no PL, 1 = PL */
+    int16_t *decresidual,  /* (i) decoded residual */
+    int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
+    size_t inlag,    /* (i) pitch lag */
+    IlbcDecoder *iLBCdec_inst
+    /* (i/o) decoder instance */
+                            ){
+  size_t i;
+  int32_t cross, ener, cross_comp, ener_comp = 0;
+  int32_t measure, maxMeasure, energy;
+  int32_t noise_energy_threshold_30dB;
+  int16_t max, crossSquareMax, crossSquare;
+  size_t j, lag, randlag;
+  int16_t tmp1, tmp2;
+  int16_t shift1, shift2, shift3, shiftMax;
+  int16_t scale3;
+  size_t corrLen;
+  int32_t tmpW32, tmp2W32;
+  int16_t use_gain;
+  int16_t tot_gain;
+  int16_t max_perSquare;
+  int16_t scale1, scale2;
+  int16_t totscale;
+  int32_t nom;
+  int16_t denom;
+  int16_t pitchfact;
+  size_t use_lag;
+  int ind;
+  int16_t randvec[BLOCKL_MAX];
+
+  /* Packet Loss */
+  if (PLI == 1) {
+
+    (*iLBCdec_inst).consPLICount += 1;
+
+    /* if previous frame not lost,
+       determine pitch pred. gain */
+
+    if (iLBCdec_inst->prevPLI != 1) {
+
+      /* Maximum 60 samples are correlated, preserve as high accuracy
+         as possible without getting overflow */
+      max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual,
+                                     iLBCdec_inst->blockl);
+      scale3 = (WebRtcSpl_GetSizeInBits(max)<<1) - 25;
+      if (scale3 < 0) {
+        scale3 = 0;
+      }
+
+      /* Store scale for use when interpolating between the
+       * concealment and the received packet */
+      iLBCdec_inst->prevScale = scale3;
+
+      /* Search around the previous lag +/-3 to find the
+         best pitch period */
+      lag = inlag - 3;
+
+      /* Guard against getting outside the frame */
+      corrLen = (size_t)WEBRTC_SPL_MIN(60, iLBCdec_inst->blockl-(inlag+3));
+
+      WebRtcIlbcfix_CompCorr( &cross, &ener,
+                              iLBCdec_inst->prevResidual, lag, iLBCdec_inst->blockl, corrLen, scale3);
+
+      /* Normalize and store cross^2 and the number of shifts */
+      shiftMax = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross))-15;
+      crossSquareMax = (int16_t)((
+          (int16_t)WEBRTC_SPL_SHIFT_W32(cross, -shiftMax) *
+          (int16_t)WEBRTC_SPL_SHIFT_W32(cross, -shiftMax)) >> 15);
+
+      for (j=inlag-2;j<=inlag+3;j++) {
+        WebRtcIlbcfix_CompCorr( &cross_comp, &ener_comp,
+                                iLBCdec_inst->prevResidual, j, iLBCdec_inst->blockl, corrLen, scale3);
+
+        /* Use the criteria (corr*corr)/energy to compare if
+           this lag is better or not. To avoid the division,
+           do a cross multiplication */
+        shift1 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross_comp))-15;
+        crossSquare = (int16_t)((
+            (int16_t)WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1) *
+            (int16_t)WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1)) >> 15);
+
+        shift2 = WebRtcSpl_GetSizeInBits(ener)-15;
+        measure = (int16_t)WEBRTC_SPL_SHIFT_W32(ener, -shift2) * crossSquare;
+
+        shift3 = WebRtcSpl_GetSizeInBits(ener_comp)-15;
+        maxMeasure = (int16_t)WEBRTC_SPL_SHIFT_W32(ener_comp, -shift3) *
+            crossSquareMax;
+
+        /* Calculate shift value, so that the two measures can
+           be put in the same Q domain */
+        if(2 * shiftMax + shift3 > 2 * shift1 + shift2) {
+          tmp1 =
+              WEBRTC_SPL_MIN(31, 2 * shiftMax + shift3 - 2 * shift1 - shift2);
+          tmp2 = 0;
+        } else {
+          tmp1 = 0;
+          tmp2 =
+              WEBRTC_SPL_MIN(31, 2 * shift1 + shift2 - 2 * shiftMax - shift3);
+        }
+
+        if ((measure>>tmp1) > (maxMeasure>>tmp2)) {
+          /* New lag is better => record lag, measure and domain */
+          lag = j;
+          crossSquareMax = crossSquare;
+          cross = cross_comp;
+          shiftMax = shift1;
+          ener = ener_comp;
+        }
+      }
+
+      /* Calculate the periodicity for the lag with the maximum correlation.
+
+         Definition of the periodicity:
+         abs(corr(vec1, vec2))/(sqrt(energy(vec1))*sqrt(energy(vec2)))
+
+         Work in the Square domain to simplify the calculations
+         max_perSquare is less than 1 (in Q15)
+      */
+      tmp2W32=WebRtcSpl_DotProductWithScale(&iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
+                                            &iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
+                                            corrLen, scale3);
+
+      if ((tmp2W32>0)&&(ener_comp>0)) {
+        /* norm energies to int16_t, compute the product of the energies and
+           use the upper int16_t as the denominator */
+
+        scale1=(int16_t)WebRtcSpl_NormW32(tmp2W32)-16;
+        tmp1=(int16_t)WEBRTC_SPL_SHIFT_W32(tmp2W32, scale1);
+
+        scale2=(int16_t)WebRtcSpl_NormW32(ener)-16;
+        tmp2=(int16_t)WEBRTC_SPL_SHIFT_W32(ener, scale2);
+        denom = (int16_t)((tmp1 * tmp2) >> 16);  /* in Q(scale1+scale2-16) */
+
+        /* Square the cross correlation and norm it such that max_perSquare
+           will be in Q15 after the division */
+
+        totscale = scale1+scale2-1;
+        tmp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, (totscale>>1));
+        tmp2 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, totscale-(totscale>>1));
+
+        nom = tmp1 * tmp2;
+        max_perSquare = (int16_t)WebRtcSpl_DivW32W16(nom, denom);
+
+      } else {
+        max_perSquare = 0;
+      }
+    }
+
+    /* previous frame lost, use recorded lag and gain */
+
+    else {
+      lag = iLBCdec_inst->prevLag;
+      max_perSquare = iLBCdec_inst->perSquare;
+    }
+
+    /* Attenuate signal and scale down pitch pred gain if
+       several frames lost consecutively */
+
+    use_gain = 32767;   /* 1.0 in Q15 */
+
+    if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>320) {
+      use_gain = 29491;  /* 0.9 in Q15 */
+    } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>640) {
+      use_gain = 22938;  /* 0.7 in Q15 */
+    } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>960) {
+      use_gain = 16384;  /* 0.5 in Q15 */
+    } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>1280) {
+      use_gain = 0;   /* 0.0 in Q15 */
+    }
+
+    /* Compute mixing factor of picth repeatition and noise:
+       for max_per>0.7 set periodicity to 1.0
+       0.4<max_per<0.7 set periodicity to (maxper-0.4)/0.7-0.4)
+       max_per<0.4 set periodicity to 0.0
+    */
+
+    if (max_perSquare>7868) { /* periodicity > 0.7  (0.7^4=0.2401 in Q15) */
+      pitchfact = 32767;
+    } else if (max_perSquare>839) { /* 0.4 < periodicity < 0.7 (0.4^4=0.0256 in Q15) */
+      /* find best index and interpolate from that */
+      ind = 5;
+      while ((max_perSquare<WebRtcIlbcfix_kPlcPerSqr[ind])&&(ind>0)) {
+        ind--;
+      }
+      /* pitch fact is approximated by first order */
+      tmpW32 = (int32_t)WebRtcIlbcfix_kPlcPitchFact[ind] +
+          ((WebRtcIlbcfix_kPlcPfSlope[ind] *
+              (max_perSquare - WebRtcIlbcfix_kPlcPerSqr[ind])) >> 11);
+
+      pitchfact = (int16_t)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */
+
+    } else { /* periodicity < 0.4 */
+      pitchfact = 0;
+    }
+
+    /* avoid repetition of same pitch cycle (buzzyness) */
+    use_lag = lag;
+    if (lag<80) {
+      use_lag = 2*lag;
+    }
+
+    /* compute concealed residual */
+    noise_energy_threshold_30dB = (int32_t)iLBCdec_inst->blockl * 900;
+    energy = 0;
+    for (i=0; i<iLBCdec_inst->blockl; i++) {
+
+      /* noise component -  52 < randlagFIX < 117 */
+      iLBCdec_inst->seed = (int16_t)(iLBCdec_inst->seed * 31821 + 13849);
+      randlag = 53 + (iLBCdec_inst->seed & 63);
+      if (randlag > i) {
+        randvec[i] =
+            iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - randlag];
+      } else {
+        randvec[i] = iLBCdec_inst->prevResidual[i - randlag];
+      }
+
+      /* pitch repeatition component */
+      if (use_lag > i) {
+        PLCresidual[i] =
+            iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - use_lag];
+      } else {
+        PLCresidual[i] = PLCresidual[i - use_lag];
+      }
+
+      /* Attinuate total gain for each 10 ms */
+      if (i<80) {
+        tot_gain=use_gain;
+      } else if (i<160) {
+        tot_gain = (int16_t)((31130 * use_gain) >> 15);  /* 0.95*use_gain */
+      } else {
+        tot_gain = (int16_t)((29491 * use_gain) >> 15);  /* 0.9*use_gain */
+      }
+
+
+      /* mix noise and pitch repeatition */
+      PLCresidual[i] = (int16_t)((tot_gain *
+          ((pitchfact * PLCresidual[i] + (32767 - pitchfact) * randvec[i] +
+              16384) >> 15)) >> 15);
+
+      /* Compute energy until threshold for noise energy is reached */
+      if (energy < noise_energy_threshold_30dB) {
+        energy += PLCresidual[i] * PLCresidual[i];
+      }
+    }
+
+    /* less than 30 dB, use only noise */
+    if (energy < noise_energy_threshold_30dB) {
+      for (i=0; i<iLBCdec_inst->blockl; i++) {
+        PLCresidual[i] = randvec[i];
+      }
+    }
+
+    /* use the old LPC */
+    WEBRTC_SPL_MEMCPY_W16(PLClpc, (*iLBCdec_inst).prevLpc, LPC_FILTERORDER+1);
+
+    /* Update state in case there are multiple frame losses */
+    iLBCdec_inst->prevLag = lag;
+    iLBCdec_inst->perSquare = max_perSquare;
+  }
+
+  /* no packet loss, copy input */
+
+  else {
+    WEBRTC_SPL_MEMCPY_W16(PLCresidual, decresidual, iLBCdec_inst->blockl);
+    WEBRTC_SPL_MEMCPY_W16(PLClpc, lpc, (LPC_FILTERORDER+1));
+    iLBCdec_inst->consPLICount = 0;
+  }
+
+  /* update state */
+  iLBCdec_inst->prevPLI = PLI;
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevLpc, PLClpc, (LPC_FILTERORDER+1));
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevResidual, PLCresidual, iLBCdec_inst->blockl);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.h
new file mode 100644
index 0000000000..c19c4eca32
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DoThePlc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Packet loss concealment routine. Conceals a residual signal
+ *  and LP parameters. If no packet loss, update state.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DoThePlc(
+    int16_t* PLCresidual, /* (o) concealed residual */
+    int16_t* PLClpc,      /* (o) concealed LP parameters */
+    int16_t PLI,          /* (i) packet loss indicator
+                                                                0 - no PL, 1 = PL */
+    int16_t* decresidual, /* (i) decoded residual */
+    int16_t* lpc,         /* (i) decoded LPC (only used for no PL) */
+    size_t inlag,         /* (i) pitch lag */
+    IlbcDecoder* iLBCdec_inst
+    /* (i/o) decoder instance */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c
new file mode 100644
index 0000000000..8e536221cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c
@@ -0,0 +1,517 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Encode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+#include "modules/audio_coding/codecs/ilbc/cb_search.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/frame_classify.h"
+#include "modules/audio_coding/codecs/ilbc/hp_input.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_enc.h"
+#include "modules/audio_coding/codecs/ilbc/lpc_encode.h"
+#include "modules/audio_coding/codecs/ilbc/pack_bits.h"
+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
+#include "modules/audio_coding/codecs/ilbc/state_search.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/arch.h"
+
+#ifdef SPLIT_10MS
+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+#endif
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
+#endif
+
+/*----------------------------------------------------------------*
+ *  main encoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EncodeImpl(
+    uint16_t *bytes,     /* (o) encoded data bits iLBC */
+    const int16_t *block, /* (i) speech vector to encode */
+    IlbcEncoder *iLBCenc_inst /* (i/o) the general encoder
+                                     state */
+                          ){
+  size_t n, meml_gotten, Nfor;
+  size_t diff, start_pos;
+  size_t index;
+  size_t subcount, subframe;
+  size_t start_count, end_count;
+  int16_t *residual;
+  int32_t en1, en2;
+  int16_t scale, max;
+  int16_t *syntdenum;
+  int16_t *decresidual;
+  int16_t *reverseResidual;
+  int16_t *reverseDecresidual;
+  /* Stack based */
+  int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+  int16_t dataVec[BLOCKL_MAX + LPC_FILTERORDER];
+  int16_t memVec[CB_MEML+CB_FILTERLEN];
+  int16_t bitsMemory[sizeof(iLBC_bits)/sizeof(int16_t)];
+  iLBC_bits *iLBCbits_inst = (iLBC_bits*)bitsMemory;
+
+
+#ifdef SPLIT_10MS
+  int16_t *weightdenumbuf = iLBCenc_inst->weightdenumbuf;
+  int16_t last_bit;
+#endif
+
+  int16_t *data = &dataVec[LPC_FILTERORDER];
+  int16_t *mem = &memVec[CB_HALFFILTERLEN];
+
+  /* Reuse som buffers to save stack memory */
+  residual = &iLBCenc_inst->lpc_buffer[LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl];
+  syntdenum = mem;      /* syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX] and mem are used non overlapping in the code */
+  decresidual = residual;     /* Already encoded residual is overwritten by the decoded version */
+  reverseResidual = data;     /* data and reverseResidual are used non overlapping in the code */
+  reverseDecresidual = reverseResidual; /* Already encoded residual is overwritten by the decoded version */
+
+#ifdef SPLIT_10MS
+
+  WebRtcSpl_MemSetW16 (  (int16_t *) iLBCbits_inst, 0,
+                         sizeof(iLBC_bits) / sizeof(int16_t)  );
+
+  start_pos = iLBCenc_inst->start_pos;
+  diff = iLBCenc_inst->diff;
+
+  if (iLBCenc_inst->section != 0){
+    WEBRTC_SPL_MEMCPY_W16 (weightdenum, weightdenumbuf,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+    /* Un-Packetize the frame into parameters */
+    last_bit = WebRtcIlbcfix_UnpackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+    if (last_bit)
+      return;
+    /* adjust index */
+    WebRtcIlbcfix_IndexConvDec (iLBCbits_inst->cb_index);
+
+    if (iLBCenc_inst->section == 1){
+      /* Save first 80 samples of a 160/240 sample frame for 20/30msec */
+      WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples, block, 80);
+    }
+    else{ // iLBCenc_inst->section == 2 AND mode = 30ms
+      /* Save second 80 samples of a 240 sample frame for 30msec */
+      WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples + 80, block, 80);
+    }
+  }
+  else{ // iLBCenc_inst->section == 0
+    /* form a complete frame of 160/240 for 20msec/30msec mode */
+    WEBRTC_SPL_MEMCPY_W16 (data + (iLBCenc_inst->mode * 8) - 80, block, 80);
+    WEBRTC_SPL_MEMCPY_W16 (data, iLBCenc_inst->past_samples,
+                           (iLBCenc_inst->mode * 8) - 80);
+    iLBCenc_inst->Nfor_flag = 0;
+    iLBCenc_inst->Nback_flag = 0;
+#else
+    /* copy input block to data*/
+    WEBRTC_SPL_MEMCPY_W16(data,block,iLBCenc_inst->blockl);
+#endif
+
+    /* high pass filtering of input signal and scale down the residual (*0.5) */
+    WebRtcIlbcfix_HpInput(data, (int16_t*)WebRtcIlbcfix_kHpInCoefs,
+                          iLBCenc_inst->hpimemy, iLBCenc_inst->hpimemx,
+                          iLBCenc_inst->blockl);
+
+    /* LPC of hp filtered input data */
+    WebRtcIlbcfix_LpcEncode(syntdenum, weightdenum, iLBCbits_inst->lsf, data,
+                            iLBCenc_inst);
+
+    /* Set up state */
+    WEBRTC_SPL_MEMCPY_W16(dataVec, iLBCenc_inst->anaMem, LPC_FILTERORDER);
+
+    /* inverse filter to get residual */
+    for (n=0; n<iLBCenc_inst->nsub; n++ ) {
+      WebRtcSpl_FilterMAFastQ12(
+          &data[n*SUBL], &residual[n*SUBL],
+          &syntdenum[n*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, SUBL);
+    }
+
+    /* Copy the state for next frame */
+    WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->anaMem, &data[iLBCenc_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+
+    /* find state location */
+
+    iLBCbits_inst->startIdx = WebRtcIlbcfix_FrameClassify(iLBCenc_inst,residual);
+
+    /* check if state should be in first or last part of the
+       two subframes */
+
+    index = (iLBCbits_inst->startIdx-1)*SUBL;
+    max=WebRtcSpl_MaxAbsValueW16(&residual[index], 2*SUBL);
+    scale = WebRtcSpl_GetSizeInBits((uint32_t)(max * max));
+
+    /* Scale to maximum 25 bits so that the MAC won't cause overflow */
+    scale = scale - 25;
+    if(scale < 0) {
+      scale = 0;
+    }
+
+    diff = STATE_LEN - iLBCenc_inst->state_short_len;
+    en1=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
+                                      iLBCenc_inst->state_short_len, scale);
+    index += diff;
+    en2=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
+                                      iLBCenc_inst->state_short_len, scale);
+    if (en1 > en2) {
+      iLBCbits_inst->state_first = 1;
+      start_pos = (iLBCbits_inst->startIdx-1)*SUBL;
+    } else {
+      iLBCbits_inst->state_first = 0;
+      start_pos = (iLBCbits_inst->startIdx-1)*SUBL + diff;
+    }
+
+    /* scalar quantization of state */
+
+    WebRtcIlbcfix_StateSearch(iLBCenc_inst, iLBCbits_inst, &residual[start_pos],
+                              &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+                              &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)]);
+
+    WebRtcIlbcfix_StateConstruct(iLBCbits_inst->idxForMax, iLBCbits_inst->idxVec,
+                                 &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+                                 &decresidual[start_pos], iLBCenc_inst->state_short_len
+                                 );
+
+    /* predictive quantization in state */
+
+    if (iLBCbits_inst->state_first) { /* put adaptive part in the end */
+
+      /* setup memory */
+
+      WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCenc_inst->state_short_len,
+                            decresidual+start_pos, iLBCenc_inst->state_short_len);
+
+      /* encode subframes */
+
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+                             &residual[start_pos+iLBCenc_inst->state_short_len],
+                             mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
+                             &weightdenum[iLBCbits_inst->startIdx*(LPC_FILTERORDER+1)], 0);
+
+      /* construct decoded vector */
+
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+          &decresidual[start_pos + iLBCenc_inst->state_short_len],
+          iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+          mem + CB_MEML - ST_MEM_L_TBL, ST_MEM_L_TBL, diff));
+
+    }
+    else { /* put adaptive part in the beginning */
+
+      /* create reversed vectors for prediction */
+
+      WebRtcSpl_MemCpyReversedOrder(&reverseResidual[diff-1],
+                                    &residual[(iLBCbits_inst->startIdx+1)*SUBL-STATE_LEN], diff);
+
+      /* setup memory */
+
+      meml_gotten = iLBCenc_inst->state_short_len;
+      WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[start_pos], meml_gotten);
+      WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
+
+      /* encode subframes */
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+                             reverseResidual, mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
+                             &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+                             0);
+
+      /* construct decoded vector */
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+            reverseDecresidual, iLBCbits_inst->cb_index,
+            iLBCbits_inst->gain_index, mem + CB_MEML - ST_MEM_L_TBL,
+            ST_MEM_L_TBL, diff));
+
+      /* get decoded residual from reversed vector */
+
+      WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1], reverseDecresidual, diff);
+    }
+
+#ifdef SPLIT_10MS
+    iLBCenc_inst->start_pos = start_pos;
+    iLBCenc_inst->diff = diff;
+    iLBCenc_inst->section++;
+    /* adjust index */
+    WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
+    /* Packetize the parameters into the frame */
+    WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+    WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+    return;
+  }
+#endif
+
+  /* forward prediction of subframes */
+
+  Nfor = iLBCenc_inst->nsub-iLBCbits_inst->startIdx-1;
+
+  /* counter for predicted subframes */
+#ifdef SPLIT_10MS
+  if (iLBCenc_inst->mode == 20)
+  {
+    subcount = 1;
+  }
+  if (iLBCenc_inst->mode == 30)
+  {
+    if (iLBCenc_inst->section == 1)
+    {
+      subcount = 1;
+    }
+    if (iLBCenc_inst->section == 2)
+    {
+      subcount = 3;
+    }
+  }
+#else
+  subcount=1;
+#endif
+
+  if( Nfor > 0 ){
+
+    /* setup memory */
+
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
+    WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
+                          decresidual+(iLBCbits_inst->startIdx-1)*SUBL, STATE_LEN);
+
+#ifdef SPLIT_10MS
+    if (iLBCenc_inst->Nfor_flag > 0)
+    {
+      for (subframe = 0; subframe < WEBRTC_SPL_MIN (Nfor, 2); subframe++)
+      {
+        /* update memory */
+        WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
+        WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
+                               &decresidual[(iLBCbits_inst->startIdx + 1 +
+                                             subframe) * SUBL], SUBL);
+      }
+    }
+
+    iLBCenc_inst->Nfor_flag++;
+
+    if (iLBCenc_inst->mode == 20)
+    {
+      start_count = 0;
+      end_count = Nfor;
+    }
+    if (iLBCenc_inst->mode == 30)
+    {
+      if (iLBCenc_inst->section == 1)
+      {
+        start_count = 0;
+        end_count = WEBRTC_SPL_MIN (Nfor, (size_t)2);
+      }
+      if (iLBCenc_inst->section == 2)
+      {
+        start_count = WEBRTC_SPL_MIN (Nfor, (size_t)2);
+        end_count = Nfor;
+      }
+    }
+#else
+    start_count = 0;
+    end_count = Nfor;
+#endif
+
+    /* loop over subframes to encode */
+
+    for (subframe = start_count; subframe < end_count; subframe++){
+
+      /* encode subframe */
+
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+                             iLBCbits_inst->gain_index+subcount*CB_NSTAGES,
+                             &residual[(iLBCbits_inst->startIdx+1+subframe)*SUBL],
+                             mem, MEM_LF_TBL, SUBL,
+                             &weightdenum[(iLBCbits_inst->startIdx+1+subframe)*(LPC_FILTERORDER+1)],
+                             subcount);
+
+      /* construct decoded vector */
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+            &decresidual[(iLBCbits_inst->startIdx + 1 + subframe) * SUBL],
+            iLBCbits_inst->cb_index + subcount * CB_NSTAGES,
+            iLBCbits_inst->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+            SUBL));
+
+      /* update memory */
+
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &decresidual[(iLBCbits_inst->startIdx+1+subframe)*SUBL], SUBL);
+
+      subcount++;
+    }
+  }
+
+#ifdef SPLIT_10MS
+  if ((iLBCenc_inst->section == 1) &&
+      (iLBCenc_inst->mode == 30) && (Nfor > 0) && (end_count == 2))
+  {
+    iLBCenc_inst->section++;
+    /* adjust index */
+    WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
+    /* Packetize the parameters into the frame */
+    WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+    WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+    return;
+  }
+#endif
+
+  /* backward prediction of subframes */
+
+  if (iLBCbits_inst->startIdx > 1) {
+
+    /* create reverse order vectors
+       (The decresidual does not need to be copied since it is
+       contained in the same vector as the residual)
+    */
+
+    size_t Nback = iLBCbits_inst->startIdx - 1;
+    WebRtcSpl_MemCpyReversedOrder(&reverseResidual[Nback*SUBL-1], residual, Nback*SUBL);
+
+    /* setup memory */
+
+    meml_gotten = SUBL*(iLBCenc_inst->nsub+1-iLBCbits_inst->startIdx);
+    if( meml_gotten > CB_MEML ) {
+      meml_gotten=CB_MEML;
+    }
+
+    WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[Nback*SUBL], meml_gotten);
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
+
+#ifdef SPLIT_10MS
+    if (iLBCenc_inst->Nback_flag > 0)
+    {
+      for (subframe = 0; subframe < WEBRTC_SPL_MAX (2 - Nfor, 0); subframe++)
+      {
+        /* update memory */
+        WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
+        WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
+                               &reverseDecresidual[subframe * SUBL], SUBL);
+      }
+    }
+
+    iLBCenc_inst->Nback_flag++;
+
+
+    if (iLBCenc_inst->mode == 20)
+    {
+      start_count = 0;
+      end_count = Nback;
+    }
+    if (iLBCenc_inst->mode == 30)
+    {
+      if (iLBCenc_inst->section == 1)
+      {
+        start_count = 0;
+        end_count = (Nfor >= 2) ? 0 : (2 - NFor);
+      }
+      if (iLBCenc_inst->section == 2)
+      {
+        start_count = (Nfor >= 2) ? 0 : (2 - NFor);
+        end_count = Nback;
+      }
+    }
+#else
+    start_count = 0;
+    end_count = Nback;
+#endif
+
+    /* loop over subframes to encode */
+
+    for (subframe = start_count; subframe < end_count; subframe++){
+
+      /* encode subframe */
+
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+                             iLBCbits_inst->gain_index+subcount*CB_NSTAGES, &reverseResidual[subframe*SUBL],
+                             mem, MEM_LF_TBL, SUBL,
+                             &weightdenum[(iLBCbits_inst->startIdx-2-subframe)*(LPC_FILTERORDER+1)],
+                             subcount);
+
+      /* construct decoded vector */
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+            &reverseDecresidual[subframe * SUBL],
+            iLBCbits_inst->cb_index + subcount * CB_NSTAGES,
+            iLBCbits_inst->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+            SUBL));
+
+      /* update memory */
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &reverseDecresidual[subframe*SUBL], SUBL);
+
+      subcount++;
+
+    }
+
+    /* get decoded residual from reversed vector */
+
+    WebRtcSpl_MemCpyReversedOrder(&decresidual[SUBL*Nback-1], reverseDecresidual, SUBL*Nback);
+  }
+  /* end encoding part */
+
+  /* adjust index */
+
+  WebRtcIlbcfix_IndexConvEnc(iLBCbits_inst->cb_index);
+
+  /* Packetize the parameters into the frame */
+
+#ifdef SPLIT_10MS
+  if( (iLBCenc_inst->mode==30) && (iLBCenc_inst->section==1) ){
+    WebRtcIlbcfix_PackBits(iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+  }
+  else{
+    WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
+  }
+#else
+  WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
+#endif
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+  /* Swap bytes for LITTLE ENDIAN since the packbits()
+     function assumes BIG_ENDIAN machine */
+#ifdef SPLIT_10MS
+  if (( (iLBCenc_inst->section == 1) && (iLBCenc_inst->mode == 20) ) ||
+      ( (iLBCenc_inst->section == 2) && (iLBCenc_inst->mode == 30) )){
+    WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words, bytes);
+  }
+#else
+  WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words, bytes);
+#endif
+#endif
+
+#ifdef SPLIT_10MS
+  if (subcount == (iLBCenc_inst->nsub - 1))
+  {
+    iLBCenc_inst->section = 0;
+  }
+  else
+  {
+    iLBCenc_inst->section++;
+    WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+  }
+#endif
+
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.h
new file mode 100644
index 0000000000..bc3e187d92
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Encode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  main encoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EncodeImpl(
+    uint16_t* bytes,      /* (o) encoded data bits iLBC */
+    const int16_t* block, /* (i) speech vector to encode */
+    IlbcEncoder* iLBCenc_inst /* (i/o) the general encoder
+                                           state */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
new file mode 100644
index 0000000000..7f00254aea
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnergyInverse.c
+
+******************************************************************/
+
+/* Inverses the in vector in into Q29 domain */
+
+#include "modules/audio_coding/codecs/ilbc/energy_inverse.h"
+
+void WebRtcIlbcfix_EnergyInverse(
+    int16_t *energy,    /* (i/o) Energy and inverse
+                                                           energy (in Q29) */
+    size_t noOfEnergies)  /* (i)   The length of the energy
+                                   vector */
+{
+  int32_t Nom=(int32_t)0x1FFFFFFF;
+  int16_t *energyPtr;
+  size_t i;
+
+  /* Set the minimum energy value to 16384 to avoid overflow */
+  energyPtr=energy;
+  for (i=0; i<noOfEnergies; i++) {
+    (*energyPtr)=WEBRTC_SPL_MAX((*energyPtr),16384);
+    energyPtr++;
+  }
+
+  /* Calculate inverse energy in Q29 */
+  energyPtr=energy;
+  for (i=0; i<noOfEnergies; i++) {
+    (*energyPtr) = (int16_t)WebRtcSpl_DivW32W16(Nom, (*energyPtr));
+    energyPtr++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
new file mode 100644
index 0000000000..15391cf230
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnergyInverse.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* Inverses the in vector in into Q29 domain */
+
+void WebRtcIlbcfix_EnergyInverse(
+    int16_t*
+        energy, /* (i/o) Energy and inverse
+                                                          energy (in Q29) */
+    size_t noOfEnergies); /* (i)   The length of the energy
+                                 vector */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c
new file mode 100644
index 0000000000..cd3d0a4db1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhUpsample.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/enh_upsample.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * upsample finite array assuming zeros outside bounds
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EnhUpsample(
+    int32_t *useq1, /* (o) upsampled output sequence */
+    int16_t *seq1 /* (i) unupsampled sequence */
+                                ){
+  int j;
+  int32_t *pu1, *pu11;
+  int16_t *ps, *w16tmp;
+  const int16_t *pp;
+
+  /* filtering: filter overhangs left side of sequence */
+  pu1=useq1;
+  for (j=0;j<ENH_UPS0; j++) {
+    pu11=pu1;
+    /* i = 2 */
+    pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+    ps=seq1+2;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+    /* i = 3 */
+    pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+    ps=seq1+3;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+    /* i = 4 */
+    pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+    ps=seq1+4;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu1++;
+  }
+
+  /* filtering: simple convolution=inner products
+     (not needed since the sequence is so short)
+  */
+
+  /* filtering: filter overhangs right side of sequence */
+
+  /* Code with loops, which is equivivalent to the expanded version below
+
+     filterlength = 5;
+     hf1 = 2;
+     for(j=0;j<ENH_UPS0; j++){
+     pu = useq1 + (filterlength-hfl)*ENH_UPS0 + j;
+     for(i=1; i<=hfl; i++){
+     *pu=0;
+     pp = polyp[j]+i;
+     ps = seq1+dim1-1;
+     for(k=0;k<filterlength-i;k++) {
+     *pu += (*ps--) * *pp++;
+     }
+     pu+=ENH_UPS0;
+     }
+     }
+  */
+  pu1 = useq1 + 12;
+  w16tmp = seq1+4;
+  for (j=0;j<ENH_UPS0; j++) {
+    pu11 = pu1;
+    /* i = 1 */
+    pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+2;
+    ps = w16tmp;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+    /* i = 2 */
+    pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+3;
+    ps = w16tmp;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+
+    pu1++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h
new file mode 100644
index 0000000000..b427eca50a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhUpsample.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * upsample finite array assuming zeros outside bounds
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EnhUpsample(
+    int32_t* useq1, /* (o) upsampled output sequence */
+    int16_t* seq1   /* (i) unupsampled sequence */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c
new file mode 100644
index 0000000000..bd4e60015c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Enhancer.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/enhancer.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/get_sync_seq.h"
+#include "modules/audio_coding/codecs/ilbc/smooth.h"
+
+/*----------------------------------------------------------------*
+ * perform enhancement on idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Enhancer(
+    int16_t *odata,   /* (o) smoothed block, dimension blockl */
+    int16_t *idata,   /* (i) data buffer used for enhancing */
+    size_t idatal,   /* (i) dimension idata */
+    size_t centerStartPos, /* (i) first sample current block within idata */
+    size_t *period,   /* (i) pitch period array (pitch bward-in time) */
+    const size_t *plocs,   /* (i) locations where period array values valid */
+    size_t periodl   /* (i) dimension of period and plocs */
+                            ){
+  /* Stack based */
+  int16_t surround[ENH_BLOCKL];
+
+  WebRtcSpl_MemSetW16(surround, 0, ENH_BLOCKL);
+
+  /* get said second sequence of segments */
+
+  WebRtcIlbcfix_GetSyncSeq(idata, idatal, centerStartPos, period, plocs,
+                           periodl, ENH_HL, surround);
+
+  /* compute the smoothed output from said second sequence */
+
+  WebRtcIlbcfix_Smooth(odata, idata + centerStartPos, surround);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.h
new file mode 100644
index 0000000000..386949347a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Enhancer.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * perform enhancement on idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Enhancer(
+    int16_t* odata,        /* (o) smoothed block, dimension blockl */
+    int16_t* idata,        /* (i) data buffer used for enhancing */
+    size_t idatal,         /* (i) dimension idata */
+    size_t centerStartPos, /* (i) first sample current block within idata */
+    size_t* period,        /* (i) pitch period array (pitch bward-in time) */
+    const size_t* plocs,   /* (i) locations where period array values valid */
+    size_t periodl         /* (i) dimension of period and plocs */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
new file mode 100644
index 0000000000..ca23e19ae3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -0,0 +1,382 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhancerInterface.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/enhancer.h"
+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+
+
+
+/*----------------------------------------------------------------*
+ * interface for enhancer
+ *---------------------------------------------------------------*/
+
+size_t  // (o) Estimated lag in end of in[]
+    WebRtcIlbcfix_EnhancerInterface(
+        int16_t* out,                 // (o) enhanced signal
+        const int16_t* in,            // (i) unenhanced signal
+        IlbcDecoder* iLBCdec_inst) {  // (i) buffers etc
+  size_t iblock;
+  size_t lag=20, tlag=20;
+  size_t inLen=iLBCdec_inst->blockl+120;
+  int16_t scale, scale1;
+  size_t plc_blockl;
+  int16_t *enh_buf;
+  size_t *enh_period;
+  int32_t tmp1, tmp2, max;
+  size_t new_blocks;
+  int16_t *enh_bufPtr1;
+  size_t i;
+  size_t k;
+  int16_t EnChange;
+  int16_t SqrtEnChange;
+  int16_t inc;
+  int16_t win;
+  int16_t *tmpW16ptr;
+  size_t startPos;
+  int16_t *plc_pred;
+  const int16_t *target, *regressor;
+  int16_t max16;
+  int shifts;
+  int32_t ener;
+  int16_t enerSh;
+  int16_t corrSh;
+  size_t ind;
+  int16_t sh;
+  size_t start, stop;
+  /* Stack based */
+  int16_t totsh[3];
+  int16_t downsampled[(BLOCKL_MAX+120)>>1]; /* length 180 */
+  int32_t corr32[50];
+  int32_t corrmax[3];
+  int16_t corr16[3];
+  int16_t en16[3];
+  size_t lagmax[3];
+
+  plc_pred = downsampled; /* Reuse memory since plc_pred[ENH_BLOCKL] and
+                              downsampled are non overlapping */
+  enh_buf=iLBCdec_inst->enh_buf;
+  enh_period=iLBCdec_inst->enh_period;
+
+  /* Copy in the new data into the enhancer buffer */
+  memmove(enh_buf, &enh_buf[iLBCdec_inst->blockl],
+          (ENH_BUFL - iLBCdec_inst->blockl) * sizeof(*enh_buf));
+
+  WEBRTC_SPL_MEMCPY_W16(&enh_buf[ENH_BUFL-iLBCdec_inst->blockl], in,
+                        iLBCdec_inst->blockl);
+
+  /* Set variables that are dependent on frame size */
+  if (iLBCdec_inst->mode==30) {
+    plc_blockl=ENH_BLOCKL;
+    new_blocks=3;
+    startPos=320;  /* Start position for enhancement
+                     (640-new_blocks*ENH_BLOCKL-80) */
+  } else {
+    plc_blockl=40;
+    new_blocks=2;
+    startPos=440;  /* Start position for enhancement
+                    (640-new_blocks*ENH_BLOCKL-40) */
+  }
+
+  /* Update the pitch prediction for each enhancer block, move the old ones */
+  memmove(enh_period, &enh_period[new_blocks],
+          (ENH_NBLOCKS_TOT - new_blocks) * sizeof(*enh_period));
+
+  WebRtcSpl_DownsampleFast(
+      enh_buf+ENH_BUFL-inLen,    /* Input samples */
+      inLen + ENH_BUFL_FILTEROVERHEAD,
+      downsampled,
+      inLen / 2,
+      (int16_t*)WebRtcIlbcfix_kLpFiltCoefs,  /* Coefficients in Q12 */
+      FILTERORDER_DS_PLUS1,    /* Length of filter (order-1) */
+      FACTOR_DS,
+      DELAY_DS);
+
+  /* Estimate the pitch in the down sampled domain. */
+  for(iblock = 0; iblock<new_blocks; iblock++){
+
+    /* references */
+    target = downsampled + 60 + iblock * ENH_BLOCKL_HALF;
+    regressor = target - 10;
+
+    /* scaling */
+    max16 = WebRtcSpl_MaxAbsValueW16(&regressor[-50], ENH_BLOCKL_HALF + 50 - 1);
+    shifts = WebRtcSpl_GetSizeInBits((uint32_t)(max16 * max16)) - 25;
+    shifts = WEBRTC_SPL_MAX(0, shifts);
+
+    /* compute cross correlation */
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, ENH_BLOCKL_HALF, 50,
+                               shifts, -1);
+
+    /* Find 3 highest correlations that should be compared for the
+       highest (corr*corr)/ener */
+
+    for (i=0;i<2;i++) {
+      lagmax[i] = WebRtcSpl_MaxIndexW32(corr32, 50);
+      corrmax[i] = corr32[lagmax[i]];
+      start = WEBRTC_SPL_MAX(2, lagmax[i]) - 2;
+      stop = WEBRTC_SPL_MIN(47, lagmax[i]) + 2;
+      for (k = start; k <= stop; k++) {
+        corr32[k] = 0;
+      }
+    }
+    lagmax[2] = WebRtcSpl_MaxIndexW32(corr32, 50);
+    corrmax[2] = corr32[lagmax[2]];
+
+    /* Calculate normalized corr^2 and ener */
+    for (i=0;i<3;i++) {
+      corrSh = 15-WebRtcSpl_GetSizeInBits(corrmax[i]);
+      ener = WebRtcSpl_DotProductWithScale(regressor - lagmax[i],
+                                           regressor - lagmax[i],
+                                           ENH_BLOCKL_HALF, shifts);
+      enerSh = 15-WebRtcSpl_GetSizeInBits(ener);
+      corr16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(corrmax[i], corrSh);
+      corr16[i] = (int16_t)((corr16[i] * corr16[i]) >> 16);
+      en16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(ener, enerSh);
+      totsh[i] = enerSh - 2 * corrSh;
+    }
+
+    /* Compare lagmax[0..3] for the (corr^2)/ener criteria */
+    ind = 0;
+    for (i=1; i<3; i++) {
+      if (totsh[ind] > totsh[i]) {
+        sh = WEBRTC_SPL_MIN(31, totsh[ind]-totsh[i]);
+        if (corr16[ind] * en16[i] < (corr16[i] * en16[ind]) >> sh) {
+          ind = i;
+        }
+      } else {
+        sh = WEBRTC_SPL_MIN(31, totsh[i]-totsh[ind]);
+        if ((corr16[ind] * en16[i]) >> sh < corr16[i] * en16[ind]) {
+          ind = i;
+        }
+      }
+    }
+
+    lag = lagmax[ind] + 10;
+
+    /* Store the estimated lag in the non-downsampled domain */
+    enh_period[ENH_NBLOCKS_TOT - new_blocks + iblock] = lag * 8;
+
+    /* Store the estimated lag for backward PLC */
+    if (iLBCdec_inst->prev_enh_pl==1) {
+      if (!iblock) {
+        tlag = lag * 2;
+      }
+    } else {
+      if (iblock==1) {
+        tlag = lag * 2;
+      }
+    }
+
+    lag *= 2;
+  }
+
+  if ((iLBCdec_inst->prev_enh_pl==1)||(iLBCdec_inst->prev_enh_pl==2)) {
+
+    /* Calculate the best lag of the new frame
+       This is used to interpolate backwards and mix with the PLC'd data
+    */
+
+    /* references */
+    target=in;
+    regressor=in+tlag-1;
+
+    /* scaling */
+    // Note that this is not abs-max, so we will take the absolute value below.
+    max16 = WebRtcSpl_MaxAbsElementW16(regressor, plc_blockl + 3 - 1);
+    const int16_t max_target =
+        WebRtcSpl_MaxAbsElementW16(target, plc_blockl + 3 - 1);
+    const int64_t max_val = plc_blockl * abs(max16 * max_target);
+    const int32_t factor = max_val >> 31;
+    shifts = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+
+    /* compute cross correlation */
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, plc_blockl, 3, shifts,
+                               1);
+
+    /* find lag */
+    lag=WebRtcSpl_MaxIndexW32(corr32, 3);
+    lag+=tlag-1;
+
+    /* Copy the backward PLC to plc_pred */
+
+    if (iLBCdec_inst->prev_enh_pl==1) {
+      if (lag>plc_blockl) {
+        WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-plc_blockl], plc_blockl);
+      } else {
+        WEBRTC_SPL_MEMCPY_W16(&plc_pred[plc_blockl-lag], in, lag);
+        WEBRTC_SPL_MEMCPY_W16(
+            plc_pred, &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl+lag],
+            (plc_blockl-lag));
+      }
+    } else {
+      size_t pos;
+
+      pos = plc_blockl;
+
+      while (lag<pos) {
+        WEBRTC_SPL_MEMCPY_W16(&plc_pred[pos-lag], in, lag);
+        pos = pos - lag;
+      }
+      WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-pos], pos);
+
+    }
+
+    if (iLBCdec_inst->prev_enh_pl==1) {
+      /* limit energy change
+         if energy in backward PLC is more than 4 times higher than the forward
+         PLC, then reduce the energy in the backward PLC vector:
+         sample 1...len-16 set energy of the to 4 times forward PLC
+         sample len-15..len interpolate between 4 times fw PLC and bw PLC energy
+
+         Note: Compared to floating point code there is a slight change,
+         the window is 16 samples long instead of 10 samples to simplify the
+         calculations
+      */
+
+      max=WebRtcSpl_MaxAbsValueW16(
+          &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl], plc_blockl);
+      max16=WebRtcSpl_MaxAbsValueW16(plc_pred, plc_blockl);
+      max = WEBRTC_SPL_MAX(max, max16);
+      scale=22-(int16_t)WebRtcSpl_NormW32(max);
+      scale=WEBRTC_SPL_MAX(scale,0);
+
+      tmp2 = WebRtcSpl_DotProductWithScale(
+          &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
+          &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
+          plc_blockl, scale);
+      tmp1 = WebRtcSpl_DotProductWithScale(plc_pred, plc_pred,
+                                           plc_blockl, scale);
+
+      /* Check the energy difference */
+      if ((tmp1>0)&&((tmp1>>2)>tmp2)) {
+        /* EnChange is now guaranteed to be <0.5
+           Calculate EnChange=tmp2/tmp1 in Q16
+        */
+
+        scale1=(int16_t)WebRtcSpl_NormW32(tmp1);
+        tmp1=WEBRTC_SPL_SHIFT_W32(tmp1, (scale1-16)); /* using 15 bits */
+
+        tmp2=WEBRTC_SPL_SHIFT_W32(tmp2, (scale1));
+        EnChange = (int16_t)WebRtcSpl_DivW32W16(tmp2,
+                                                      (int16_t)tmp1);
+
+        /* Calculate the Sqrt of the energy in Q15 ((14+16)/2) */
+        SqrtEnChange = (int16_t)WebRtcSpl_SqrtFloor(EnChange << 14);
+
+
+        /* Multiply first part of vector with 2*SqrtEnChange */
+        WebRtcSpl_ScaleVector(plc_pred, plc_pred, SqrtEnChange, plc_blockl-16,
+                              14);
+
+        /* Calculate increase parameter for window part (16 last samples) */
+        /* (1-2*SqrtEnChange)/16 in Q15 */
+        inc = 2048 - (SqrtEnChange >> 3);
+
+        win=0;
+        tmpW16ptr=&plc_pred[plc_blockl-16];
+
+        for (i=16;i>0;i--) {
+          *tmpW16ptr = (int16_t)(
+              (*tmpW16ptr * (SqrtEnChange + (win >> 1))) >> 14);
+          /* multiply by (2.0*SqrtEnChange+win) */
+
+          win += inc;
+          tmpW16ptr++;
+        }
+      }
+
+      /* Make the linear interpolation between the forward PLC'd data
+         and the backward PLC'd data (from the new frame)
+      */
+
+      if (plc_blockl==40) {
+        inc=400; /* 1/41 in Q14 */
+      } else { /* plc_blockl==80 */
+        inc=202; /* 1/81 in Q14 */
+      }
+      win=0;
+      enh_bufPtr1=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
+      for (i=0; i<plc_blockl; i++) {
+        win+=inc;
+        *enh_bufPtr1 = (int16_t)((*enh_bufPtr1 * win) >> 14);
+        *enh_bufPtr1 += (int16_t)(
+            ((16384 - win) * plc_pred[plc_blockl - 1 - i]) >> 14);
+        enh_bufPtr1--;
+      }
+    } else {
+      int16_t *synt = &downsampled[LPC_FILTERORDER];
+
+      enh_bufPtr1=&enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl];
+      WEBRTC_SPL_MEMCPY_W16(enh_bufPtr1, plc_pred, plc_blockl);
+
+      /* Clear fileter memory */
+      WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
+      WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
+      WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
+
+      /* Initialize filter memory by filtering through 2 lags */
+      WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], iLBCdec_inst->syntMem,
+                            LPC_FILTERORDER);
+      WebRtcSpl_FilterARFastQ12(
+          enh_bufPtr1,
+          synt,
+          &iLBCdec_inst->old_syntdenum[
+              (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, lag);
+
+      WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], &synt[lag-LPC_FILTERORDER],
+                            LPC_FILTERORDER);
+      WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
+                             iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+                             lag);
+      WebRtcSpl_FilterARFastQ12(
+          enh_bufPtr1, synt,
+          &iLBCdec_inst->old_syntdenum[
+              (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, lag);
+
+      WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &synt[lag-LPC_FILTERORDER],
+                            LPC_FILTERORDER);
+      WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
+                             iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+                             lag);
+    }
+  }
+
+
+  /* Perform enhancement block by block */
+
+  for (iblock = 0; iblock<new_blocks; iblock++) {
+    WebRtcIlbcfix_Enhancer(out + iblock * ENH_BLOCKL,
+                           enh_buf,
+                           ENH_BUFL,
+                           iblock * ENH_BLOCKL + startPos,
+                           enh_period,
+                           WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT);
+  }
+
+  return (lag);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
new file mode 100644
index 0000000000..5022a47c3a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhancerInterface.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * interface for enhancer
+ *---------------------------------------------------------------*/
+
+size_t  // (o) Estimated lag in end of in[]
+WebRtcIlbcfix_EnhancerInterface(int16_t* out,       // (o) enhanced signal
+                                const int16_t* in,  // (i) unenhanced signal
+                                IlbcDecoder* iLBCdec_inst);  // (i) buffers etc
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
new file mode 100644
index 0000000000..6b4f30c96b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FilteredCbVecs.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Construct an additional codebook vector by filtering the
+ *  initial codebook buffer. This vector is then used to expand
+ *  the codebook with an additional section.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_FilteredCbVecs(
+    int16_t *cbvectors, /* (o) Codebook vector for the higher section */
+    int16_t *CBmem,  /* (i) Codebook memory that is filtered to create a
+                                           second CB section */
+    size_t lMem,  /* (i) Length of codebook memory */
+    size_t samples    /* (i) Number of samples to filter */
+                                  ) {
+
+  /* Set up the memory, start with zero state */
+  WebRtcSpl_MemSetW16(CBmem+lMem, 0, CB_HALFFILTERLEN);
+  WebRtcSpl_MemSetW16(CBmem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
+  WebRtcSpl_MemSetW16(cbvectors, 0, lMem-samples);
+
+  /* Filter to obtain the filtered CB memory */
+
+  WebRtcSpl_FilterMAFastQ12(
+      CBmem+CB_HALFFILTERLEN+lMem-samples, cbvectors+lMem-samples,
+      (int16_t*)WebRtcIlbcfix_kCbFiltersRev, CB_FILTERLEN, samples);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
new file mode 100644
index 0000000000..661262e42e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FilteredCbVecs.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Construct an additional codebook vector by filtering the
+ *  initial codebook buffer. This vector is then used to expand
+ *  the codebook with an additional section.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_FilteredCbVecs(
+    int16_t* cbvectors, /* (o) Codebook vector for the higher section */
+    int16_t* CBmem,     /* (i) Codebook memory that is filtered to create a
+                                              second CB section */
+    size_t lMem,        /* (i) Length of codebook memory */
+    size_t samples      /* (i) Number of samples to filter */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
new file mode 100644
index 0000000000..c1084b1645
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FrameClassify.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/frame_classify.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Classification of subframes to localize start state
+ *---------------------------------------------------------------*/
+
+size_t WebRtcIlbcfix_FrameClassify(
+    /* (o) Index to the max-energy sub frame */
+    IlbcEncoder *iLBCenc_inst,
+    /* (i/o) the encoder state structure */
+    int16_t *residualFIX /* (i) lpc residual signal */
+                                                ){
+  int16_t max, scale;
+  int32_t ssqEn[NSUB_MAX-1];
+  int16_t *ssqPtr;
+  int32_t *seqEnPtr;
+  int32_t maxW32;
+  int16_t scale1;
+  size_t pos;
+  size_t n;
+
+  /*
+    Calculate the energy of each of the 80 sample blocks
+    in the draft the 4 first and last samples are windowed with 1/5...4/5
+    and 4/5...1/5 respectively. To simplify for the fixpoint we have changed
+    this to 0 0 1 1 and 1 1 0 0
+  */
+
+  max = WebRtcSpl_MaxAbsValueW16(residualFIX, iLBCenc_inst->blockl);
+  scale = WebRtcSpl_GetSizeInBits((uint32_t)(max * max));
+
+  /* Scale to maximum 24 bits so that it won't overflow for 76 samples */
+  scale = scale-24;
+  scale1 = WEBRTC_SPL_MAX(0, scale);
+
+  /* Calculate energies */
+  ssqPtr=residualFIX + 2;
+  seqEnPtr=ssqEn;
+  for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
+    (*seqEnPtr) = WebRtcSpl_DotProductWithScale(ssqPtr, ssqPtr, 76, scale1);
+    ssqPtr += 40;
+    seqEnPtr++;
+  }
+
+  /* Scale to maximum 20 bits in order to allow for the 11 bit window */
+  maxW32 = WebRtcSpl_MaxValueW32(ssqEn, iLBCenc_inst->nsub - 1);
+  scale = WebRtcSpl_GetSizeInBits(maxW32) - 20;
+  scale1 = WEBRTC_SPL_MAX(0, scale);
+
+  /* Window each 80 block with the ssqEn_winTbl window to give higher probability for
+     the blocks in the middle
+  */
+  seqEnPtr=ssqEn;
+  if (iLBCenc_inst->mode==20) {
+    ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin+1;
+  } else {
+    ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin;
+  }
+  for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
+    (*seqEnPtr)=WEBRTC_SPL_MUL(((*seqEnPtr)>>scale1), (*ssqPtr));
+    seqEnPtr++;
+    ssqPtr++;
+  }
+
+  /* Extract the best choise of start state */
+  pos = WebRtcSpl_MaxIndexW32(ssqEn, iLBCenc_inst->nsub - 1) + 1;
+
+  return(pos);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
new file mode 100644
index 0000000000..7615106d70
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FrameClassify.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+size_t WebRtcIlbcfix_FrameClassify(
+    /* (o) Index to the max-energy sub frame */
+    IlbcEncoder* iLBCenc_inst,
+    /* (i/o) the encoder state structure */
+    int16_t* residualFIX /* (i) lpc residual signal */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c
new file mode 100644
index 0000000000..1357dece33
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainDequant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/gain_dequant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  decoder for quantized gains in the gain-shape coding of
+ *  residual
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_GainDequant(
+    /* (o) quantized gain value (Q14) */
+    int16_t index, /* (i) quantization index */
+    int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
+    int16_t stage /* (i) The stage of the search */
+                                                ){
+  int16_t scale;
+  const int16_t *gain;
+
+  /* obtain correct scale factor */
+
+  scale=WEBRTC_SPL_ABS_W16(maxIn);
+  scale = WEBRTC_SPL_MAX(1638, scale);  /* if lower than 0.1, set it to 0.1 */
+
+  /* select the quantization table and return the decoded value */
+  gain = WebRtcIlbcfix_kGain[stage];
+
+  return (int16_t)((scale * gain[index] + 8192) >> 14);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h
new file mode 100644
index 0000000000..2b97550b6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainDequant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  decoder for quantized gains in the gain-shape coding of
+ *  residual
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_GainDequant(
+    /* (o) quantized gain value (Q14) */
+    int16_t index, /* (i) quantization index */
+    int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
+    int16_t stage  /* (i) The stage of the search */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c
new file mode 100644
index 0000000000..9a6d49d51a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainQuant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/gain_quant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  quantizer for the gain in the gain-shape coding of residual
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
+    int16_t gain, /* (i) gain value Q14 */
+    int16_t maxIn, /* (i) maximum of gain value Q14 */
+    int16_t stage, /* (i) The stage of the search */
+    int16_t *index /* (o) quantization index */
+                                        ) {
+
+  int16_t scale, cblen;
+  int32_t gainW32, measure1, measure2;
+  const int16_t *cbPtr, *cb;
+  int loc, noMoves, noChecks, i;
+
+  /* ensure a lower bound (0.1) on the scaling factor */
+
+  scale = WEBRTC_SPL_MAX(1638, maxIn);
+
+  /* select the quantization table and calculate
+     the length of the table and the number of
+     steps in the binary search that are needed */
+  cb = WebRtcIlbcfix_kGain[stage];
+  cblen = 32>>stage;
+  noChecks = 4-stage;
+
+  /* Multiply the gain with 2^14 to make the comparison
+     easier and with higher precision */
+  gainW32 = gain << 14;
+
+  /* Do a binary search, starting in the middle of the CB
+     loc - defines the current position in the table
+     noMoves - defines the number of steps to move in the CB in order
+     to get next CB location
+  */
+
+  loc = cblen>>1;
+  noMoves = loc;
+  cbPtr = cb + loc; /* Centre of CB */
+
+  for (i=noChecks;i>0;i--) {
+    noMoves>>=1;
+    measure1 = scale * *cbPtr;
+
+    /* Move up if gain is larger, otherwise move down in table */
+    measure1 = measure1 - gainW32;
+
+    if (0>measure1) {
+      cbPtr+=noMoves;
+      loc+=noMoves;
+    } else {
+      cbPtr-=noMoves;
+      loc-=noMoves;
+    }
+  }
+
+  /* Check which value is the closest one: loc-1, loc or loc+1 */
+
+  measure1 = scale * *cbPtr;
+  if (gainW32>measure1) {
+    /* Check against value above loc */
+    measure2 = scale * cbPtr[1];
+    if ((measure2-gainW32)<(gainW32-measure1)) {
+      loc+=1;
+    }
+  } else {
+    /* Check against value below loc */
+    measure2 = scale * cbPtr[-1];
+    if ((gainW32-measure2)<=(measure1-gainW32)) {
+      loc-=1;
+    }
+  }
+
+  /* Guard against getting outside the table. The calculation above can give a location
+     which is one above the maximum value (in very rare cases) */
+  loc=WEBRTC_SPL_MIN(loc, (cblen-1));
+  *index=loc;
+
+  /* Calculate and return the quantized gain value (in Q14) */
+  return (int16_t)((scale * cb[loc] + 8192) >> 14);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.h
new file mode 100644
index 0000000000..761f7d2f79
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainQuant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  quantizer for the gain in the gain-shape coding of residual
+ *---------------------------------------------------------------*/
+
+int16_t
+WebRtcIlbcfix_GainQuant(               /* (o) quantized gain value */
+                        int16_t gain,  /* (i) gain value Q14 */
+                        int16_t maxIn, /* (i) maximum of gain value Q14 */
+                        int16_t stage, /* (i) The stage of the search */
+                        int16_t* index /* (o) quantization index */
+                        );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
new file mode 100644
index 0000000000..e9cd2008e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetCbVec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/get_cd_vec.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Construct codebook vector for given index.
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_GetCbVec(
+    int16_t *cbvec,   /* (o) Constructed codebook vector */
+    int16_t *mem,   /* (i) Codebook buffer */
+    size_t index,   /* (i) Codebook index */
+    size_t lMem,   /* (i) Length of codebook buffer */
+    size_t cbveclen   /* (i) Codebook vector length */
+                            ){
+  size_t k, base_size;
+  size_t lag;
+  /* Stack based */
+  int16_t tempbuff2[SUBL+5];
+
+  /* Determine size of codebook sections */
+
+  base_size=lMem-cbveclen+1;
+
+  if (cbveclen==SUBL) {
+    base_size += cbveclen / 2;
+  }
+
+  /* No filter -> First codebook section */
+
+  if (index<lMem-cbveclen+1) {
+
+    /* first non-interpolated vectors */
+
+    k=index+cbveclen;
+    /* get vector */
+    WEBRTC_SPL_MEMCPY_W16(cbvec, mem+lMem-k, cbveclen);
+
+  } else if (index < base_size) {
+
+    /* Calculate lag */
+
+    k = (2 * (index - (lMem - cbveclen + 1))) + cbveclen;
+
+    lag = k / 2;
+
+    WebRtcIlbcfix_CreateAugmentedVec(lag, mem+lMem, cbvec);
+
+  }
+
+  /* Higher codebbok section based on filtering */
+
+  else {
+
+    size_t memIndTest;
+
+    /* first non-interpolated vectors */
+
+    if (index-base_size<lMem-cbveclen+1) {
+
+      /* Set up filter memory, stuff zeros outside memory buffer */
+
+      memIndTest = lMem-(index-base_size+cbveclen);
+
+      WebRtcSpl_MemSetW16(mem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
+      WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
+
+      /* do filtering to get the codebook vector */
+
+      WebRtcSpl_FilterMAFastQ12(
+          &mem[memIndTest+4], cbvec, (int16_t*)WebRtcIlbcfix_kCbFiltersRev,
+          CB_FILTERLEN, cbveclen);
+    }
+
+    /* interpolated vectors */
+
+    else {
+      if (cbveclen < SUBL) {
+        // We're going to fill in cbveclen + 5 elements of tempbuff2 in
+        // WebRtcSpl_FilterMAFastQ12, less than the SUBL + 5 elements we'll be
+        // using in WebRtcIlbcfix_CreateAugmentedVec. This error is caused by
+        // bad values in `index` (which come from the encoded stream). Tell the
+        // caller that things went south, and that the decoder state is now
+        // corrupt (because it's half-way through an update that we can't
+        // complete).
+        return false;
+      }
+
+      /* Stuff zeros outside memory buffer  */
+      memIndTest = lMem-cbveclen-CB_FILTERLEN;
+      WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
+
+      /* do filtering */
+      WebRtcSpl_FilterMAFastQ12(
+          &mem[memIndTest+7], tempbuff2, (int16_t*)WebRtcIlbcfix_kCbFiltersRev,
+          CB_FILTERLEN, cbveclen+5);
+
+      /* Calculate lag index */
+      lag = (cbveclen<<1)-20+index-base_size-lMem-1;
+
+      WebRtcIlbcfix_CreateAugmentedVec(lag, tempbuff2+SUBL+5, cbvec);
+    }
+  }
+
+  return true;  // Success.
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
new file mode 100644
index 0000000000..99537dd0f7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetCbVec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+
+#include <stdbool.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+// Returns true on success, false on failure. In case of failure, the decoder
+// state may be corrupted and needs resetting.
+ABSL_MUST_USE_RESULT
+bool WebRtcIlbcfix_GetCbVec(
+    int16_t* cbvec, /* (o) Constructed codebook vector */
+    int16_t* mem,   /* (i) Codebook buffer */
+    size_t index,   /* (i) Codebook index */
+    size_t lMem,    /* (i) Length of codebook buffer */
+    size_t cbveclen /* (i) Codebook vector length */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
new file mode 100644
index 0000000000..e0fb21caf0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetLspPoly.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/get_lsp_poly.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * Construct the polynomials F1(z) and F2(z) from the LSP
+ * (Computations are done in Q24)
+ *
+ * The expansion is performed using the following recursion:
+ *
+ * f[0] = 1;
+ * tmp = -2.0 * lsp[0];
+ * f[1] = tmp;
+ * for (i=2; i<=5; i++) {
+ *    b = -2.0 * lsp[2*i-2];
+ *    f[i] = tmp*f[i-1] + 2.0*f[i-2];
+ *    for (j=i; j>=2; j--) {
+ *       f[j] = f[j] + tmp*f[j-1] + f[j-2];
+ *    }
+ *    f[i] = f[i] + tmp;
+ * }
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetLspPoly(
+    int16_t *lsp, /* (i) LSP in Q15 */
+    int32_t *f)  /* (o) polonymial in Q24 */
+{
+  int32_t tmpW32;
+  int i, j;
+  int16_t high, low;
+  int16_t *lspPtr;
+  int32_t *fPtr;
+
+  lspPtr = lsp;
+  fPtr = f;
+  /* f[0] = 1.0 (Q24) */
+  (*fPtr) = (int32_t)16777216;
+  fPtr++;
+
+  (*fPtr) = WEBRTC_SPL_MUL((*lspPtr), -1024);
+  fPtr++;
+  lspPtr+=2;
+
+  for(i=2; i<=5; i++)
+  {
+    (*fPtr) = fPtr[-2];
+
+    for(j=i; j>1; j--)
+    {
+      /* Compute f[j] = f[j] + tmp*f[j-1] + f[j-2]; */
+      high = (int16_t)(fPtr[-1] >> 16);
+      low = (int16_t)((fPtr[-1] & 0xffff) >> 1);
+
+      tmpW32 = 4 * high * *lspPtr + 4 * ((low * *lspPtr) >> 15);
+
+      (*fPtr) += fPtr[-2];
+      (*fPtr) -= tmpW32;
+      fPtr--;
+    }
+    *fPtr -= *lspPtr * (1 << 10);
+
+    fPtr+=i;
+    lspPtr+=2;
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
new file mode 100644
index 0000000000..70c9c4d4b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetLspPoly.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * Construct the polynomials F1(z) and F2(z) from the LSP
+ * (Computations are done in Q24)
+ *
+ * The expansion is performed using the following recursion:
+ *
+ * f[0] = 1;
+ * tmp = -2.0 * lsp[0];
+ * f[1] = tmp;
+ * for (i=2; i<=5; i++) {
+ *    b = -2.0 * lsp[2*i-2];
+ *    f[i] = tmp*f[i-1] + 2.0*f[i-2];
+ *    for (j=i; j>=2; j--) {
+ *       f[j] = f[j] + tmp*f[j-1] + f[j-2];
+ *    }
+ *    f[i] = f[i] + tmp;
+ * }
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetLspPoly(int16_t* lsp, /* (i) LSP in Q15 */
+                              int32_t* f);  /* (o) polonymial in Q24 */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
new file mode 100644
index 0000000000..68a569a40a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetSyncSeq.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/get_sync_seq.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/nearest_neighbor.h"
+#include "modules/audio_coding/codecs/ilbc/refiner.h"
+
+/*----------------------------------------------------------------*
+ * get the pitch-synchronous sample sequence
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetSyncSeq(
+    int16_t *idata,   /* (i) original data */
+    size_t idatal,   /* (i) dimension of data */
+    size_t centerStartPos, /* (i) where current block starts */
+    size_t *period,   /* (i) rough-pitch-period array       (Q-2) */
+    const size_t *plocs, /* (i) where periods of period array are taken (Q-2) */
+    size_t periodl,   /* (i) dimension period array */
+    size_t hl,    /* (i) 2*hl+1 is the number of sequences */
+    int16_t *surround  /* (i/o) The contribution from this sequence
+                                summed with earlier contributions */
+                              ){
+  size_t i, centerEndPos, q;
+  /* Stack based */
+  size_t lagBlock[2 * ENH_HL + 1];
+  size_t blockStartPos[2 * ENH_HL + 1]; /* The position to search around (Q2) */
+  size_t plocs2[ENH_PLOCSL];
+
+  centerEndPos = centerStartPos + ENH_BLOCKL - 1;
+
+  /* present (find predicted lag from this position) */
+
+  WebRtcIlbcfix_NearestNeighbor(lagBlock + hl,
+                                plocs,
+                                2 * (centerStartPos + centerEndPos),
+                                periodl);
+
+  blockStartPos[hl] = 4 * centerStartPos;
+
+  /* past (find predicted position and perform a refined
+     search to find the best sequence) */
+
+  for (q = hl; q > 0; q--) {
+    size_t qq = q - 1;
+    size_t period_q = period[lagBlock[q]];
+    /* Stop if this sequence would be outside the buffer; that means all
+       further-past sequences would also be outside the buffer. */
+    if (blockStartPos[q] < period_q + (4 * ENH_OVERHANG))
+      break;
+    blockStartPos[qq] = blockStartPos[q] - period_q;
+
+    size_t value = blockStartPos[qq] + 4 * ENH_BLOCKL_HALF;
+    value = (value > period_q) ? (value - period_q) : 0;
+    WebRtcIlbcfix_NearestNeighbor(lagBlock + qq, plocs, value, periodl);
+
+    /* Find the best possible sequence in the 4 times upsampled
+        domain around blockStartPos+q */
+    WebRtcIlbcfix_Refiner(blockStartPos + qq, idata, idatal, centerStartPos,
+                          blockStartPos[qq], surround,
+                          WebRtcIlbcfix_kEnhWt[qq]);
+  }
+
+  /* future (find predicted position and perform a refined
+     search to find the best sequence) */
+
+  for (i = 0; i < periodl; i++) {
+    plocs2[i] = plocs[i] - period[i];
+  }
+
+  for (q = hl + 1; q <= (2 * hl); q++) {
+
+    WebRtcIlbcfix_NearestNeighbor(
+        lagBlock + q,
+        plocs2,
+        blockStartPos[q - 1] + 4 * ENH_BLOCKL_HALF,
+        periodl);
+
+    blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
+
+    if (blockStartPos[q] + 4 * (ENH_BLOCKL + ENH_OVERHANG) < 4 * idatal) {
+
+      /* Find the best possible sequence in the 4 times upsampled
+         domain around blockStartPos+q */
+      WebRtcIlbcfix_Refiner(blockStartPos + q, idata, idatal, centerStartPos,
+                            blockStartPos[q], surround,
+                            WebRtcIlbcfix_kEnhWt[2 * hl - q]);
+
+    } else {
+      /* Don't add anything since this sequence would
+         be outside the buffer */
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
new file mode 100644
index 0000000000..90962fa063
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetSyncSeq.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * get the pitch-synchronous sample sequence
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetSyncSeq(
+    int16_t* idata,        /* (i) original data */
+    size_t idatal,         /* (i) dimension of data */
+    size_t centerStartPos, /* (i) where current block starts */
+    size_t* period,        /* (i) rough-pitch-period array       (Q-2) */
+    const size_t* plocs, /* (i) where periods of period array are taken (Q-2) */
+    size_t periodl,      /* (i) dimension period array */
+    size_t hl,           /* (i) 2*hl+1 is the number of sequences */
+    int16_t* surround    /* (i/o) The contribution from this sequence
+                                  summed with earlier contributions */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c
new file mode 100644
index 0000000000..be582f2e23
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpInput.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/hp_input.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  high-pass filter of input with *0.5 and saturation
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_HpInput(
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
+                                                                   {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                                                                   is assumed to be 1.0 */
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                                                                   yhi[n-2] ylow[n-2] */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len)      /* (i)   Number of samples to filter */
+{
+  size_t i;
+  int32_t tmpW32;
+  int32_t tmpW32b;
+
+  for (i=0; i<len; i++) {
+
+    /*
+        y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
+        + (-a[1])*y[i-1] + (-a[2])*y[i-2];
+    */
+
+    tmpW32 = y[1] * ba[3];  /* (-a[1])*y[i-1] (low part) */
+    tmpW32 += y[3] * ba[4];  /* (-a[2])*y[i-2] (low part) */
+    tmpW32 = (tmpW32>>15);
+    tmpW32 += y[0] * ba[3];  /* (-a[1])*y[i-1] (high part) */
+    tmpW32 += y[2] * ba[4];  /* (-a[2])*y[i-2] (high part) */
+    tmpW32 = (tmpW32<<1);
+
+    tmpW32 += signal[i] * ba[0];  /* b[0]*x[0] */
+    tmpW32 += x[0] * ba[1];  /* b[1]*x[i-1] */
+    tmpW32 += x[1] * ba[2];  /* b[2]*x[i-2] */
+
+    /* Update state (input part) */
+    x[1] = x[0];
+    x[0] = signal[i];
+
+    /* Rounding in Q(12+1), i.e. add 2^12 */
+    tmpW32b = tmpW32 + 4096;
+
+    /* Saturate (to 2^28) so that the HP filtered signal does not overflow */
+    tmpW32b = WEBRTC_SPL_SAT((int32_t)268435455, tmpW32b, (int32_t)-268435456);
+
+    /* Convert back to Q0 and multiply with 0.5 */
+    signal[i] = (int16_t)(tmpW32b >> 13);
+
+    /* Update state (filtered part) */
+    y[2] = y[0];
+    y[3] = y[1];
+
+    /* upshift tmpW32 by 3 with saturation */
+    if (tmpW32>268435455) {
+      tmpW32 = WEBRTC_SPL_WORD32_MAX;
+    } else if (tmpW32<-268435456) {
+      tmpW32 = WEBRTC_SPL_WORD32_MIN;
+    } else {
+      tmpW32 <<= 3;
+    }
+
+    y[0] = (int16_t)(tmpW32 >> 16);
+    y[1] = (int16_t)((tmpW32 - (y[0] << 16)) >> 1);
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.h
new file mode 100644
index 0000000000..9143d8efed
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpInput.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+// clang-format off
+// Bad job here. https://bugs.llvm.org/show_bug.cgi?id=34274
+void WebRtcIlbcfix_HpInput(
+    int16_t* signal, /* (i/o) signal vector */
+    int16_t* ba,     /* (i)   B- and A-coefficients (2:nd order)
+                              {b[0] b[1] b[2] -a[1] -a[2]}
+                              a[0] is assumed to be 1.0 */
+    int16_t* y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                              yhi[n-2] ylow[n-2] */
+    int16_t* x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len);     /* (i)   Number of samples to filter */
+// clang-format on
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c
new file mode 100644
index 0000000000..cc5f6dcd37
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpOutput.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  high-pass filter of output and *2 with saturation
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_HpOutput(
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
+                                                                   {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                                                                   is assumed to be 1.0 */
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                                                                   yhi[n-2] ylow[n-2] */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len)      /* (i)   Number of samples to filter */
+{
+  size_t i;
+  int32_t tmpW32;
+  int32_t tmpW32b;
+
+  for (i=0; i<len; i++) {
+
+    /*
+      y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
+      + (-a[1])*y[i-1] + (-a[2])*y[i-2];
+    */
+
+    tmpW32 = y[1] * ba[3];  /* (-a[1])*y[i-1] (low part) */
+    tmpW32 += y[3] * ba[4];  /* (-a[2])*y[i-2] (low part) */
+    tmpW32 = (tmpW32>>15);
+    tmpW32 += y[0] * ba[3];  /* (-a[1])*y[i-1] (high part) */
+    tmpW32 += y[2] * ba[4];  /* (-a[2])*y[i-2] (high part) */
+    tmpW32 *= 2;
+
+    tmpW32 += signal[i] * ba[0];  /* b[0]*x[0] */
+    tmpW32 += x[0] * ba[1];  /* b[1]*x[i-1] */
+    tmpW32 += x[1] * ba[2];  /* b[2]*x[i-2] */
+
+    /* Update state (input part) */
+    x[1] = x[0];
+    x[0] = signal[i];
+
+    /* Rounding in Q(12-1), i.e. add 2^10 */
+    tmpW32b = tmpW32 + 1024;
+
+    /* Saturate (to 2^26) so that the HP filtered signal does not overflow */
+    tmpW32b = WEBRTC_SPL_SAT((int32_t)67108863, tmpW32b, (int32_t)-67108864);
+
+    /* Convert back to Q0 and multiply with 2 */
+    signal[i] = (int16_t)(tmpW32b >> 11);
+
+    /* Update state (filtered part) */
+    y[2] = y[0];
+    y[3] = y[1];
+
+    /* upshift tmpW32 by 3 with saturation */
+    if (tmpW32>268435455) {
+      tmpW32 = WEBRTC_SPL_WORD32_MAX;
+    } else if (tmpW32<-268435456) {
+      tmpW32 = WEBRTC_SPL_WORD32_MIN;
+    } else {
+      tmpW32 *= 8;
+    }
+
+    y[0] = (int16_t)(tmpW32 >> 16);
+    y[1] = (int16_t)((tmpW32 & 0xffff) >> 1);
+
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.h
new file mode 100644
index 0000000000..6d1bd3cd88
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpOutput.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+// clang-format off
+// Bad job here. https://bugs.llvm.org/show_bug.cgi?id=34274
+void WebRtcIlbcfix_HpOutput(
+    int16_t* signal, /* (i/o) signal vector */
+    int16_t* ba,     /* (i)   B- and A-coefficients (2:nd order)
+                              {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                              is assumed to be 1.0 */
+    int16_t* y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                              yhi[n-2] ylow[n-2] */
+    int16_t* x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len);     /* (i)   Number of samples to filter */
+// clang-format on
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c
new file mode 100644
index 0000000000..ba6c3e46c3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -0,0 +1,288 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBCInterface.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+#include <stdlib.h>
+
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/init_encode.h"
+#include "rtc_base/checks.h"
+
+int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
+                                    int16_t* ILBCENC_inst_Addr,
+                                    int16_t* size) {
+  *iLBC_encinst=(IlbcEncoderInstance*)ILBCENC_inst_Addr;
+  *size=sizeof(IlbcEncoder)/sizeof(int16_t);
+  if (*iLBC_encinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance** iLBC_decinst,
+                                    int16_t* ILBCDEC_inst_Addr,
+                                    int16_t* size) {
+  *iLBC_decinst=(IlbcDecoderInstance*)ILBCDEC_inst_Addr;
+  *size=sizeof(IlbcDecoder)/sizeof(int16_t);
+  if (*iLBC_decinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance **iLBC_encinst) {
+  *iLBC_encinst=(IlbcEncoderInstance*)malloc(sizeof(IlbcEncoder));
+  if (*iLBC_encinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance **iLBC_decinst) {
+  *iLBC_decinst=(IlbcDecoderInstance*)malloc(sizeof(IlbcDecoder));
+  if (*iLBC_decinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance *iLBC_encinst) {
+  free(iLBC_encinst);
+  return(0);
+}
+
+int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance *iLBC_decinst) {
+  free(iLBC_decinst);
+  return(0);
+}
+
+int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
+                                  int16_t mode) {
+  if ((mode==20)||(mode==30)) {
+    WebRtcIlbcfix_InitEncode((IlbcEncoder*) iLBCenc_inst, mode);
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded) {
+  size_t pos = 0;
+  size_t encpos = 0;
+
+  if ((len != ((IlbcEncoder*)iLBCenc_inst)->blockl) &&
+#ifdef SPLIT_10MS
+      (len != 80) &&
+#endif
+      (len != 2*((IlbcEncoder*)iLBCenc_inst)->blockl) &&
+      (len != 3*((IlbcEncoder*)iLBCenc_inst)->blockl))
+  {
+    /* A maximum of 3 frames/packet is allowed */
+    return(-1);
+  } else {
+
+    /* call encoder */
+    while (pos<len) {
+      WebRtcIlbcfix_EncodeImpl((uint16_t*)&encoded[2 * encpos], &speechIn[pos],
+                               (IlbcEncoder*)iLBCenc_inst);
+#ifdef SPLIT_10MS
+      pos += 80;
+      if(((IlbcEncoder*)iLBCenc_inst)->section == 0)
+#else
+        pos += ((IlbcEncoder*)iLBCenc_inst)->blockl;
+#endif
+      encpos += ((IlbcEncoder*)iLBCenc_inst)->no_of_words;
+    }
+    return (int)(encpos*2);
+  }
+}
+
+int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
+                                  int16_t mode) {
+  if ((mode==20)||(mode==30)) {
+    WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, mode, 1);
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst) {
+  WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 20, 1);
+}
+void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst) {
+  WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 30, 1);
+}
+
+
+int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType)
+{
+  size_t i=0;
+  /* Allow for automatic switching between the frame sizes
+     (although you do get some discontinuity) */
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
+    /* ok, do nothing */
+  } else {
+    /* Test if the mode has changed */
+    if (((IlbcDecoder*)iLBCdec_inst)->mode==20) {
+      if ((len==NO_OF_BYTES_30MS)||
+          (len==2*NO_OF_BYTES_30MS)||
+          (len==3*NO_OF_BYTES_30MS)) {
+        WebRtcIlbcfix_InitDecode(
+            ((IlbcDecoder*)iLBCdec_inst), 30,
+            ((IlbcDecoder*)iLBCdec_inst)->use_enhancer);
+      } else {
+        /* Unsupported frame length */
+        return(-1);
+      }
+    } else {
+      if ((len==NO_OF_BYTES_20MS)||
+          (len==2*NO_OF_BYTES_20MS)||
+          (len==3*NO_OF_BYTES_20MS)) {
+        WebRtcIlbcfix_InitDecode(
+            ((IlbcDecoder*)iLBCdec_inst), 20,
+            ((IlbcDecoder*)iLBCdec_inst)->use_enhancer);
+      } else {
+        /* Unsupported frame length */
+        return(-1);
+      }
+    }
+  }
+
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
+    if (WebRtcIlbcfix_DecodeImpl(
+            &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
+            (const uint16_t*)&encoded
+                [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+            (IlbcDecoder*)iLBCdec_inst, 1) == -1)
+      return -1;
+    i++;
+  }
+  /* iLBC does not support VAD/CNG yet */
+  *speechType=1;
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
+{
+  size_t i=0;
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
+    /* ok, do nothing */
+  } else {
+    return(-1);
+  }
+
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
+    if (!WebRtcIlbcfix_DecodeImpl(
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
+        (const uint16_t*)&encoded
+            [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+        (IlbcDecoder*)iLBCdec_inst, 1))
+      return -1;
+    i++;
+  }
+  /* iLBC does not support VAD/CNG yet */
+  *speechType=1;
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
+{
+  size_t i=0;
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
+    /* ok, do nothing */
+  } else {
+    return(-1);
+  }
+
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
+    if (!WebRtcIlbcfix_DecodeImpl(
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
+        (const uint16_t*)&encoded
+            [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+        (IlbcDecoder*)iLBCdec_inst, 1))
+      return -1;
+    i++;
+  }
+  /* iLBC does not support VAD/CNG yet */
+  *speechType=1;
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
+                               int16_t* decoded,
+                               size_t noOfLostFrames) {
+  size_t i;
+  uint16_t dummy;
+
+  for (i=0;i<noOfLostFrames;i++) {
+    // PLC decoding shouldn't fail, because there is no external input data
+    // that can be bad.
+    int result = WebRtcIlbcfix_DecodeImpl(
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl], &dummy,
+        (IlbcDecoder*)iLBCdec_inst, 0);
+    RTC_CHECK_EQ(result, 0);
+  }
+  return (noOfLostFrames*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
+                              int16_t* decoded,
+                              size_t noOfLostFrames) {
+  /* Two input parameters not used, but needed for function pointers in NetEQ */
+  (void)(decoded = NULL);
+  (void)(noOfLostFrames = 0);
+
+  WebRtcSpl_MemSetW16(((IlbcDecoder*)iLBCdec_inst)->enh_buf, 0, ENH_BUFL);
+  ((IlbcDecoder*)iLBCdec_inst)->prev_enh_pl = 2;
+
+  return (0);
+}
+
+void WebRtcIlbcfix_version(char *version)
+{
+  strcpy((char*)version, "1.1.1");
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.h
new file mode 100644
index 0000000000..de8cfde111
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.h
@@ -0,0 +1,251 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * ilbc.h
+ *
+ * This header file contains all of the API's for iLBC.
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*
+ * Solution to support multiple instances
+ * Customer has to cast instance to proper type
+ */
+
+typedef struct iLBC_encinst_t_ IlbcEncoderInstance;
+
+typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
+
+/*
+ * Comfort noise constants
+ */
+
+#define ILBC_SPEECH 1
+#define ILBC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcIlbcfix_XxxAssign(...)
+ *
+ * These functions assigns the encoder/decoder instance to the specified
+ * memory location
+ *
+ * Input:
+ *     - XXX_xxxinst       : Pointer to created instance that should be
+ *                           assigned
+ *     - ILBCXXX_inst_Addr : Pointer to the desired memory space
+ *     - size              : The size that this structure occupies (in Word16)
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
+                                    int16_t* ILBCENC_inst_Addr,
+                                    int16_t* size);
+int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance** iLBC_decinst,
+                                    int16_t* ILBCDEC_inst_Addr,
+                                    int16_t* size);
+
+/****************************************************************************
+ * WebRtcIlbcfix_XxxAssign(...)
+ *
+ * These functions create a instance to the specified structure
+ *
+ * Input:
+ *      - XXX_inst        : Pointer to created instance that should be created
+ *
+ * Return value           :  0 - Ok
+ *                          -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance** iLBC_encinst);
+int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance** iLBC_decinst);
+
+/****************************************************************************
+ * WebRtcIlbcfix_XxxFree(...)
+ *
+ * These functions frees the dynamic memory of a specified instance
+ *
+ * Input:
+ *      - XXX_inst          : Pointer to created instance that should be freed
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance* iLBC_encinst);
+int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance* iLBC_decinst);
+
+/****************************************************************************
+ * WebRtcIlbcfix_EncoderInit(...)
+ *
+ * This function initializes a iLBC instance
+ *
+ * Input:
+ *      - iLBCenc_inst      : iLBC instance, i.e. the user that should receive
+ *                            be initialized
+ *      - frameLen          : The frame length of the codec 20/30 (ms)
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
+                                  int16_t frameLen);
+
+/****************************************************************************
+ * WebRtcIlbcfix_Encode(...)
+ *
+ * This function encodes one iLBC frame. Input speech length has be a
+ * multiple of the frame length.
+ *
+ * Input:
+ *      - iLBCenc_inst      : iLBC instance, i.e. the user that should encode
+ *                            a package
+ *      - speechIn          : Input speech vector
+ *      - len               : Samples in speechIn (160, 240, 320 or 480)
+ *
+ * Output:
+ *  - encoded               : The encoded data vector
+ *
+ * Return value             : >0 - Length (in bytes) of coded data
+ *                            -1 - Error
+ */
+
+int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcIlbcfix_DecoderInit(...)
+ *
+ * This function initializes a iLBC instance with either 20 or 30 ms frames
+ * Alternatively the WebRtcIlbcfix_DecoderInit_XXms can be used. Then it's
+ * not needed to specify the frame length with a variable.
+ *
+ * Input:
+ *      - IlbcDecoderInstance : iLBC decoder instance
+ *      - frameLen            : The frame length of the codec 20/30 (ms)
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
+                                  int16_t frameLen);
+void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst);
+void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst);
+
+/****************************************************************************
+ * WebRtcIlbcfix_Decode(...)
+ *
+ * This function decodes a packet with iLBC frame(s). Output speech length
+ * will be a multiple of 160 or 240 samples ((160 or 240)*frames/packet).
+ *
+ * Input:
+ *      - iLBCdec_inst      : iLBC instance, i.e. the user that should decode
+ *                            a packet
+ *      - encoded           : Encoded iLBC frame(s)
+ *      - len               : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded           : The decoded vector
+ *      - speechType        : 1 normal, 2 CNG
+ *
+ * Return value             : >0 - Samples in decoded vector
+ *                            -1 - Error
+ */
+
+int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType);
+int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType);
+int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcIlbcfix_DecodePlc(...)
+ *
+ * This function conducts PLC for iLBC frame(s). Output speech length
+ * will be a multiple of 160 or 240 samples.
+ *
+ * Input:
+ *      - iLBCdec_inst      : iLBC instance, i.e. the user that should perform
+ *                            a PLC
+ *      - noOfLostFrames    : Number of PLC frames to produce
+ *
+ * Output:
+ *      - decoded           : The "decoded" vector
+ *
+ * Return value             : Samples in decoded PLC vector
+ */
+
+size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
+                               int16_t* decoded,
+                               size_t noOfLostFrames);
+
+/****************************************************************************
+ * WebRtcIlbcfix_NetEqPlc(...)
+ *
+ * This function updates the decoder when a packet loss has occured, but it
+ * does not produce any PLC data. Function can be used if another PLC method
+ * is used (i.e NetEq).
+ *
+ * Input:
+ *      - iLBCdec_inst      : iLBC instance that should be updated
+ *      - noOfLostFrames    : Number of lost frames
+ *
+ * Output:
+ *      - decoded           : The "decoded" vector (nothing in this case)
+ *
+ * Return value             : Samples in decoded PLC vector
+ */
+
+size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
+                              int16_t* decoded,
+                              size_t noOfLostFrames);
+
+/****************************************************************************
+ * WebRtcIlbcfix_version(...)
+ *
+ * This function returns the version number of iLBC
+ *
+ * Output:
+ *      - version           : Version number of iLBC (maximum 20 char)
+ */
+
+void WebRtcIlbcfix_version(char* version);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
new file mode 100644
index 0000000000..689292f131
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
@@ -0,0 +1,140 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(IlbcTest, BadPacket) {
+  // Get a good packet.
+  AudioEncoderIlbcConfig config;
+  config.frame_size_ms = 20;  // We need 20 ms rather than the default 30 ms;
+                              // otherwise, all possible values of cb_index[2]
+                              // are valid.
+  AudioEncoderIlbcImpl encoder(config, 102);
+  std::vector<int16_t> samples(encoder.SampleRateHz() / 100, 4711);
+  rtc::Buffer packet;
+  int num_10ms_chunks = 0;
+  while (packet.size() == 0) {
+    encoder.Encode(0, samples, &packet);
+    num_10ms_chunks += 1;
+  }
+
+  // Break the packet by setting all bits of the unsigned 7-bit number
+  // cb_index[2] to 1, giving it a value of 127. For a 20 ms packet, this is
+  // too large.
+  EXPECT_EQ(38u, packet.size());
+  rtc::Buffer bad_packet(packet.data(), packet.size());
+  bad_packet[29] |= 0x3f;  // Bits 1-6.
+  bad_packet[30] |= 0x80;  // Bit 0.
+
+  // Decode the bad packet. We expect the decoder to respond by returning -1.
+  AudioDecoderIlbcImpl decoder;
+  std::vector<int16_t> decoded_samples(num_10ms_chunks * samples.size());
+  AudioDecoder::SpeechType speech_type;
+  EXPECT_EQ(-1, decoder.Decode(bad_packet.data(), bad_packet.size(),
+                               encoder.SampleRateHz(),
+                               sizeof(int16_t) * decoded_samples.size(),
+                               decoded_samples.data(), &speech_type));
+
+  // Decode the good packet. This should work, because the failed decoding
+  // should not have left the decoder in a broken state.
+  EXPECT_EQ(static_cast<int>(decoded_samples.size()),
+            decoder.Decode(packet.data(), packet.size(), encoder.SampleRateHz(),
+                           sizeof(int16_t) * decoded_samples.size(),
+                           decoded_samples.data(), &speech_type));
+}
+
+class SplitIlbcTest : public ::testing::TestWithParam<std::pair<int, int> > {
+ protected:
+  virtual void SetUp() {
+    const std::pair<int, int> parameters = GetParam();
+    num_frames_ = parameters.first;
+    frame_length_ms_ = parameters.second;
+    frame_length_bytes_ = (frame_length_ms_ == 20) ? 38 : 50;
+  }
+  size_t num_frames_;
+  int frame_length_ms_;
+  size_t frame_length_bytes_;
+};
+
+TEST_P(SplitIlbcTest, NumFrames) {
+  AudioDecoderIlbcImpl decoder;
+  const size_t frame_length_samples = frame_length_ms_ * 8;
+  const auto generate_payload = [](size_t payload_length_bytes) {
+    rtc::Buffer payload(payload_length_bytes);
+    // Fill payload with increasing integers {0, 1, 2, ...}.
+    for (size_t i = 0; i < payload.size(); ++i) {
+      payload[i] = static_cast<uint8_t>(i);
+    }
+    return payload;
+  };
+
+  const auto results = decoder.ParsePayload(
+      generate_payload(frame_length_bytes_ * num_frames_), 0);
+  EXPECT_EQ(num_frames_, results.size());
+
+  size_t frame_num = 0;
+  uint8_t payload_value = 0;
+  for (const auto& result : results) {
+    EXPECT_EQ(frame_length_samples * frame_num, result.timestamp);
+    const LegacyEncodedAudioFrame* frame =
+        static_cast<const LegacyEncodedAudioFrame*>(result.frame.get());
+    const rtc::Buffer& payload = frame->payload();
+    EXPECT_EQ(frame_length_bytes_, payload.size());
+    for (size_t i = 0; i < payload.size(); ++i, ++payload_value) {
+      EXPECT_EQ(payload_value, payload[i]);
+    }
+    ++frame_num;
+  }
+}
+
+// Test 1 through 5 frames of 20 and 30 ms size.
+// Also test the maximum number of frames in one packet for 20 and 30 ms.
+// The maximum is defined by the largest payload length that can be uniquely
+// resolved to a frame size of either 38 bytes (20 ms) or 50 bytes (30 ms).
+INSTANTIATE_TEST_SUITE_P(
+    IlbcTest,
+    SplitIlbcTest,
+    ::testing::Values(std::pair<int, int>(1, 20),  // 1 frame, 20 ms.
+                      std::pair<int, int>(2, 20),  // 2 frames, 20 ms.
+                      std::pair<int, int>(3, 20),  // And so on.
+                      std::pair<int, int>(4, 20),
+                      std::pair<int, int>(5, 20),
+                      std::pair<int, int>(24, 20),
+                      std::pair<int, int>(1, 30),
+                      std::pair<int, int>(2, 30),
+                      std::pair<int, int>(3, 30),
+                      std::pair<int, int>(4, 30),
+                      std::pair<int, int>(5, 30),
+                      std::pair<int, int>(18, 30)));
+
+// Test too large payload size.
+TEST(IlbcTest, SplitTooLargePayload) {
+  AudioDecoderIlbcImpl decoder;
+  constexpr size_t kPayloadLengthBytes = 950;
+  const auto results =
+      decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0);
+  EXPECT_TRUE(results.empty());
+}
+
+// Payload not an integer number of frames.
+TEST(IlbcTest, SplitUnevenPayload) {
+  AudioDecoderIlbcImpl decoder;
+  constexpr size_t kPayloadLengthBytes = 39;  // Not an even number of frames.
+  const auto results =
+      decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0);
+  EXPECT_TRUE(results.empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c
new file mode 100644
index 0000000000..d78f81a897
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvDec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_IndexConvDec(
+    int16_t *index   /* (i/o) Codebook indexes */
+                                ){
+  int k;
+
+  for (k=4;k<6;k++) {
+    /* Readjust the second and third codebook index for the first 40 sample
+       so that they look the same as the first (in terms of lag)
+    */
+    if ((index[k]>=44)&&(index[k]<108)) {
+      index[k]+=64;
+    } else if ((index[k]>=108)&&(index[k]<128)) {
+      index[k]+=128;
+    } else {
+      /* ERROR */
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h
new file mode 100644
index 0000000000..4f08ce04df
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvDec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_IndexConvDec(int16_t* index /* (i/o) Codebook indexes */
+                                );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c
new file mode 100644
index 0000000000..83144150b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ IiLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvEnc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/index_conv_enc.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Convert the codebook indexes to make the search easier
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_IndexConvEnc(
+    int16_t *index   /* (i/o) Codebook indexes */
+                                ){
+  int k;
+
+  for (k=4;k<6;k++) {
+    /* Readjust the second and third codebook index so that it is
+       packetized into 7 bits (before it was put in lag-wise the same
+       way as for the first codebook which uses 8 bits)
+    */
+    if ((index[k]>=108)&&(index[k]<172)) {
+      index[k]-=64;
+    } else if (index[k]>=236) {
+      index[k]-=128;
+    } else {
+      /* ERROR */
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h
new file mode 100644
index 0000000000..4fbf98084e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvEnc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Convert the codebook indexes to make the search easier
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_IndexConvEnc(int16_t* index /* (i/o) Codebook indexes */
+                                );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c
new file mode 100644
index 0000000000..3eb41e33b0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+	iLBC Speech Coder ANSI-C Source Code
+
+	WebRtcIlbcfix_InitDecode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of decoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
+    IlbcDecoder *iLBCdec_inst,  /* (i/o) Decoder instance */
+    int16_t mode,  /* (i) frame size mode */
+    int use_enhancer) {  /* (i) 1: use enhancer, 0: no enhancer */
+  int i;
+
+  iLBCdec_inst->mode = mode;
+
+  /* Set all the variables that are dependent on the frame size mode */
+  if (mode==30) {
+    iLBCdec_inst->blockl = BLOCKL_30MS;
+    iLBCdec_inst->nsub = NSUB_30MS;
+    iLBCdec_inst->nasub = NASUB_30MS;
+    iLBCdec_inst->lpc_n = LPC_N_30MS;
+    iLBCdec_inst->no_of_bytes = NO_OF_BYTES_30MS;
+    iLBCdec_inst->no_of_words = NO_OF_WORDS_30MS;
+    iLBCdec_inst->state_short_len=STATE_SHORT_LEN_30MS;
+  }
+  else if (mode==20) {
+    iLBCdec_inst->blockl = BLOCKL_20MS;
+    iLBCdec_inst->nsub = NSUB_20MS;
+    iLBCdec_inst->nasub = NASUB_20MS;
+    iLBCdec_inst->lpc_n = LPC_N_20MS;
+    iLBCdec_inst->no_of_bytes = NO_OF_BYTES_20MS;
+    iLBCdec_inst->no_of_words = NO_OF_WORDS_20MS;
+    iLBCdec_inst->state_short_len=STATE_SHORT_LEN_20MS;
+  }
+  else {
+    return(-1);
+  }
+
+  /* Reset all the previous LSF to mean LSF */
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+
+  /* Clear the synthesis filter memory */
+  WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
+
+  /* Set the old synthesis filter to {1.0 0.0 ... 0.0} */
+  WebRtcSpl_MemSetW16(iLBCdec_inst->old_syntdenum, 0, ((LPC_FILTERORDER + 1)*NSUB_MAX));
+  for (i=0; i<NSUB_MAX; i++) {
+    iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)] = 4096;
+  }
+
+  /* Clear the variables that are used for the PLC */
+  iLBCdec_inst->last_lag = 20;
+  iLBCdec_inst->consPLICount = 0;
+  iLBCdec_inst->prevPLI = 0;
+  iLBCdec_inst->perSquare = 0;
+  iLBCdec_inst->prevLag = 120;
+  iLBCdec_inst->prevLpc[0] = 4096;
+  WebRtcSpl_MemSetW16(iLBCdec_inst->prevLpc+1, 0, LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(iLBCdec_inst->prevResidual, 0, BLOCKL_MAX);
+
+  /* Initialize the seed for the random number generator */
+  iLBCdec_inst->seed = 777;
+
+  /* Set the filter state of the HP filter to 0 */
+  WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
+  WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
+
+  /* Set the variables that are used in the ehnahcer */
+  iLBCdec_inst->use_enhancer = use_enhancer;
+  WebRtcSpl_MemSetW16(iLBCdec_inst->enh_buf, 0, (ENH_BUFL+ENH_BUFL_FILTEROVERHEAD));
+  for (i=0;i<ENH_NBLOCKS_TOT;i++) {
+    iLBCdec_inst->enh_period[i]=160; /* Q(-4) */
+  }
+
+  iLBCdec_inst->prev_enh_pl = 0;
+
+  return (int)(iLBCdec_inst->blockl);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.h
new file mode 100644
index 0000000000..a2b7b91287
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitDecode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of decoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitDecode(/* (o) Number of decoded samples */
+                             IlbcDecoder*
+                                 iLBCdec_inst, /* (i/o) Decoder instance */
+                             int16_t mode,     /* (i) frame size mode */
+                             int use_enhancer  /* (i) 1 to use enhancer
+                                                  0 to run without enhancer */
+                             );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c
new file mode 100644
index 0000000000..aa858e94bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitEncode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/init_encode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of encoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
+    IlbcEncoder *iLBCenc_inst,  /* (i/o) Encoder instance */
+    int16_t mode) {  /* (i) frame size mode */
+  iLBCenc_inst->mode = mode;
+
+  /* Set all the variables that are dependent on the frame size mode */
+  if (mode==30) {
+    iLBCenc_inst->blockl = BLOCKL_30MS;
+    iLBCenc_inst->nsub = NSUB_30MS;
+    iLBCenc_inst->nasub = NASUB_30MS;
+    iLBCenc_inst->lpc_n = LPC_N_30MS;
+    iLBCenc_inst->no_of_bytes = NO_OF_BYTES_30MS;
+    iLBCenc_inst->no_of_words = NO_OF_WORDS_30MS;
+    iLBCenc_inst->state_short_len=STATE_SHORT_LEN_30MS;
+  }
+  else if (mode==20) {
+    iLBCenc_inst->blockl = BLOCKL_20MS;
+    iLBCenc_inst->nsub = NSUB_20MS;
+    iLBCenc_inst->nasub = NASUB_20MS;
+    iLBCenc_inst->lpc_n = LPC_N_20MS;
+    iLBCenc_inst->no_of_bytes = NO_OF_BYTES_20MS;
+    iLBCenc_inst->no_of_words = NO_OF_WORDS_20MS;
+    iLBCenc_inst->state_short_len=STATE_SHORT_LEN_20MS;
+  }
+  else {
+    return(-1);
+  }
+
+  /* Clear the buffers and set the previous LSF and LSP to the mean value */
+  WebRtcSpl_MemSetW16(iLBCenc_inst->anaMem, 0, LPC_FILTERORDER);
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(iLBCenc_inst->lpc_buffer, 0, LPC_LOOKBACK + BLOCKL_MAX);
+
+  /* Set the filter state of the HP filter to 0 */
+  WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemx, 0, 2);
+  WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemy, 0, 4);
+
+#ifdef SPLIT_10MS
+  /*Zeroing the past samples for 10msec Split*/
+  WebRtcSpl_MemSetW16(iLBCenc_inst->past_samples,0,160);
+  iLBCenc_inst->section = 0;
+#endif
+
+  return (int)(iLBCenc_inst->no_of_bytes);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.h
new file mode 100644
index 0000000000..4ada6a30c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitEncode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of encoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitEncode(/* (o) Number of bytes encoded */
+                             IlbcEncoder*
+                                 iLBCenc_inst, /* (i/o) Encoder instance */
+                             int16_t mode      /* (i) frame size mode */
+                             );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c
new file mode 100644
index 0000000000..17ed244bd4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Interpolate.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  interpolation between vectors
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Interpolate(
+    int16_t *out, /* (o) output vector */
+    int16_t *in1, /* (i) first input vector */
+    int16_t *in2, /* (i) second input vector */
+    int16_t coef, /* (i) weight coefficient in Q14 */
+    int16_t length)  /* (i) number of sample is vectors */
+{
+  int i;
+  int16_t invcoef;
+
+  /*
+    Performs the operation out[i] = in[i]*coef + (1-coef)*in2[i] (with rounding)
+  */
+
+  invcoef = 16384 - coef; /* 16384 = 1.0 (Q14)*/
+  for (i = 0; i < length; i++) {
+    out[i] = (int16_t)((coef * in1[i] + invcoef * in2[i] + 8192) >> 14);
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.h
new file mode 100644
index 0000000000..892082b75c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Interpolate.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  interpolation between vectors
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Interpolate(
+    int16_t* out,    /* (o) output vector */
+    int16_t* in1,    /* (i) first input vector */
+    int16_t* in2,    /* (i) second input vector */
+    int16_t coef,    /* (i) weight coefficient in Q14 */
+    int16_t length); /* (i) number of sample is vectors */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
new file mode 100644
index 0000000000..6dddd6fb86
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InterpolateSamples.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/interpolate_samples.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_InterpolateSamples(
+    int16_t *interpSamples, /* (o) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    size_t lMem    /* (i) Length of the CB memory */
+                                      ) {
+  int16_t *ppi, *ppo, i, j, temp1, temp2;
+  int16_t *tmpPtr;
+
+  /* Calculate the 20 vectors of interpolated samples (4 samples each)
+     that are used in the codebooks for lag 20 to 39 */
+  tmpPtr = interpSamples;
+  for (j=0; j<20; j++) {
+    temp1 = 0;
+    temp2 = 3;
+    ppo = CBmem+lMem-4;
+    ppi = CBmem+lMem-j-24;
+    for (i=0; i<4; i++) {
+
+      *tmpPtr++ = (int16_t)((WebRtcIlbcfix_kAlpha[temp2] * *ppo) >> 15) +
+          (int16_t)((WebRtcIlbcfix_kAlpha[temp1] * *ppi) >> 15);
+
+      ppo++;
+      ppi++;
+      temp1++;
+      temp2--;
+    }
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
new file mode 100644
index 0000000000..bc665d7854
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InterpolateSamples.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Construct the interpolated samples for the Augmented CB
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_InterpolateSamples(
+    int16_t* interpSamples, /* (o) The interpolated samples */
+    int16_t* CBmem,         /* (i) The CB memory */
+    size_t lMem             /* (i) Length of the CB memory */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
new file mode 100644
index 0000000000..89f6d29724
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LpcEncode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lpc_encode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+#include "modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h"
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_quant.h"
+
+/*----------------------------------------------------------------*
+ *  lpc encoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LpcEncode(
+    int16_t *syntdenum,  /* (i/o) synthesis filter coefficients
+                                           before/after encoding */
+    int16_t *weightdenum, /* (i/o) weighting denumerator coefficients
+                                   before/after encoding */
+    int16_t *lsf_index,  /* (o) lsf quantization index */
+    int16_t *data,   /* (i) Speech to do LPC analysis on */
+    IlbcEncoder *iLBCenc_inst
+    /* (i/o) the encoder state structure */
+                              ) {
+  /* Stack based */
+  int16_t lsf[LPC_FILTERORDER * LPC_N_MAX];
+  int16_t lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
+
+  /* Calculate LSF's from the input speech */
+  WebRtcIlbcfix_SimpleLpcAnalysis(lsf, data, iLBCenc_inst);
+
+  /* Quantize the LSF's */
+  WebRtcIlbcfix_SimpleLsfQ(lsfdeq, lsf_index, lsf, iLBCenc_inst->lpc_n);
+
+  /* Stableize the LSF's if needed */
+  WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCenc_inst->lpc_n);
+
+  /* Calculate the synthesis and weighting filter coefficients from
+     the optimal LSF and the dequantized LSF */
+  WebRtcIlbcfix_SimpleInterpolateLsf(syntdenum, weightdenum,
+                                     lsf, lsfdeq, iLBCenc_inst->lsfold,
+                                     iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
new file mode 100644
index 0000000000..a67b77acbf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LpcEncode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc encoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LpcEncode(
+    int16_t* syntdenum,   /* (i/o) synthesis filter coefficients
+                                   before/after encoding */
+    int16_t* weightdenum, /* (i/o) weighting denumerator coefficients
+                                   before/after encoding */
+    int16_t* lsf_index,   /* (o) lsf quantization index */
+    int16_t* data,        /* (i) Speech to do LPC analysis on */
+    IlbcEncoder* iLBCenc_inst
+    /* (i/o) the encoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c
new file mode 100644
index 0000000000..9f0e19a2d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfCheck.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  check for stability of lsf coefficients
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_LsfCheck(
+    int16_t *lsf, /* LSF parameters */
+    int dim, /* dimension of LSF */
+    int NoAn)  /* No of analysis per frame */
+{
+  int k,n,m, Nit=2, change=0,pos;
+  const int16_t eps=319;  /* 0.039 in Q13 (50 Hz)*/
+  const int16_t eps2=160;  /* eps/2.0 in Q13;*/
+  const int16_t maxlsf=25723; /* 3.14; (4000 Hz)*/
+  const int16_t minlsf=82;  /* 0.01; (0 Hz)*/
+
+  /* LSF separation check*/
+  for (n=0;n<Nit;n++) {  /* Run through a 2 times */
+    for (m=0;m<NoAn;m++) { /* Number of analyses per frame */
+      for (k=0;k<(dim-1);k++) {
+        pos=m*dim+k;
+
+        /* Seperate coefficients with a safety margin of 50 Hz */
+        if ((lsf[pos+1]-lsf[pos])<eps) {
+
+          if (lsf[pos+1]<lsf[pos]) {
+            lsf[pos+1]= lsf[pos]+eps2;
+            lsf[pos]= lsf[pos+1]-eps2;
+          } else {
+            lsf[pos]-=eps2;
+            lsf[pos+1]+=eps2;
+          }
+          change=1;
+        }
+
+        /* Limit minimum and maximum LSF */
+        if (lsf[pos]<minlsf) {
+          lsf[pos]=minlsf;
+          change=1;
+        }
+
+        if (lsf[pos]>maxlsf) {
+          lsf[pos]=maxlsf;
+          change=1;
+        }
+      }
+    }
+  }
+
+  return change;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.h
new file mode 100644
index 0000000000..9ba90a31e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfCheck.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  check for stability of lsf coefficients
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_LsfCheck(int16_t* lsf, /* LSF parameters */
+                           int dim,      /* dimension of LSF */
+                           int NoAn);    /* No of analysis per frame */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
new file mode 100644
index 0000000000..04de5e7e6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LspInterpolate2PolyDec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
+
+/*----------------------------------------------------------------*
+ *  interpolation of lsf coefficients for the decoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LspInterpolate2PolyDec(
+    int16_t *a,   /* (o) lpc coefficients Q12 */
+    int16_t *lsf1,  /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2,  /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
+                                   lsf1 and lsf2 Q14 */
+    int16_t length  /* (i) length of coefficient vectors */
+                                          ){
+  int16_t lsftmp[LPC_FILTERORDER];
+
+  /* interpolate LSF */
+  WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
+
+  /* Compute the filter coefficients from the LSF */
+  WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
new file mode 100644
index 0000000000..6cc9d9746d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LspInterpolate2PolyDec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  interpolation of lsf coefficients for the decoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LspInterpolate2PolyDec(
+    int16_t* a,    /* (o) lpc coefficients Q12 */
+    int16_t* lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t* lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
+                                   lsf1 and lsf2 Q14 */
+    int16_t length /* (i) length of coefficient vectors */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
new file mode 100644
index 0000000000..618821216c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator and conversion from lsf to a coefficients
+ *  (subrutine to SimpleInterpolateLSF)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
+    int16_t *a,  /* (o) lpc coefficients Q12 */
+    int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef, /* (i) weighting coefficient to use between
+                           lsf1 and lsf2 Q14 */
+    int16_t length /* (i) length of coefficient vectors */
+                                          ) {
+  /* Stack based */
+  int16_t lsftmp[LPC_FILTERORDER];
+
+  /* interpolate LSF */
+  WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
+
+  /* Compute the filter coefficients from the LSF */
+  WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
new file mode 100644
index 0000000000..b278a10f4b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator and conversion from lsf to a coefficients
+ *  (subrutine to SimpleInterpolateLSF)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
+    int16_t* a,    /* (o) lpc coefficients Q12 */
+    int16_t* lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t* lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
+                            lsf1 and lsf2 Q14 */
+    int16_t length /* (i) length of coefficient vectors */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
new file mode 100644
index 0000000000..ee8292f394
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Lsp.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_to_lsp.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  conversion from lsf to lsp coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Lsp(
+    int16_t *lsf, /* (i) lsf in Q13 values between 0 and pi */
+    int16_t *lsp, /* (o) lsp in Q15 values between -1 and 1 */
+    int16_t m  /* (i) number of coefficients */
+                           ) {
+  int16_t i, k;
+  int16_t diff; /* difference, which is used for the
+                           linear approximation (Q8) */
+  int16_t freq; /* normalized frequency in Q15 (0..1) */
+  int32_t tmpW32;
+
+  for(i=0; i<m; i++)
+  {
+    freq = (int16_t)((lsf[i] * 20861) >> 15);
+    /* 20861: 1.0/(2.0*PI) in Q17 */
+    /*
+       Upper 8 bits give the index k and
+       Lower 8 bits give the difference, which needs
+       to be approximated linearly
+    */
+    k = freq >> 8;
+    diff = (freq&0x00ff);
+
+    /* Guard against getting outside table */
+
+    if (k>63) {
+      k = 63;
+    }
+
+    /* Calculate linear approximation */
+    tmpW32 = WebRtcIlbcfix_kCosDerivative[k] * diff;
+    lsp[i] = WebRtcIlbcfix_kCos[k] + (int16_t)(tmpW32 >> 12);
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
new file mode 100644
index 0000000000..6bc6c44dbd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Lsp.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  conversion from lsf to lsp coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Lsp(
+    int16_t* lsf, /* (i) lsf in Q13 values between 0 and pi */
+    int16_t* lsp, /* (o) lsp in Q15 values between -1 and 1 */
+    int16_t m     /* (i) number of coefficients */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
new file mode 100644
index 0000000000..8ca91d82f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Poly.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/get_lsp_poly.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_to_lsp.h"
+
+void WebRtcIlbcfix_Lsf2Poly(
+    int16_t *a,     /* (o) predictor coefficients (order = 10) in Q12 */
+    int16_t *lsf    /* (i) line spectral frequencies in Q13 */
+                            ) {
+  int32_t f[2][6]; /* f[0][] and f[1][] corresponds to
+                            F1(z) and F2(z) respectivly */
+  int32_t *f1ptr, *f2ptr;
+  int16_t *a1ptr, *a2ptr;
+  int32_t tmpW32;
+  int16_t lsp[10];
+  int i;
+
+  /* Convert lsf to lsp */
+  WebRtcIlbcfix_Lsf2Lsp(lsf, lsp, LPC_FILTERORDER);
+
+  /* Get F1(z) and F2(z) from the lsp */
+  f1ptr=f[0];
+  f2ptr=f[1];
+  WebRtcIlbcfix_GetLspPoly(&lsp[0],f1ptr);
+  WebRtcIlbcfix_GetLspPoly(&lsp[1],f2ptr);
+
+  /* for i = 5 down to 1
+     Compute f1[i] += f1[i-1];
+     and     f2[i] += f2[i-1];
+  */
+  f1ptr=&f[0][5];
+  f2ptr=&f[1][5];
+  for (i=5; i>0; i--)
+  {
+    (*f1ptr) += (*(f1ptr-1));
+    (*f2ptr) -= (*(f2ptr-1));
+    f1ptr--;
+    f2ptr--;
+  }
+
+  /* Get the A(z) coefficients
+     a[0] = 1.0
+     for i = 1 to 5
+     a[i] = (f1[i] + f2[i] + round)>>13;
+     for i = 1 to 5
+     a[11-i] = (f1[i] - f2[i] + round)>>13;
+  */
+  a[0]=4096;
+  a1ptr=&a[1];
+  a2ptr=&a[10];
+  f1ptr=&f[0][1];
+  f2ptr=&f[1][1];
+  for (i=5; i>0; i--)
+  {
+    tmpW32 = (*f1ptr) + (*f2ptr);
+    *a1ptr = (int16_t)((tmpW32 + 4096) >> 13);
+
+    tmpW32 = (*f1ptr) - (*f2ptr);
+    *a2ptr = (int16_t)((tmpW32 + 4096) >> 13);
+
+    a1ptr++;
+    a2ptr--;
+    f1ptr++;
+    f2ptr++;
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
new file mode 100644
index 0000000000..f26d3a8d2d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Poly.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Convert from LSF coefficients to A coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Poly(
+    int16_t* a,  /* (o) predictor coefficients (order = 10) in Q12 */
+    int16_t* lsf /* (i) line spectral frequencies in Q13 */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
new file mode 100644
index 0000000000..227f4d45b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsp2Lsf.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsp_to_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  conversion from LSP coefficients to LSF coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsp2Lsf(
+    int16_t *lsp, /* (i) lsp vector -1...+1 in Q15 */
+    int16_t *lsf, /* (o) Lsf vector 0...Pi in Q13
+                           (ordered, so that lsf[i]<lsf[i+1]) */
+    int16_t m  /* (i) Number of coefficients */
+                           )
+{
+  int16_t i, k;
+  int16_t diff; /* diff between table value and desired value (Q15) */
+  int16_t freq; /* lsf/(2*pi) (Q16) */
+  int16_t *lspPtr, *lsfPtr, *cosTblPtr;
+  int16_t tmp;
+
+  /* set the index to maximum index value in WebRtcIlbcfix_kCos */
+  k = 63;
+
+  /*
+     Start with the highest LSP and then work the way down
+     For each LSP the lsf is calculated by first order approximation
+     of the acos(x) function
+  */
+  lspPtr = &lsp[9];
+  lsfPtr = &lsf[9];
+  cosTblPtr=(int16_t*)&WebRtcIlbcfix_kCos[k];
+  for(i=m-1; i>=0; i--)
+  {
+    /*
+       locate value in the table, which is just above lsp[i],
+       basically an approximation to acos(x)
+    */
+    while( (((int32_t)(*cosTblPtr)-(*lspPtr)) < 0)&&(k>0) )
+    {
+      k-=1;
+      cosTblPtr--;
+    }
+
+    /* Calculate diff, which is used in the linear approximation of acos(x) */
+    diff = (*lspPtr)-(*cosTblPtr);
+
+    /*
+       The linear approximation of acos(lsp[i]) :
+       acos(lsp[i])= k*512 + (WebRtcIlbcfix_kAcosDerivative[ind]*offset >> 11)
+    */
+
+    /* tmp (linear offset) in Q16 */
+    tmp = (int16_t)((WebRtcIlbcfix_kAcosDerivative[k] * diff) >> 11);
+
+    /* freq in Q16 */
+    freq = (k << 9) + tmp;
+
+    /* lsf = freq*2*pi */
+    (*lsfPtr) = (int16_t)(((int32_t)freq*25736)>>15);
+
+    lsfPtr--;
+    lspPtr--;
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
new file mode 100644
index 0000000000..c2f4b7692d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsp2Lsf.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  conversion from LSP coefficients to LSF coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsp2Lsf(
+    int16_t* lsp, /* (i) lsp vector -1...+1 in Q15 */
+    int16_t* lsf, /* (o) Lsf vector 0...Pi in Q13
+                           (ordered, so that lsf[i]<lsf[i+1]) */
+    int16_t m     /* (i) Number of coefficients */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c
new file mode 100644
index 0000000000..9b870e0ef0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_MyCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/my_corr.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * compute cross correlation between sequences
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_MyCorr(
+    int32_t* corr,  /* (o) correlation of seq1 and seq2 */
+    const int16_t* seq1,  /* (i) first sequence */
+    size_t dim1,  /* (i) dimension first seq1 */
+    const int16_t* seq2, /* (i) second sequence */
+    size_t dim2   /* (i) dimension seq2 */
+                          ){
+  uint32_t max1, max2;
+  size_t loops;
+  int right_shift;
+
+  // Calculate a right shift that will let us sum dim2 pairwise products of
+  // values from the two sequences without overflowing an int32_t. (The +1 in
+  // max1 and max2 are because WebRtcSpl_MaxAbsValueW16 will return 2**15 - 1
+  // if the input array contains -2**15.)
+  max1 = WebRtcSpl_MaxAbsValueW16(seq1, dim1) + 1;
+  max2 = WebRtcSpl_MaxAbsValueW16(seq2, dim2) + 1;
+  right_shift =
+      (64 - 31) - WebRtcSpl_CountLeadingZeros64((max1 * max2) * (uint64_t)dim2);
+  if (right_shift < 0) {
+    right_shift = 0;
+  }
+
+  loops=dim1-dim2+1;
+
+  /* Calculate the cross correlations */
+  WebRtcSpl_CrossCorrelation(corr, seq2, seq1, dim2, loops, right_shift, 1);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.h
new file mode 100644
index 0000000000..c0c2fa4a48
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_MyCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * compute cross correlation between sequences
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_MyCorr(int32_t* corr, /* (o) correlation of seq1 and seq2 */
+                          const int16_t* seq1, /* (i) first sequence */
+                          size_t dim1,         /* (i) dimension first seq1 */
+                          const int16_t* seq2, /* (i) second sequence */
+                          size_t dim2          /* (i) dimension seq2 */
+                          );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
new file mode 100644
index 0000000000..1ecdd96d5a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_NearestNeighbor.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/nearest_neighbor.h"
+
+void WebRtcIlbcfix_NearestNeighbor(size_t* index,
+                                   const size_t* array,
+                                   size_t value,
+                                   size_t arlength) {
+  size_t i;
+  size_t min_diff = (size_t)-1;
+  for (i = 0; i < arlength; i++) {
+    const size_t diff =
+        (array[i] < value) ? (value - array[i]) : (array[i] - value);
+    if (diff < min_diff) {
+      *index = i;
+      min_diff = diff;
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
new file mode 100644
index 0000000000..704cf2a37d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_NearestNeighbor.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * Find index in array such that the array element with said
+ * index is the element of said array closest to "value"
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_NearestNeighbor(
+    size_t* index,       /* (o) index of array element closest to value */
+    const size_t* array, /* (i) data array (Q2) */
+    size_t value,        /* (i) value (Q2) */
+    size_t arlength      /* (i) dimension of data array (==ENH_NBLOCKS_TOT) */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c
new file mode 100644
index 0000000000..dd44eb8fb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c
@@ -0,0 +1,253 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_PackBits.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/pack_bits.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_PackBits(
+    uint16_t *bitstream,   /* (o) The packetized bitstream */
+    iLBC_bits *enc_bits,  /* (i) Encoded bits */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
+                             ){
+  uint16_t *bitstreamPtr;
+  int i, k;
+  int16_t *tmpPtr;
+
+  bitstreamPtr=bitstream;
+
+  /* Class 1 bits of ULP */
+  /* First int16_t */
+  (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[0])<<10;   /* Bit 0..5  */
+  (*bitstreamPtr) |= (enc_bits->lsf[1])<<3;     /* Bit 6..12 */
+  (*bitstreamPtr) |= (enc_bits->lsf[2]&0x70)>>4;    /* Bit 13..15 */
+  bitstreamPtr++;
+  /* Second int16_t */
+  (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[2]&0xF)<<12;  /* Bit 0..3  */
+
+  if (mode==20) {
+    (*bitstreamPtr) |= (enc_bits->startIdx)<<10;    /* Bit 4..5  */
+    (*bitstreamPtr) |= (enc_bits->state_first)<<9;    /* Bit 6  */
+    (*bitstreamPtr) |= (enc_bits->idxForMax)<<3;    /* Bit 7..12 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[0])&0x70)>>4;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    (*bitstreamPtr) = ((enc_bits->cb_index[0])&0xE)<<12;  /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x18)<<8;  /* Bit 3..4  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x8)<<7;  /* Bit 5  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0xFE)<<2;  /* Bit 6..12 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x10)>>2;  /* Bit 13  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x8)>>2;  /* Bit 14  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x10)>>4;  /* Bit 15  */
+  } else { /* mode==30 */
+    (*bitstreamPtr) |= (enc_bits->lsf[3])<<6;     /* Bit 4..9  */
+    (*bitstreamPtr) |= (enc_bits->lsf[4]&0x7E)>>1;    /* Bit 10..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[4]&0x1)<<15;  /* Bit 0  */
+    (*bitstreamPtr) |= (enc_bits->lsf[5])<<8;     /* Bit 1..7  */
+    (*bitstreamPtr) |= (enc_bits->startIdx)<<5;     /* Bit 8..10 */
+    (*bitstreamPtr) |= (enc_bits->state_first)<<4;    /* Bit 11  */
+    (*bitstreamPtr) |= ((enc_bits->idxForMax)&0x3C)>>2;   /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 4:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->idxForMax&0x3)<<14; /* Bit 0..1  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x78)<<7;   /* Bit 2..5  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x10)<<5;  /* Bit 6  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x8)<<5;  /* Bit 7  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[3]&0xFC);   /* Bit 8..13 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[3]&0x10)>>3;  /* Bit 14  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x8)>>3;  /* Bit 15  */
+  }
+  /* Class 2 bits of ULP */
+  /* 4:th to 6:th int16_t for 20 ms case
+     5:th to 7:th int16_t for 30 ms case */
+  bitstreamPtr++;
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<3; k++) {
+    (*bitstreamPtr) = 0;
+    for (i=15; i>=0; i--) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 7:th int16_t */
+    (*bitstreamPtr) = 0;
+    for (i=15; i>6; i--) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4)<<4;  /* Bit 9  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<2;  /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x4)<<1;  /* Bit 12  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x8)>>1;  /* Bit 13  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)>>2;  /* Bit 14..15 */
+
+  } else { /* mode==30 */
+    /* 8:th int16_t */
+    (*bitstreamPtr) = 0;
+    for (i=15; i>5; i--) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x6)<<3;   /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x8);   /* Bit 12  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4);   /* Bit 13  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[3]&0x2);    /* Bit 14  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[6]&0x80)>>7;   /* Bit 15  */
+    bitstreamPtr++;
+    /* 9:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->cb_index[6]&0x7E)<<9;/* Bit 0..5  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[9]&0xFE)<<2;   /* Bit 6..12 */
+    (*bitstreamPtr) |= (enc_bits->cb_index[12]&0xE0)>>5;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 10:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->cb_index[12]&0x1E)<<11;/* Bit 0..3 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<8;  /* Bit 4..5  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x6)<<7;  /* Bit 6..7  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x18)<<3;  /* Bit 8..9  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)<<2;  /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[9]&0x10)>>1;  /* Bit 12  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[10]&0x8)>>1;  /* Bit 13  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[12]&0x10)>>3;  /* Bit 14  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[13]&0x8)>>3;  /* Bit 15  */
+  }
+  bitstreamPtr++;
+  /* Class 3 bits of ULP */
+  /*  8:th to 14:th int16_t for 20 ms case
+      11:th to 17:th int16_t for 30 ms case */
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<7; k++) {
+    (*bitstreamPtr) = 0;
+    for (i=14; i>=0; i-=2) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x3))<<i; /* Bit 15-i..14-i*/
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 15:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+    (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<13;  /* Bit 2  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<6;   /* Bit 3..9  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x7E)>>1;  /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 16:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[2])&0x1))<<15;
+    /* Bit 0  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<12;  /* Bit 1..3  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<10;  /* Bit 4..5  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[2]))<<7;   /* Bit 6..8  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<6;  /* Bit 9  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x7E)>>1;  /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 17:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[4])&0x1))<<15;
+    /* Bit 0  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[5])<<8;    /* Bit 1..7  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[6]);     /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 18:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[7]))<<8; /* Bit 0..7  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[8]);     /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->gain_index[3])&0x3))<<14;
+    /* Bit 0..1  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x3)<<12;  /* Bit 2..3  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[5]))<<9;   /* Bit 4..6  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<6;  /* Bit 7..9  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<4;  /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[8])<<1;   /* Bit 12..14 */
+  } else { /* mode==30 */
+    /* 18:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+    (*bitstreamPtr) |= (((enc_bits->idxVec[57])&0x3))<<12;  /* Bit 2..3  */
+    (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<11;  /* Bit 4  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<4;   /* Bit 5..11 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x78)>>3;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[2])&0x7)<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<10;  /* Bit 3..5  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<8;  /* Bit 6..7  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[2])&0x7)<<5;  /* Bit 8..10 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<4;  /* Bit 11  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x78)>>3;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 20:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[4])&0x7)<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[5]))<<6;   /* Bit 3..9  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[6])&0x1)<<5;  /* Bit 10  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[7])&0xF8)>>3;  /* Bit 11..15 */
+    bitstreamPtr++;
+    /* 21:st int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[7])&0x7)<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[8]))<<5;   /* Bit 3..10 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[9])&0x1)<<4;  /* Bit 11  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[10])&0xF0)>>4;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 22:nd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[10])&0xF)<<12;
+    /* Bit 0..3  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[11]))<<4;   /* Bit 4..11 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[12])&0x1)<<3;  /* Bit 12  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[13])&0xE0)>>5;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 23:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[13])&0x1F)<<11;
+    /* Bit 0..4  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[14]))<<3;   /* Bit 5..12 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x3)<<1;  /* Bit 13..14 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x1);   /* Bit 15  */
+    bitstreamPtr++;
+    /* 24:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->gain_index[5]))<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<10;  /* Bit 3..5  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<8;  /* Bit 6..7  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[8]))<<5;   /* Bit 8..10 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[9])&0xF)<<1;  /* Bit 11..14 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[10])&0x4)>>2;  /* Bit 15  */
+    bitstreamPtr++;
+    /* 25:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->gain_index[10])&0x3)<<14;
+    /* Bit 0..1  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[11]))<<11;  /* Bit 2..4  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[12])&0xF)<<7;  /* Bit 5..8  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[13])&0x7)<<4;  /* Bit 9..11 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[14]))<<1;   /* Bit 12..14 */
+  }
+  /* Last bit is automatically zero */
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.h
new file mode 100644
index 0000000000..8dcf41ce08
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_PackBits.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_PackBits(
+    uint16_t* bitstream, /* (o) The packetized bitstream */
+    iLBC_bits* enc_bits, /* (i) Encoded bits */
+    int16_t mode         /* (i) Codec mode (20 or 30) */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
new file mode 100644
index 0000000000..7192eaab49
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsf.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/lsp_to_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsp.h"
+
+void WebRtcIlbcfix_Poly2Lsf(
+    int16_t *lsf,   /* (o) lsf coefficients (Q13) */
+    int16_t *a    /* (i) A coefficients (Q12) */
+                            ) {
+  int16_t lsp[10];
+  WebRtcIlbcfix_Poly2Lsp(a, lsp, (int16_t*)WebRtcIlbcfix_kLspMean);
+  WebRtcIlbcfix_Lsp2Lsf(lsp, lsf, 10);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
new file mode 100644
index 0000000000..363e392bb2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsf.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  conversion from lpc coefficients to lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsf(int16_t* lsf, /* (o) lsf coefficients (Q13) */
+                            int16_t* a    /* (i) A coefficients (Q12) */
+                            );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
new file mode 100644
index 0000000000..ad0ecd70ab
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
@@ -0,0 +1,159 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsp.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsp.h"
+
+#include "modules/audio_coding/codecs/ilbc/chebyshev.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+/*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsp coefficients
+ * function is only for 10:th order LPC
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsp(
+    int16_t *a,  /* (o) A coefficients in Q12 */
+    int16_t *lsp, /* (i) LSP coefficients in Q15 */
+    int16_t *old_lsp /* (i) old LSP coefficients that are used if the new
+                              coefficients turn out to be unstable */
+                            ) {
+  int16_t f[2][6]; /* f[0][] represents f1 and f[1][] represents f2 */
+  int16_t *a_i_ptr, *a_10mi_ptr;
+  int16_t *f1ptr, *f2ptr;
+  int32_t tmpW32;
+  int16_t x, y, xlow, ylow, xmid, ymid, xhigh, yhigh, xint;
+  int16_t shifts, sign;
+  int i, j;
+  int foundFreqs;
+  int fi_select;
+
+  /*
+     Calculate the two polynomials f1(z) and f2(z)
+     (the sum and the diff polynomial)
+     f1[0] = f2[0] = 1.0;
+     f1[i+1] = a[i+1] + a[10-i] - f1[i];
+     f2[i+1] = a[i+1] - a[10-i] - f1[i];
+  */
+
+  a_i_ptr = a + 1;
+  a_10mi_ptr = a + 10;
+  f1ptr = f[0];
+  f2ptr = f[1];
+  (*f1ptr) = 1024; /* 1.0 in Q10 */
+  (*f2ptr) = 1024; /* 1.0 in Q10 */
+  for (i = 0; i < 5; i++) {
+    *(f1ptr + 1) =
+        (int16_t)((((int32_t)(*a_i_ptr) + *a_10mi_ptr) >> 2) - *f1ptr);
+    *(f2ptr + 1) =
+        (int16_t)((((int32_t)(*a_i_ptr) - *a_10mi_ptr) >> 2) + *f2ptr);
+    a_i_ptr++;
+    a_10mi_ptr--;
+    f1ptr++;
+    f2ptr++;
+  }
+
+  /*
+    find the LSPs using the Chebychev pol. evaluation
+  */
+
+  fi_select = 0; /* selector between f1 and f2, start with f1 */
+
+  foundFreqs = 0;
+
+  xlow = WebRtcIlbcfix_kCosGrid[0];
+  ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+
+  /*
+     Iterate until all the 10 LSP's have been found or
+     all the grid points have been tried. If the 10 LSP's can
+     not be found, set the LSP vector to previous LSP
+  */
+
+  for (j = 1; j < COS_GRID_POINTS && foundFreqs < 10; j++) {
+    xhigh = xlow;
+    yhigh = ylow;
+    xlow = WebRtcIlbcfix_kCosGrid[j];
+    ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+
+    if (ylow * yhigh <= 0) {
+      /* Run 4 times to reduce the interval */
+      for (i = 0; i < 4; i++) {
+        /* xmid =(xlow + xhigh)/2 */
+        xmid = (xlow >> 1) + (xhigh >> 1);
+        ymid = WebRtcIlbcfix_Chebyshev(xmid, f[fi_select]);
+
+        if (ylow * ymid <= 0) {
+          yhigh = ymid;
+          xhigh = xmid;
+        } else {
+          ylow = ymid;
+          xlow = xmid;
+        }
+      }
+
+      /*
+        Calculater xint by linear interpolation:
+        xint = xlow - ylow*(xhigh-xlow)/(yhigh-ylow);
+      */
+
+      x = xhigh - xlow;
+      y = yhigh - ylow;
+
+      if (y == 0) {
+        xint = xlow;
+      } else {
+        sign = y;
+        y = WEBRTC_SPL_ABS_W16(y);
+        shifts = (int16_t)WebRtcSpl_NormW32(y)-16;
+        y <<= shifts;
+        y = (int16_t)WebRtcSpl_DivW32W16(536838144, y); /* 1/(yhigh-ylow) */
+
+        tmpW32 = (x * y) >> (19 - shifts);
+
+        /* y=(xhigh-xlow)/(yhigh-ylow) */
+        y = (int16_t)(tmpW32&0xFFFF);
+
+        if (sign < 0) {
+          y = -y;
+        }
+        /* tmpW32 = ylow*(xhigh-xlow)/(yhigh-ylow) */
+        tmpW32 = (ylow * y) >> 10;
+        xint = xlow-(int16_t)(tmpW32&0xFFFF);
+      }
+
+      /* Store the calculated lsp */
+      lsp[foundFreqs] = (int16_t)xint;
+      foundFreqs++;
+
+      /* if needed, set xlow and ylow for next recursion */
+      if (foundFreqs<10) {
+        xlow = xint;
+        /* Swap between f1 and f2 (f[0][] and f[1][]) */
+        fi_select = ((fi_select+1)&0x1);
+
+        ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+      }
+    }
+  }
+
+  /* Check if M roots found, if not then use the old LSP */
+  if (foundFreqs < 10) {
+    WEBRTC_SPL_MEMCPY_W16(lsp, old_lsp, 10);
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
new file mode 100644
index 0000000000..928ee4efdb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsp.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsp coefficients
+ * function is only for 10:th order LPC
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsp(
+    int16_t* a,      /* (o) A coefficients in Q12 */
+    int16_t* lsp,    /* (i) LSP coefficients in Q15 */
+    int16_t* old_lsp /* (i) old LSP coefficients that are used if the new
+                              coefficients turn out to be unstable */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c
new file mode 100644
index 0000000000..5bdab7a4b0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Refiner.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/refiner.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/enh_upsample.h"
+#include "modules/audio_coding/codecs/ilbc/my_corr.h"
+
+/*----------------------------------------------------------------*
+ * find segment starting near idata+estSegPos that has highest
+ * correlation with idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
+ * resolution of ENH_UPSO times the original of the original
+ * sampling rate
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Refiner(
+    size_t *updStartPos, /* (o) updated start point (Q-2) */
+    int16_t *idata,   /* (i) original data buffer */
+    size_t idatal,   /* (i) dimension of idata */
+    size_t centerStartPos, /* (i) beginning center segment */
+    size_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
+    int16_t *surround,  /* (i/o) The contribution from this sequence
+                                           summed with earlier contributions */
+    int16_t gain    /* (i) Gain to use for this sequence */
+                           ){
+  size_t estSegPosRounded, searchSegStartPos, searchSegEndPos, corrdim;
+  size_t tloc, tloc2, i;
+
+  int32_t maxtemp, scalefact;
+  int16_t *filtStatePtr, *polyPtr;
+  /* Stack based */
+  int16_t filt[7];
+  int32_t corrVecUps[ENH_CORRDIM*ENH_UPS0];
+  int32_t corrVecTemp[ENH_CORRDIM];
+  int16_t vect[ENH_VECTL];
+  int16_t corrVec[ENH_CORRDIM];
+
+  /* defining array bounds */
+
+  estSegPosRounded = (estSegPos - 2) >> 2;
+
+  searchSegStartPos =
+      (estSegPosRounded < ENH_SLOP) ? 0 : (estSegPosRounded - ENH_SLOP);
+
+  searchSegEndPos = estSegPosRounded + ENH_SLOP;
+  if ((searchSegEndPos + ENH_BLOCKL) >= idatal) {
+    searchSegEndPos = idatal - ENH_BLOCKL - 1;
+  }
+
+  corrdim = searchSegEndPos + 1 - searchSegStartPos;
+
+  /* compute upsampled correlation and find
+     location of max */
+
+  WebRtcIlbcfix_MyCorr(corrVecTemp, idata + searchSegStartPos,
+                       corrdim + ENH_BLOCKL - 1, idata + centerStartPos,
+                       ENH_BLOCKL);
+
+  /* Calculate the rescaling factor for the correlation in order to
+     put the correlation in a int16_t vector instead */
+  maxtemp = WebRtcSpl_MaxAbsValueW32(corrVecTemp, corrdim);
+
+  scalefact = WebRtcSpl_GetSizeInBits(maxtemp) - 15;
+
+  if (scalefact > 0) {
+    for (i = 0; i < corrdim; i++) {
+      corrVec[i] = (int16_t)(corrVecTemp[i] >> scalefact);
+    }
+  } else {
+    for (i = 0; i < corrdim; i++) {
+      corrVec[i] = (int16_t)corrVecTemp[i];
+    }
+  }
+  /* In order to guarantee that all values are initialized */
+  for (i = corrdim; i < ENH_CORRDIM; i++) {
+    corrVec[i] = 0;
+  }
+
+  /* Upsample the correlation */
+  WebRtcIlbcfix_EnhUpsample(corrVecUps, corrVec);
+
+  /* Find maximum */
+  tloc = WebRtcSpl_MaxIndexW32(corrVecUps, ENH_UPS0 * corrdim);
+
+  /* make vector can be upsampled without ever running outside
+     bounds */
+  *updStartPos = searchSegStartPos * 4 + tloc + 4;
+
+  tloc2 = (tloc + 3) >> 2;
+
+  /* initialize the vector to be filtered, stuff with zeros
+     when data is outside idata buffer */
+  if (ENH_FL0 > (searchSegStartPos + tloc2)) {
+    const size_t st = ENH_FL0 - searchSegStartPos - tloc2;
+    WebRtcSpl_MemSetW16(vect, 0, st);
+    WEBRTC_SPL_MEMCPY_W16(&vect[st], idata, ENH_VECTL - st);
+  } else {
+    const size_t st = searchSegStartPos + tloc2 - ENH_FL0;
+    if ((st + ENH_VECTL) > idatal) {
+      const size_t en = st + ENH_VECTL - idatal;
+      WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL - en);
+      WebRtcSpl_MemSetW16(&vect[ENH_VECTL - en], 0, en);
+    } else {
+      WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL);
+    }
+  }
+
+  /* compute the segment (this is actually a convolution) */
+  filtStatePtr = filt + 6;
+  polyPtr = (int16_t*)WebRtcIlbcfix_kEnhPolyPhaser[tloc2 * ENH_UPS0 - tloc];
+  for (i = 0; i < 7; i++) {
+    *filtStatePtr-- = *polyPtr++;
+  }
+
+  WebRtcSpl_FilterMAFastQ12(&vect[6], vect, filt, ENH_FLO_MULT2_PLUS1,
+                            ENH_BLOCKL);
+
+  /* Add the contribution from this vector (scaled with gain) to the total
+     surround vector */
+  WebRtcSpl_AddAffineVectorToVector(surround, vect, gain, 32768, 16,
+                                    ENH_BLOCKL);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.h
new file mode 100644
index 0000000000..564c9d96e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Refiner.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * find segment starting near idata+estSegPos that has highest
+ * correlation with idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
+ * resolution of ENH_UPSO times the original of the original
+ * sampling rate
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Refiner(
+    size_t* updStartPos,   /* (o) updated start point (Q-2) */
+    int16_t* idata,        /* (i) original data buffer */
+    size_t idatal,         /* (i) dimension of idata */
+    size_t centerStartPos, /* (i) beginning center segment */
+    size_t estSegPos,      /* (i) estimated beginning other segment (Q-2) */
+    int16_t* surround,     /* (i/o) The contribution from this sequence
+                                    summed with earlier contributions */
+    int16_t gain           /* (i) Gain to use for this sequence */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
new file mode 100644
index 0000000000..7343530a5e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleInterpolateLsf.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h"
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleInterpolateLsf(
+    int16_t *syntdenum, /* (o) the synthesis filter denominator
+                                   resulting from the quantized
+                                   interpolated lsf Q12 */
+    int16_t *weightdenum, /* (o) the weighting filter denominator
+                                   resulting from the unquantized
+                                   interpolated lsf Q12 */
+    int16_t *lsf,  /* (i) the unquantized lsf coefficients Q13 */
+    int16_t *lsfdeq,  /* (i) the dequantized lsf coefficients Q13 */
+    int16_t *lsfold,  /* (i) the unquantized lsf coefficients of
+                                           the previous signal frame Q13 */
+    int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the
+                                   previous signal frame Q13 */
+    int16_t length,  /* (i) should equate FILTERORDER */
+    IlbcEncoder *iLBCenc_inst
+    /* (i/o) the encoder state structure */
+                                        ) {
+  size_t i;
+  int pos, lp_length;
+
+  int16_t *lsf2, *lsfdeq2;
+  /* Stack based */
+  int16_t lp[LPC_FILTERORDER + 1];
+
+  lsf2 = lsf + length;
+  lsfdeq2 = lsfdeq + length;
+  lp_length = length + 1;
+
+  if (iLBCenc_inst->mode==30) {
+    /* subframe 1: Interpolation between old and first set of
+       lsf coefficients */
+
+    /* Calculate Analysis/Syntehsis filter from quantized LSF */
+    WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
+                                         WebRtcIlbcfix_kLsfWeight30ms[0],
+                                         length);
+    WEBRTC_SPL_MEMCPY_W16(syntdenum, lp, lp_length);
+
+    /* Calculate Weighting filter from quantized LSF */
+    WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
+                                         WebRtcIlbcfix_kLsfWeight30ms[0],
+                                         length);
+    WebRtcIlbcfix_BwExpand(weightdenum, lp,
+                           (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                           (int16_t)lp_length);
+
+    /* subframe 2 to 6: Interpolation between first and second
+       set of lsf coefficients */
+
+    pos = lp_length;
+    for (i = 1; i < iLBCenc_inst->nsub; i++) {
+
+      /* Calculate Analysis/Syntehsis filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeq, lsfdeq2,
+                                           WebRtcIlbcfix_kLsfWeight30ms[i],
+                                           length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
+
+      /* Calculate Weighting filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsf, lsf2,
+                                           WebRtcIlbcfix_kLsfWeight30ms[i],
+                                           length);
+      WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                             (int16_t)lp_length);
+
+      pos += lp_length;
+    }
+
+    /* update memory */
+
+    WEBRTC_SPL_MEMCPY_W16(lsfold, lsf2, length);
+    WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq2, length);
+
+  } else { /* iLBCenc_inst->mode==20 */
+    pos = 0;
+    for (i = 0; i < iLBCenc_inst->nsub; i++) {
+
+      /* Calculate Analysis/Syntehsis filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
+                                           WebRtcIlbcfix_kLsfWeight20ms[i],
+                                           length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
+
+      /* Calculate Weighting filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
+                                           WebRtcIlbcfix_kLsfWeight20ms[i],
+                                           length);
+      WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                             (int16_t)lp_length);
+
+      pos += lp_length;
+    }
+
+    /* update memory */
+
+    WEBRTC_SPL_MEMCPY_W16(lsfold, lsf, length);
+    WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq, length);
+
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
new file mode 100644
index 0000000000..ee53e4bd08
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleInterpolateLsf.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleInterpolateLsf(
+    int16_t* syntdenum,   /* (o) the synthesis filter denominator
+                                     resulting from the quantized
+                                     interpolated lsf Q12 */
+    int16_t* weightdenum, /* (o) the weighting filter denominator
+                                   resulting from the unquantized
+                                   interpolated lsf Q12 */
+    int16_t* lsf,         /* (i) the unquantized lsf coefficients Q13 */
+    int16_t* lsfdeq,      /* (i) the dequantized lsf coefficients Q13 */
+    int16_t* lsfold,      /* (i) the unquantized lsf coefficients of
+                                               the previous signal frame Q13 */
+    int16_t* lsfdeqold,   /* (i) the dequantized lsf coefficients of the
+                                     previous signal frame Q13 */
+    int16_t length,       /* (i) should equate FILTERORDER */
+    IlbcEncoder* iLBCenc_inst
+    /* (i/o) the encoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
new file mode 100644
index 0000000000..fdc4553d95
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLpcAnalysis.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/window32_w32.h"
+
+/*----------------------------------------------------------------*
+ *  lpc analysis (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLpcAnalysis(
+    int16_t *lsf,   /* (o) lsf coefficients */
+    int16_t *data,   /* (i) new block of speech */
+    IlbcEncoder *iLBCenc_inst
+    /* (i/o) the encoder state structure */
+                                     ) {
+  int k;
+  int scale;
+  size_t is;
+  int16_t stability;
+  /* Stack based */
+  int16_t A[LPC_FILTERORDER + 1];
+  int32_t R[LPC_FILTERORDER + 1];
+  int16_t windowedData[BLOCKL_MAX];
+  int16_t rc[LPC_FILTERORDER];
+
+  is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer+is,data,iLBCenc_inst->blockl);
+
+  /* No lookahead, last window is asymmetric */
+
+  for (k = 0; k < iLBCenc_inst->lpc_n; k++) {
+
+    is = LPC_LOOKBACK;
+
+    if (k < (iLBCenc_inst->lpc_n - 1)) {
+
+      /* Hanning table WebRtcIlbcfix_kLpcWin[] is in Q15-domain so the output is right-shifted 15 */
+      WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer, WebRtcIlbcfix_kLpcWin, BLOCKL_MAX, 15);
+    } else {
+
+      /* Hanning table WebRtcIlbcfix_kLpcAsymWin[] is in Q15-domain so the output is right-shifted 15 */
+      WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer+is, WebRtcIlbcfix_kLpcAsymWin, BLOCKL_MAX, 15);
+    }
+
+    /* Compute autocorrelation */
+    WebRtcSpl_AutoCorrelation(windowedData, BLOCKL_MAX, LPC_FILTERORDER, R, &scale);
+
+    /* Window autocorrelation vector */
+    WebRtcIlbcfix_Window32W32(R, R, WebRtcIlbcfix_kLpcLagWin, LPC_FILTERORDER + 1 );
+
+    /* Calculate the A coefficients from the Autocorrelation using Levinson Durbin algorithm */
+    stability=WebRtcSpl_LevinsonDurbin(R, A, rc, LPC_FILTERORDER);
+
+    /*
+       Set the filter to {1.0, 0.0, 0.0,...} if filter from Levinson Durbin algorithm is unstable
+       This should basically never happen...
+    */
+    if (stability!=1) {
+      A[0]=4096;
+      WebRtcSpl_MemSetW16(&A[1], 0, LPC_FILTERORDER);
+    }
+
+    /* Bandwidth expand the filter coefficients */
+    WebRtcIlbcfix_BwExpand(A, A, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, LPC_FILTERORDER+1);
+
+    /* Convert from A to LSF representation */
+    WebRtcIlbcfix_Poly2Lsf(lsf + k*LPC_FILTERORDER, A);
+  }
+
+  is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer,
+                        iLBCenc_inst->lpc_buffer+LPC_LOOKBACK+BLOCKL_MAX-is, is);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
new file mode 100644
index 0000000000..b5c839ba2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLpcAnalysis.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc analysis (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLpcAnalysis(
+    int16_t* lsf,  /* (o) lsf coefficients */
+    int16_t* data, /* (i) new block of speech */
+    IlbcEncoder* iLBCenc_inst
+    /* (i/o) the encoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
new file mode 100644
index 0000000000..e7494ceb59
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfDeQ.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  obtain dequantized lsf coefficients from quantization index
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfDeQ(
+    int16_t *lsfdeq,  /* (o) dequantized lsf coefficients */
+    int16_t *index,  /* (i) quantization index */
+    int16_t lpc_n  /* (i) number of LPCs */
+                                ){
+  int i, j, pos, cb_pos;
+
+  /* decode first LSF */
+
+  pos = 0;
+  cb_pos = 0;
+  for (i = 0; i < LSF_NSPLIT; i++) {
+    for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
+      lsfdeq[pos + j] = WebRtcIlbcfix_kLsfCb[cb_pos + j + index[i] *
+                                             WebRtcIlbcfix_kLsfDimCb[i]];
+    }
+    pos += WebRtcIlbcfix_kLsfDimCb[i];
+    cb_pos += WebRtcIlbcfix_kLsfSizeCb[i] * WebRtcIlbcfix_kLsfDimCb[i];
+  }
+
+  if (lpc_n>1) {
+    /* decode last LSF */
+    pos = 0;
+    cb_pos = 0;
+    for (i = 0; i < LSF_NSPLIT; i++) {
+      for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
+        lsfdeq[LPC_FILTERORDER + pos + j] = WebRtcIlbcfix_kLsfCb[
+            cb_pos + index[LSF_NSPLIT + i] * WebRtcIlbcfix_kLsfDimCb[i] + j];
+      }
+      pos += WebRtcIlbcfix_kLsfDimCb[i];
+      cb_pos += WebRtcIlbcfix_kLsfSizeCb[i] * WebRtcIlbcfix_kLsfDimCb[i];
+    }
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
new file mode 100644
index 0000000000..6d97d3df33
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfDeQ.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  obtain dequantized lsf coefficients from quantization index
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfDeQ(
+    int16_t* lsfdeq, /* (o) dequantized lsf coefficients */
+    int16_t* index,  /* (i) quantization index */
+    int16_t lpc_n    /* (i) number of LPCs */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
new file mode 100644
index 0000000000..1291d1442e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfQ.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_quant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/split_vq.h"
+
+/*----------------------------------------------------------------*
+ *  lsf quantizer (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfQ(
+    int16_t *lsfdeq, /* (o) dequantized lsf coefficients
+                                   (dimension FILTERORDER) Q13 */
+    int16_t *index, /* (o) quantization index */
+    int16_t *lsf, /* (i) the lsf coefficient vector to be
+                           quantized (dimension FILTERORDER) Q13 */
+    int16_t lpc_n /* (i) number of lsf sets to quantize */
+                              ){
+
+  /* Quantize first LSF with memoryless split VQ */
+  WebRtcIlbcfix_SplitVq( lsfdeq, index, lsf,
+                         (int16_t*)WebRtcIlbcfix_kLsfCb, (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb);
+
+  if (lpc_n==2) {
+    /* Quantize second LSF with memoryless split VQ */
+    WebRtcIlbcfix_SplitVq( lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT,
+                           lsf + LPC_FILTERORDER, (int16_t*)WebRtcIlbcfix_kLsfCb,
+                           (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb);
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
new file mode 100644
index 0000000000..66b553213a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfQ.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  lsf quantizer (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfQ(
+    int16_t* lsfdeq, /* (o) dequantized lsf coefficients
+                                   (dimension FILTERORDER) Q13 */
+    int16_t* index,  /* (o) quantization index */
+    int16_t* lsf,    /* (i) the lsf coefficient vector to be
+                              quantized (dimension FILTERORDER) Q13 */
+    int16_t lpc_n    /* (i) number of lsf sets to quantize */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c
new file mode 100644
index 0000000000..631b2f432a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c
@@ -0,0 +1,212 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/smooth.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/smooth_out_data.h"
+
+/*----------------------------------------------------------------*
+ * find the smoothed output data
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Smooth(
+    int16_t *odata,   /* (o) smoothed output */
+    int16_t *current,  /* (i) the un enhanced residual for
+                                this block */
+    int16_t *surround  /* (i) The approximation from the
+                                surrounding sequences */
+                          ) {
+  int16_t scale, scale1, scale2;
+  int16_t A, B, C, denomW16;
+  int32_t B_W32, denom, num;
+  int32_t errs;
+  int32_t w00,w10,w11, endiff, crit;
+  int32_t w00prim, w10prim, w11_div_w00;
+  int16_t w11prim;
+  int16_t bitsw00, bitsw10, bitsw11;
+  int32_t w11w00, w10w10, w00w00;
+  uint32_t max1, max2, max12;
+
+  /* compute some inner products (ensure no overflow by first calculating proper scale factor) */
+
+  w00 = w10 = w11 = 0;
+
+  // Calculate a right shift that will let us sum ENH_BLOCKL pairwise products
+  // of values from the two sequences without overflowing an int32_t. (The +1
+  // in max1 and max2 are because WebRtcSpl_MaxAbsValueW16 will return 2**15 -
+  // 1 if the input array contains -2**15.)
+  max1 = WebRtcSpl_MaxAbsValueW16(current, ENH_BLOCKL) + 1;
+  max2 = WebRtcSpl_MaxAbsValueW16(surround, ENH_BLOCKL) + 1;
+  max12 = WEBRTC_SPL_MAX(max1, max2);
+  scale = (64 - 31) -
+          WebRtcSpl_CountLeadingZeros64((max12 * max12) * (uint64_t)ENH_BLOCKL);
+  scale=WEBRTC_SPL_MAX(0, scale);
+
+  w00=WebRtcSpl_DotProductWithScale(current,current,ENH_BLOCKL,scale);
+  w11=WebRtcSpl_DotProductWithScale(surround,surround,ENH_BLOCKL,scale);
+  w10=WebRtcSpl_DotProductWithScale(surround,current,ENH_BLOCKL,scale);
+
+  if (w00<0) w00 = WEBRTC_SPL_WORD32_MAX;
+  if (w11<0) w11 = WEBRTC_SPL_WORD32_MAX;
+
+  /* Rescale w00 and w11 to w00prim and w11prim, so that w00prim/w11prim
+     is in Q16 */
+
+  bitsw00 = WebRtcSpl_GetSizeInBits(w00);
+  bitsw11 = WebRtcSpl_GetSizeInBits(w11);
+  bitsw10 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(w10));
+  scale1 = 31 - bitsw00;
+  scale2 = 15 - bitsw11;
+
+  if (scale2>(scale1-16)) {
+    scale2 = scale1 - 16;
+  } else {
+    scale1 = scale2 + 16;
+  }
+
+  w00prim = w00 << scale1;
+  w11prim = (int16_t) WEBRTC_SPL_SHIFT_W32(w11, scale2);
+
+  /* Perform C = sqrt(w11/w00) (C is in Q11 since (16+6)/2=11) */
+  if (w11prim>64) {
+    endiff = WebRtcSpl_DivW32W16(w00prim, w11prim) << 6;
+    C = (int16_t)WebRtcSpl_SqrtFloor(endiff); /* C is in Q11 */
+  } else {
+    C = 1;
+  }
+
+  /* first try enhancement without power-constraint */
+
+  errs = WebRtcIlbcfix_Smooth_odata(odata, current, surround, C);
+
+
+
+  /* if constraint violated by first try, add constraint */
+
+  if ( (6-scale+scale1) > 31) {
+    crit=0;
+  } else {
+    /* crit = 0.05 * w00 (Result in Q-6) */
+    crit = WEBRTC_SPL_SHIFT_W32(
+        WEBRTC_SPL_MUL(ENH_A0, w00prim >> 14),
+        -(6-scale+scale1));
+  }
+
+  if (errs > crit) {
+
+    if( w00 < 1) {
+      w00=1;
+    }
+
+    /* Calculate w11*w00, w10*w10 and w00*w00 in the same Q domain */
+
+    scale1 = bitsw00-15;
+    scale2 = bitsw11-15;
+
+    if (scale2>scale1) {
+      scale = scale2;
+    } else {
+      scale = scale1;
+    }
+
+    w11w00 = (int16_t)WEBRTC_SPL_SHIFT_W32(w11, -scale) *
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale);
+
+    w10w10 = (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale) *
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale);
+
+    w00w00 = (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale) *
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale);
+
+    /* Calculate (w11*w00-w10*w10)/(w00*w00) in Q16 */
+    if (w00w00>65536) {
+      endiff = (w11w00-w10w10);
+      endiff = WEBRTC_SPL_MAX(0, endiff);
+      /* denom is in Q16 */
+      denom = WebRtcSpl_DivW32W16(endiff, (int16_t)(w00w00 >> 16));
+    } else {
+      denom = 65536;
+    }
+
+    if( denom > 7){ /* eliminates numerical problems
+                       for if smooth */
+
+      scale=WebRtcSpl_GetSizeInBits(denom)-15;
+
+      if (scale>0) {
+        /* denomW16 is in Q(16+scale) */
+        denomW16 = (int16_t)(denom >> scale);
+
+        /* num in Q(34-scale) */
+        num = ENH_A0_MINUS_A0A0DIV4 >> scale;
+      } else {
+        /* denomW16 is in Q16 */
+        denomW16=(int16_t)denom;
+
+        /* num in Q34 */
+        num=ENH_A0_MINUS_A0A0DIV4;
+      }
+
+      /* A sqrt( (ENH_A0-(ENH_A0^2)/4)*(w00*w00)/(w11*w00 + w10*w10) ) in Q9 */
+      A = (int16_t)WebRtcSpl_SqrtFloor(WebRtcSpl_DivW32W16(num, denomW16));
+
+      /* B_W32 is in Q30 ( B = 1 - ENH_A0/2 - A * w10/w00 ) */
+      scale1 = 31-bitsw10;
+      scale2 = 21-scale1;
+      w10prim = w10 == 0 ? 0 : w10 * (1 << scale1);
+      w00prim = WEBRTC_SPL_SHIFT_W32(w00, -scale2);
+      scale = bitsw00-scale2-15;
+
+      if (scale>0) {
+        w10prim >>= scale;
+        w00prim >>= scale;
+      }
+
+      if ((w00prim>0)&&(w10prim>0)) {
+        w11_div_w00=WebRtcSpl_DivW32W16(w10prim, (int16_t)w00prim);
+
+        if (WebRtcSpl_GetSizeInBits(w11_div_w00)+WebRtcSpl_GetSizeInBits(A)>31) {
+          B_W32 = 0;
+        } else {
+          B_W32 = (int32_t)1073741824 - (int32_t)ENH_A0DIV2 -
+              WEBRTC_SPL_MUL(A, w11_div_w00);
+        }
+        B = (int16_t)(B_W32 >> 16);  /* B in Q14. */
+      } else {
+        /* No smoothing */
+        A = 0;
+        B = 16384; /* 1 in Q14 */
+      }
+    }
+    else{ /* essentially no difference between cycles;
+             smoothing not needed */
+
+      A = 0;
+      B = 16384; /* 1 in Q14 */
+    }
+
+    /* create smoothed sequence */
+
+    WebRtcSpl_ScaleAndAddVectors(surround, A, 9,
+                                current, B, 14,
+                                odata, ENH_BLOCKL);
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.h
new file mode 100644
index 0000000000..c8752be64f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * find the smoothed output data
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Smooth(int16_t* odata,   /* (o) smoothed output */
+                          int16_t* current, /* (i) the un enhanced residual for
+                                                     this block */
+                          int16_t* surround /* (i) The approximation from the
+                                                     surrounding sequences */
+                          );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
new file mode 100644
index 0000000000..9f952bfb93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth_odata.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/smooth_out_data.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "rtc_base/sanitizer.h"
+
+// An s32 + s32 -> s32 addition that's allowed to overflow. (It's still
+// undefined behavior, so not a good idea; this just makes UBSan ignore the
+// violation, so that our old code can continue to do what it's always been
+// doing.)
+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
+    OverflowingAdd_S32_S32_To_S32(int32_t a, int32_t b) {
+  return a + b;
+}
+
+int32_t WebRtcIlbcfix_Smooth_odata(
+    int16_t *odata,
+    int16_t *psseq,
+    int16_t *surround,
+    int16_t C)
+{
+  int i;
+
+  int16_t err;
+  int32_t errs;
+
+  for(i=0;i<80;i++) {
+    odata[i]= (int16_t)((C * surround[i] + 1024) >> 11);
+  }
+
+  errs=0;
+  for(i=0;i<80;i++) {
+    err = (psseq[i] - odata[i]) >> 3;
+    errs = OverflowingAdd_S32_S32_To_S32(errs, err * err);  // errs in Q-6
+  }
+
+  return errs;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h
new file mode 100644
index 0000000000..318e7b04a2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth_odata.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * help function to WebRtcIlbcfix_Smooth()
+ *---------------------------------------------------------------*/
+
+int32_t WebRtcIlbcfix_Smooth_odata(int16_t* odata,
+                                   int16_t* psseq,
+                                   int16_t* surround,
+                                   int16_t C);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c
new file mode 100644
index 0000000000..c3a24750f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SortSq.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/sort_sq.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  scalar quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SortSq(
+    int16_t *xq,   /* (o) the quantized value */
+    int16_t *index,  /* (o) the quantization index */
+    int16_t x,   /* (i) the value to quantize */
+    const int16_t *cb, /* (i) the quantization codebook */
+    int16_t cb_size  /* (i) the size of the quantization codebook */
+                          ){
+  int i;
+
+  if (x <= cb[0]) {
+    *index = 0;
+    *xq = cb[0];
+  } else {
+    i = 0;
+    while ((x > cb[i]) && (i < (cb_size-1))) {
+      i++;
+    }
+
+    if (x > (((int32_t)cb[i] + cb[i - 1] + 1) >> 1)) {
+      *index = i;
+      *xq = cb[i];
+    } else {
+      *index = i - 1;
+      *xq = cb[i - 1];
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.h
new file mode 100644
index 0000000000..02028dae93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SortSq.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  scalar quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SortSq(
+    int16_t* xq,       /* (o) the quantized value */
+    int16_t* index,    /* (o) the quantization index */
+    int16_t x,         /* (i) the value to quantize */
+    const int16_t* cb, /* (i) the quantization codebook */
+    int16_t cb_size    /* (i) the size of the quantization codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c
new file mode 100644
index 0000000000..c1f04d2287
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SplitVq.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/split_vq.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/vq3.h"
+#include "modules/audio_coding/codecs/ilbc/vq4.h"
+
+/*----------------------------------------------------------------*
+ *  split vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SplitVq(
+    int16_t *qX,  /* (o) the quantized vector in Q13 */
+    int16_t *index, /* (o) a vector of indexes for all vector
+                                   codebooks in the split */
+    int16_t *X,  /* (i) the vector to quantize */
+    int16_t *CB,  /* (i) the quantizer codebook in Q13 */
+    int16_t *dim, /* (i) the dimension of X and qX */
+    int16_t *cbsize /* (i) the number of vectors in the codebook */
+                           ) {
+
+  int16_t *qXPtr, *indexPtr, *CBPtr, *XPtr;
+
+  /* Quantize X with the 3 vectror quantization tables */
+
+  qXPtr=qX;
+  indexPtr=index;
+  CBPtr=CB;
+  XPtr=X;
+  WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[0]);
+
+  qXPtr+=3;
+  indexPtr+=1;
+  CBPtr+=(dim[0]*cbsize[0]);
+  XPtr+=3;
+  WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[1]);
+
+  qXPtr+=3;
+  indexPtr+=1;
+  CBPtr+=(dim[1]*cbsize[1]);
+  XPtr+=3;
+  WebRtcIlbcfix_Vq4(qXPtr, indexPtr, CBPtr, XPtr, cbsize[2]);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.h
new file mode 100644
index 0000000000..e4b02a2bc2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SplitVq.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  split vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SplitVq(
+    int16_t* qX,    /* (o) the quantized vector in Q13 */
+    int16_t* index, /* (o) a vector of indexes for all vector
+                                   codebooks in the split */
+    int16_t* X,     /* (i) the vector to quantize */
+    int16_t* CB,    /* (i) the quantizer codebook in Q13 */
+    int16_t* dim,   /* (i) the dimension of X and qX */
+    int16_t* cbsize /* (i) the number of vectors in the codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c
new file mode 100644
index 0000000000..c58086c03b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateConstruct.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  decoding of the start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateConstruct(
+    size_t idxForMax,   /* (i) 6-bit index for the quantization of
+                                           max amplitude */
+    int16_t *idxVec,   /* (i) vector of quantization indexes */
+    int16_t *syntDenum,  /* (i) synthesis filter denumerator */
+    int16_t *Out_fix,  /* (o) the decoded state vector */
+    size_t len    /* (i) length of a state vector */
+                                  ) {
+  size_t k;
+  int16_t maxVal;
+  int16_t *tmp1, *tmp2, *tmp3;
+  /* Stack based */
+  int16_t numerator[1+LPC_FILTERORDER];
+  int16_t sampleValVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t sampleMaVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t *sampleVal = &sampleValVec[LPC_FILTERORDER];
+  int16_t *sampleMa = &sampleMaVec[LPC_FILTERORDER];
+  int16_t *sampleAr = &sampleValVec[LPC_FILTERORDER];
+
+  /* initialization of coefficients */
+
+  for (k=0; k<LPC_FILTERORDER+1; k++){
+    numerator[k] = syntDenum[LPC_FILTERORDER-k];
+  }
+
+  /* decoding of the maximum value */
+
+  maxVal = WebRtcIlbcfix_kFrgQuantMod[idxForMax];
+
+  /* decoding of the sample values */
+  tmp1 = sampleVal;
+  tmp2 = &idxVec[len-1];
+
+  if (idxForMax<37) {
+    for(k=0; k<len; k++){
+      /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 2097152 (= 0.5 << 22)
+        maxVal is in Q8 and result is in Q(-1) */
+      *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 2097152) >>
+          22);
+      tmp1++;
+      tmp2--;
+    }
+  } else if (idxForMax<59) {
+    for(k=0; k<len; k++){
+      /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 262144 (= 0.5 << 19)
+        maxVal is in Q5 and result is in Q(-1) */
+      *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 262144) >>
+          19);
+      tmp1++;
+      tmp2--;
+    }
+  } else {
+    for(k=0; k<len; k++){
+      /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 65536 (= 0.5 << 17)
+        maxVal is in Q3 and result is in Q(-1) */
+      *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 65536) >>
+          17);
+      tmp1++;
+      tmp2--;
+    }
+  }
+
+  /* Set the rest of the data to zero */
+  WebRtcSpl_MemSetW16(&sampleVal[len], 0, len);
+
+  /* circular convolution with all-pass filter */
+
+  /* Set the state to zero */
+  WebRtcSpl_MemSetW16(sampleValVec, 0, (LPC_FILTERORDER));
+
+  /* Run MA filter + AR filter */
+  WebRtcSpl_FilterMAFastQ12(
+      sampleVal, sampleMa,
+      numerator, LPC_FILTERORDER+1, len + LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER));
+  WebRtcSpl_FilterARFastQ12(
+      sampleMa, sampleAr,
+      syntDenum, LPC_FILTERORDER+1, 2 * len);
+
+  tmp1 = &sampleAr[len-1];
+  tmp2 = &sampleAr[2*len-1];
+  tmp3 = Out_fix;
+  for(k=0;k<len;k++){
+    (*tmp3) = (*tmp1) + (*tmp2);
+    tmp1--;
+    tmp2--;
+    tmp3++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.h
new file mode 100644
index 0000000000..4c3011937d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateConstruct.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Generate the start state from the quantized indexes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateConstruct(
+    size_t idxForMax,   /* (i) 6-bit index for the quantization of
+                                           max amplitude */
+    int16_t* idxVec,    /* (i) vector of quantization indexes */
+    int16_t* syntDenum, /* (i) synthesis filter denumerator */
+    int16_t* Out_fix,   /* (o) the decoded state vector */
+    size_t len          /* (i) length of a state vector */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c
new file mode 100644
index 0000000000..7227ac9d45
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateSearch.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/state_search.h"
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  encoding of start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateSearch(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
+                               and idxVec, input state_first) */
+    int16_t *residual,   /* (i) target residual vector */
+    int16_t *syntDenum,  /* (i) lpc synthesis filter */
+    int16_t *weightDenum  /* (i) weighting filter denuminator */
+                               ) {
+  size_t k, index;
+  int16_t maxVal;
+  int16_t scale, shift;
+  int32_t maxValsq;
+  int16_t scaleRes;
+  int16_t max;
+  int i;
+  /* Stack based */
+  int16_t numerator[1+LPC_FILTERORDER];
+  int16_t residualLongVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t sampleMa[2*STATE_SHORT_LEN_30MS];
+  int16_t *residualLong = &residualLongVec[LPC_FILTERORDER];
+  int16_t *sampleAr = residualLong;
+
+  /* Scale to maximum 12 bits to avoid saturation in circular convolution filter */
+  max = WebRtcSpl_MaxAbsValueW16(residual, iLBCenc_inst->state_short_len);
+  scaleRes = WebRtcSpl_GetSizeInBits(max)-12;
+  scaleRes = WEBRTC_SPL_MAX(0, scaleRes);
+  /* Set up the filter coefficients for the circular convolution */
+  for (i=0; i<LPC_FILTERORDER+1; i++) {
+    numerator[i] = (syntDenum[LPC_FILTERORDER-i]>>scaleRes);
+  }
+
+  /* Copy the residual to a temporary buffer that we can filter
+   * and set the remaining samples to zero.
+   */
+  WEBRTC_SPL_MEMCPY_W16(residualLong, residual, iLBCenc_inst->state_short_len);
+  WebRtcSpl_MemSetW16(residualLong + iLBCenc_inst->state_short_len, 0, iLBCenc_inst->state_short_len);
+
+  /* Run the Zero-Pole filter (Ciurcular convolution) */
+  WebRtcSpl_MemSetW16(residualLongVec, 0, LPC_FILTERORDER);
+  WebRtcSpl_FilterMAFastQ12(residualLong, sampleMa, numerator,
+                            LPC_FILTERORDER + 1,
+                            iLBCenc_inst->state_short_len + LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(&sampleMa[iLBCenc_inst->state_short_len + LPC_FILTERORDER], 0, iLBCenc_inst->state_short_len - LPC_FILTERORDER);
+
+  WebRtcSpl_FilterARFastQ12(
+      sampleMa, sampleAr,
+      syntDenum, LPC_FILTERORDER+1, 2 * iLBCenc_inst->state_short_len);
+
+  for(k=0;k<iLBCenc_inst->state_short_len;k++){
+    sampleAr[k] += sampleAr[k+iLBCenc_inst->state_short_len];
+  }
+
+  /* Find maximum absolute value in the vector */
+  maxVal=WebRtcSpl_MaxAbsValueW16(sampleAr, iLBCenc_inst->state_short_len);
+
+  /* Find the best index */
+
+  if ((((int32_t)maxVal)<<scaleRes)<23170) {
+    maxValsq=((int32_t)maxVal*maxVal)<<(2+2*scaleRes);
+  } else {
+    maxValsq=(int32_t)WEBRTC_SPL_WORD32_MAX;
+  }
+
+  index=0;
+  for (i=0;i<63;i++) {
+
+    if (maxValsq>=WebRtcIlbcfix_kChooseFrgQuant[i]) {
+      index=i+1;
+    } else {
+      i=63;
+    }
+  }
+  iLBC_encbits->idxForMax=index;
+
+  /* Rescale the vector before quantization */
+  scale=WebRtcIlbcfix_kScale[index];
+
+  if (index<27) { /* scale table is in Q16, fout[] is in Q(-1) and we want the result to be in Q11 */
+    shift=4;
+  } else { /* scale table is in Q21, fout[] is in Q(-1) and we want the result to be in Q11 */
+    shift=9;
+  }
+
+  /* Set up vectors for AbsQuant and rescale it with the scale factor */
+  WebRtcSpl_ScaleVectorWithSat(sampleAr, sampleAr, scale,
+                              iLBCenc_inst->state_short_len, (int16_t)(shift-scaleRes));
+
+  /* Quantize the values in fout[] */
+  WebRtcIlbcfix_AbsQuant(iLBCenc_inst, iLBC_encbits, sampleAr, weightDenum);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.h
new file mode 100644
index 0000000000..6469138a0e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateSearch.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  encoding of start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateSearch(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits* iLBC_encbits, /* (i/o) Encoded bits (output idxForMax
+                                and idxVec, input state_first) */
+    int16_t* residual,       /* (i) target residual vector */
+    int16_t* syntDenum,      /* (i) lpc synthesis filter */
+    int16_t* weightDenum     /* (i) weighting filter denuminator */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
new file mode 100644
index 0000000000..bbafc1a2ed
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SwapBytes.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
+
+/*----------------------------------------------------------------*
+ * Swap bytes (to simplify operations on Little Endian machines)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SwapBytes(
+    const uint16_t* input,   /* (i) the sequence to swap */
+    size_t wordLength,      /* (i) number or uint16_t to swap */
+    uint16_t* output         /* (o) the swapped sequence */
+                              ) {
+  size_t k;
+  for (k = wordLength; k > 0; k--) {
+    *output++ = (*input >> 8)|(*input << 8);
+    input++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
new file mode 100644
index 0000000000..c59bf3068a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SwapBytes.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * Swap bytes (to simplify operations on Little Endian machines)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SwapBytes(
+    const uint16_t* input, /* (i) the sequence to swap */
+    size_t wordLength,     /* (i) number or uint16_t to swap */
+    uint16_t* output       /* (o) the swapped sequence */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/empty.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/empty.cc
new file mode 100644
index 0000000000..e69de29bb2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/empty.cc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
new file mode 100644
index 0000000000..e0ca075eda
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
@@ -0,0 +1,238 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+	iLBC Speech Coder ANSI-C Source Code
+
+        iLBC_test.c
+
+******************************************************************/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+/*---------------------------------------------------------------*
+ *  Main program to test iLBC encoding and decoding
+ *
+ *  Usage:
+ *	  exefile_name.exe <infile> <bytefile> <outfile> <channel>
+ *
+ *    <infile>   : Input file, speech for encoder (16-bit pcm file)
+ *    <bytefile> : Bit stream output from the encoder
+ *    <outfile>  : Output file, decoded speech (16-bit pcm file)
+ *    <channel>  : Bit error file, optional (16-bit)
+ *                     1 - Packet received correctly
+ *                     0 - Packet Lost
+ *
+ *--------------------------------------------------------------*/
+
+#define BLOCKL_MAX			240
+#define ILBCNOOFWORDS_MAX	25
+
+
+int main(int argc, char* argv[])
+{
+
+  FILE *ifileid,*efileid,*ofileid, *cfileid;
+  int16_t data[BLOCKL_MAX];
+  uint8_t encoded_data[2 * ILBCNOOFWORDS_MAX];
+  int16_t decoded_data[BLOCKL_MAX];
+  int len_int, mode;
+  short pli;
+  int blockcount = 0;
+  size_t frameLen, len, len_i16s;
+  int16_t speechType;
+  IlbcEncoderInstance *Enc_Inst;
+  IlbcDecoderInstance *Dec_Inst;
+
+#ifdef __ILBC_WITH_40BITACC
+  /* Doublecheck that long long exists */
+  if (sizeof(long)>=sizeof(long long)) {
+    fprintf(stderr, "40-bit simulation is not be supported on this platform\n");
+    exit(0);
+  }
+#endif
+
+  /* get arguments and open files */
+
+  if ((argc!=5) && (argc!=6)) {
+    fprintf(stderr,
+            "\n*-----------------------------------------------*\n");
+    fprintf(stderr,
+            "   %s <20,30> input encoded decoded (channel)\n\n",
+            argv[0]);
+    fprintf(stderr,
+            "   mode    : Frame size for the encoding/decoding\n");
+    fprintf(stderr,
+            "                 20 - 20 ms\n");
+    fprintf(stderr,
+            "                 30 - 30 ms\n");
+    fprintf(stderr,
+            "   input   : Speech for encoder (16-bit pcm file)\n");
+    fprintf(stderr,
+            "   encoded : Encoded bit stream\n");
+    fprintf(stderr,
+            "   decoded : Decoded speech (16-bit pcm file)\n");
+    fprintf(stderr,
+            "   channel : Packet loss pattern, optional (16-bit)\n");
+    fprintf(stderr,
+            "                  1 - Packet received correctly\n");
+    fprintf(stderr,
+            "                  0 - Packet Lost\n");
+    fprintf(stderr,
+            "*-----------------------------------------------*\n\n");
+    exit(1);
+  }
+  mode=atoi(argv[1]);
+  if (mode != 20 && mode != 30) {
+    fprintf(stderr,"Wrong mode %s, must be 20, or 30\n",
+            argv[1]);
+    exit(2);
+  }
+  if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+    exit(2);}
+  if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open encoded file file %s\n",
+            argv[3]); exit(1);}
+  if ( (ofileid=fopen(argv[4],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open decoded file %s\n",
+            argv[4]); exit(1);}
+  if (argc==6) {
+    if( (cfileid=fopen(argv[5],"rb")) == NULL) {
+      fprintf(stderr, "Cannot open channel file %s\n",
+              argv[5]);
+      exit(1);
+    }
+  } else {
+    cfileid=NULL;
+  }
+
+  /* print info */
+
+  fprintf(stderr, "\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*      iLBC test program                            *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,"\nMode           : %2d ms\n", mode);
+  fprintf(stderr,"Input file     : %s\n", argv[2]);
+  fprintf(stderr,"Encoded file   : %s\n", argv[3]);
+  fprintf(stderr,"Output file    : %s\n", argv[4]);
+  if (argc==6) {
+    fprintf(stderr,"Channel file   : %s\n", argv[5]);
+  }
+  fprintf(stderr,"\n");
+
+  /* Create structs */
+  WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
+  WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
+
+
+  /* Initialization */
+
+  WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
+  WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
+  frameLen = (size_t)(mode*8);
+
+  /* loop over input blocks */
+
+  while (fread(data,sizeof(int16_t),frameLen,ifileid) == frameLen) {
+
+    blockcount++;
+
+    /* encoding */
+
+    fprintf(stderr, "--- Encoding block %i --- ",blockcount);
+    len_int = WebRtcIlbcfix_Encode(Enc_Inst, data, frameLen, encoded_data);
+    if (len_int < 0) {
+      fprintf(stderr, "Error encoding\n");
+      exit(0);
+    }
+    len = (size_t)len_int;
+    fprintf(stderr, "\r");
+
+    /* write byte file */
+
+    len_i16s = (len + 1) / sizeof(int16_t);
+    if (fwrite(encoded_data, sizeof(int16_t), len_i16s, efileid) != len_i16s) {
+      return -1;
+    }
+
+    /* get channel data if provided */
+    if (argc==6) {
+      if (fread(&pli, sizeof(int16_t), 1, cfileid)) {
+        if ((pli!=0)&&(pli!=1)) {
+          fprintf(stderr, "Error in channel file\n");
+          exit(0);
+        }
+        if (pli==0) {
+          /* Packet loss -> remove info from frame */
+          memset(encoded_data, 0,
+                 sizeof(int16_t)*ILBCNOOFWORDS_MAX);
+        }
+      } else {
+        fprintf(stderr, "Error. Channel file too short\n");
+        exit(0);
+      }
+    } else {
+      pli=1;
+    }
+
+    /* decoding */
+
+    fprintf(stderr, "--- Decoding block %i --- ",blockcount);
+    if (pli==1) {
+      len_int=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data,
+                                   len, decoded_data,&speechType);
+      if (len_int < 0) {
+        fprintf(stderr, "Error decoding\n");
+        exit(0);
+      }
+      len = (size_t)len_int;
+    } else {
+      len=WebRtcIlbcfix_DecodePlc(Dec_Inst, decoded_data, 1);
+    }
+    fprintf(stderr, "\r");
+
+    /* write output file */
+
+    if (fwrite(decoded_data, sizeof(int16_t), len, ofileid) != len) {
+      return -1;
+    }
+  }
+
+  /* close files */
+
+  fclose(ifileid);  fclose(efileid); fclose(ofileid);
+  if (argc==6) {
+    fclose(cfileid);
+  }
+
+  /* Free structs */
+  WebRtcIlbcfix_EncoderFree(Enc_Inst);
+  WebRtcIlbcfix_DecoderFree(Dec_Inst);
+
+
+  printf("\nDone with simulation\n\n");
+
+  return(0);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
new file mode 100644
index 0000000000..132f3bdb37
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -0,0 +1,215 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+iLBC Speech Coder ANSI-C Source Code
+
+iLBC_test.c
+
+******************************************************************/
+
+#include <math.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <time.h>
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+//#define JUNK_DATA
+#ifdef JUNK_DATA
+#define SEED_FILE "randseed.txt"
+#endif
+
+
+/*----------------------------------------------------------------*
+*  Main program to test iLBC encoding and decoding
+*
+*  Usage:
+*		exefile_name.exe <infile> <bytefile> <outfile>
+*
+*---------------------------------------------------------------*/
+
+int main(int argc, char* argv[])
+{
+  FILE *ifileid,*efileid,*ofileid, *chfileid;
+  short encoded_data[55], data[240], speechType;
+  int len_int, mode;
+  short pli;
+  size_t len, readlen;
+  int blockcount = 0;
+
+  IlbcEncoderInstance *Enc_Inst;
+  IlbcDecoderInstance *Dec_Inst;
+#ifdef JUNK_DATA
+  size_t i;
+  FILE *seedfile;
+  unsigned int random_seed = (unsigned int) time(NULL);//1196764538
+#endif
+
+  /* Create structs */
+  WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
+  WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
+
+  /* get arguments and open files */
+
+  if (argc != 6 ) {
+    fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
+            argv[0]);
+    fprintf(stderr, "Example:\n");
+    fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
+    exit(1);
+  }
+  mode=atoi(argv[1]);
+  if (mode != 20 && mode != 30) {
+    fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
+    exit(2);
+  }
+  if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+    exit(2);}
+  if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open channelfile file %s\n",
+            argv[3]); exit(3);}
+  if( (ofileid=fopen(argv[4],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open output file %s\n",
+            argv[4]); exit(3);}
+  if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
+    exit(2);
+  }
+  /* print info */
+  fprintf(stderr, "\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*      iLBCtest                                     *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+		"*---------------------------------------------------*\n");
+#ifdef SPLIT_10MS
+  fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
+#else
+  fprintf(stderr,"\nMode          : %2d ms\n", mode);
+#endif
+  fprintf(stderr,"\nInput file    : %s\n", argv[2]);
+  fprintf(stderr,"Coded file    : %s\n", argv[3]);
+  fprintf(stderr,"Output file   : %s\n\n", argv[4]);
+  fprintf(stderr,"Channel file  : %s\n\n", argv[5]);
+
+#ifdef JUNK_DATA
+  srand(random_seed);
+
+  if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+    fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+  }
+  else {
+    fprintf(seedfile, "%u\n", random_seed);
+    fclose(seedfile);
+  }
+#endif
+
+  /* Initialization */
+  WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
+  WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
+
+  /* loop over input blocks */
+#ifdef SPLIT_10MS
+  readlen = 80;
+#else
+  readlen = (size_t)(mode << 3);
+#endif
+  while(fread(data, sizeof(short), readlen, ifileid) == readlen) {
+    blockcount++;
+
+    /* encoding */
+    fprintf(stderr, "--- Encoding block %i --- ",blockcount);
+    len_int=WebRtcIlbcfix_Encode(Enc_Inst, data, readlen, encoded_data);
+    if (len_int < 0) {
+      fprintf(stderr, "Error encoding\n");
+      exit(0);
+    }
+    len = (size_t)len_int;
+    fprintf(stderr, "\r");
+
+#ifdef JUNK_DATA
+    for ( i = 0; i < len; i++) {
+      encoded_data[i] = (short) (encoded_data[i] + (short) rand());
+    }
+#endif
+    /* write byte file */
+    if(len != 0){ //len may be 0 in 10ms split case
+      fwrite(encoded_data,1,len,efileid);
+
+      /* get channel data if provided */
+      if (argc==6) {
+        if (fread(&pli, sizeof(int16_t), 1, chfileid)) {
+          if ((pli!=0)&&(pli!=1)) {
+            fprintf(stderr, "Error in channel file\n");
+            exit(0);
+          }
+          if (pli==0) {
+            /* Packet loss -> remove info from frame */
+            memset(encoded_data, 0, sizeof(int16_t)*25);
+          }
+        } else {
+          fprintf(stderr, "Error. Channel file too short\n");
+          exit(0);
+        }
+      } else {
+        pli=1;
+      }
+
+      /* decoding */
+      fprintf(stderr, "--- Decoding block %i --- ",blockcount);
+      if (pli==1) {
+        len_int = WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data,
+                                       &speechType);
+        if (len_int < 0) {
+          fprintf(stderr, "Error decoding\n");
+          exit(0);
+        }
+        len = (size_t)len_int;
+      } else {
+        len=WebRtcIlbcfix_DecodePlc(Dec_Inst, data, 1);
+      }
+      fprintf(stderr, "\r");
+
+      /* write output file */
+      fwrite(data,sizeof(short),len,ofileid);
+    }
+  }
+
+#ifdef JUNK_DATA
+  if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+    fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+  }
+  else {
+    fprintf(seedfile, "ok\n\n");
+    fclose(seedfile);
+  }
+#endif
+
+  /* free structs */
+  WebRtcIlbcfix_EncoderFree(Enc_Inst);
+  WebRtcIlbcfix_DecoderFree(Dec_Inst);
+
+  /* close files */
+  fclose(ifileid);
+  fclose(efileid);
+  fclose(ofileid);
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
new file mode 100644
index 0000000000..a62a42edf6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
@@ -0,0 +1,343 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+	iLBC Speech Coder ANSI-C Source Code
+
+        iLBC_test.c
+
+******************************************************************/
+
+#include <math.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/nit_encode.h"
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+#define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS)/2
+
+/* Runtime statistics */
+#include <time.h>
+/* #define CLOCKS_PER_SEC  1000 */
+
+/*----------------------------------------------------------------*
+ *  Encoder interface function
+ *---------------------------------------------------------------*/
+
+short encode(                         /* (o) Number of bytes encoded */
+    IlbcEncoder *iLBCenc_inst,    /* (i/o) Encoder instance */
+    int16_t *encoded_data,      /* (o) The encoded bytes */
+    int16_t *data               /* (i) The signal block to encode */
+                                                        ){
+
+  /* do the actual encoding */
+  WebRtcIlbcfix_Encode((uint16_t *)encoded_data, data, iLBCenc_inst);
+
+  return (iLBCenc_inst->no_of_bytes);
+}
+
+/*----------------------------------------------------------------*
+ *  Decoder interface function
+ *---------------------------------------------------------------*/
+
+short decode( /* (o) Number of decoded samples */
+    IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
+    short *decoded_data, /* (o) Decoded signal block */
+    short *encoded_data, /* (i) Encoded bytes */
+    short mode           /* (i) 0=PL, 1=Normal */
+              ){
+
+  /* check if mode is valid */
+
+  if (mode<0 || mode>1) {
+    printf("\nERROR - Wrong mode - 0, 1 allowed\n"); exit(3);}
+
+  /* do actual decoding of block */
+
+  WebRtcIlbcfix_Decode(decoded_data, (uint16_t *)encoded_data,
+                       iLBCdec_inst, mode);
+
+  return (iLBCdec_inst->blockl);
+}
+
+/*----------------------------------------------------------------*
+ *  Main program to test iLBC encoding and decoding
+ *
+ *  Usage:
+ *		exefile_name.exe <infile> <bytefile> <outfile> <channelfile>
+ *
+ *---------------------------------------------------------------*/
+
+#define MAXFRAMES   10000
+#define MAXFILELEN (BLOCKL_MAX*MAXFRAMES)
+
+int main(int argc, char* argv[])
+{
+
+  /* Runtime statistics */
+
+  float starttime1, starttime2;
+  float runtime1, runtime2;
+  float outtime;
+
+  FILE *ifileid,*efileid,*ofileid, *chfileid;
+  short *inputdata, *encodeddata, *decodeddata;
+  short *channeldata;
+  int blockcount = 0, noOfBlocks=0, i, noOfLostBlocks=0;
+  short mode;
+  IlbcEncoder Enc_Inst;
+  IlbcDecoder Dec_Inst;
+
+  short frameLen;
+  short count;
+#ifdef SPLIT_10MS
+  short size;
+#endif
+
+  inputdata=(short*) malloc(MAXFILELEN*sizeof(short));
+  if (inputdata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    exit(0);
+  }
+  encodeddata=(short*) malloc(ILBCNOOFWORDS_MAX*MAXFRAMES*sizeof(short));
+  if (encodeddata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    free(inputdata);
+    exit(0);
+  }
+  decodeddata=(short*) malloc(MAXFILELEN*sizeof(short));
+  if (decodeddata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    free(inputdata);
+    free(encodeddata);
+    exit(0);
+  }
+  channeldata=(short*) malloc(MAXFRAMES*sizeof(short));
+  if (channeldata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    free(inputdata);
+    free(encodeddata);
+    free(decodeddata);
+    exit(0);
+  }
+
+  /* get arguments and open files */
+
+  if (argc != 6 ) {
+    fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
+            argv[0]);
+    fprintf(stderr, "Example:\n");
+    fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
+    exit(1);
+  }
+  mode=atoi(argv[1]);
+  if (mode != 20 && mode != 30) {
+    fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
+    exit(2);
+  }
+  if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+    exit(2);}
+  if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open channelfile file %s\n",
+            argv[3]); exit(3);}
+  if( (ofileid=fopen(argv[4],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open output file %s\n",
+            argv[4]); exit(3);}
+  if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
+    exit(2);}
+
+
+  /* print info */
+#ifndef PRINT_MIPS
+  fprintf(stderr, "\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*      iLBCtest                                     *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+#ifdef SPLIT_10MS
+  fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
+#else
+  fprintf(stderr,"\nMode          : %2d ms\n", mode);
+#endif
+  fprintf(stderr,"\nInput file    : %s\n", argv[2]);
+  fprintf(stderr,"Coded file    : %s\n", argv[3]);
+  fprintf(stderr,"Output file   : %s\n\n", argv[4]);
+  fprintf(stderr,"Channel file  : %s\n\n", argv[5]);
+#endif
+
+  /* Initialization */
+
+  WebRtcIlbcfix_EncoderInit(&Enc_Inst, mode);
+  WebRtcIlbcfix_DecoderInit(&Dec_Inst, mode, 1);
+
+  /* extract the input file and channel file */
+
+#ifdef SPLIT_10MS
+  frameLen = (mode==20)? 80:160;
+  fread(Enc_Inst.past_samples, sizeof(short), frameLen, ifileid);
+  Enc_Inst.section = 0;
+
+  while( fread(&inputdata[noOfBlocks*80], sizeof(short),
+               80, ifileid) == 80 ) {
+    noOfBlocks++;
+  }
+
+  noOfBlocks += frameLen/80;
+  frameLen = 80;
+#else
+  frameLen = Enc_Inst.blockl;
+
+  while( fread(&inputdata[noOfBlocks*Enc_Inst.blockl],sizeof(short),
+               Enc_Inst.blockl,ifileid)==(uint16_t)Enc_Inst.blockl){
+    noOfBlocks++;
+  }
+#endif
+
+
+  while ((fread(&channeldata[blockcount],sizeof(short), 1,chfileid)==1)
+            && ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) )) {
+    blockcount++;
+  }
+
+  if ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) ) {
+    fprintf(stderr,"Channel file %s is too short\n", argv[4]);
+    free(inputdata);
+    free(encodeddata);
+    free(decodeddata);
+    free(channeldata);
+    exit(0);
+  }
+
+  count=0;
+
+  /* Runtime statistics */
+
+  starttime1 = clock()/(float)CLOCKS_PER_SEC;
+
+  /* Encoding loop */
+#ifdef PRINT_MIPS
+  printf("-1 -1\n");
+#endif
+
+#ifdef SPLIT_10MS
+  /* "Enc_Inst.section != 0" is to make sure we run through full
+     lengths of all vectors for 10ms split mode.
+  */
+  //   while( (count < noOfBlocks) || (Enc_Inst.section != 0) )    {
+  while( count < blockcount * (Enc_Inst.blockl/frameLen) )    {
+
+    encode(&Enc_Inst, &encodeddata[Enc_Inst.no_of_words *
+                                   (count/(Enc_Inst.nsub/2))],
+           &inputdata[frameLen * count] );
+#else
+    while (count < noOfBlocks) {
+      encode( &Enc_Inst, &encodeddata[Enc_Inst.no_of_words * count],
+              &inputdata[frameLen * count] );
+#endif
+
+#ifdef PRINT_MIPS
+      printf("-1 -1\n");
+#endif
+
+      count++;
+    }
+
+    count=0;
+
+    /* Runtime statistics */
+
+    starttime2=clock()/(float)CLOCKS_PER_SEC;
+    runtime1 = (float)(starttime2-starttime1);
+
+    /* Decoding loop */
+
+    while (count < blockcount) {
+      if (channeldata[count]==1) {
+        /* Normal decoding */
+        decode(&Dec_Inst, &decodeddata[count * Dec_Inst.blockl],
+               &encodeddata[Dec_Inst.no_of_words * count], 1);
+      } else if (channeldata[count]==0) {
+        /* PLC */
+        short emptydata[ILBCNOOFWORDS_MAX];
+        memset(emptydata, 0, Dec_Inst.no_of_words*sizeof(short));
+        decode(&Dec_Inst, &decodeddata[count*Dec_Inst.blockl],
+               emptydata, 0);
+        noOfLostBlocks++;
+      } else {
+        printf("Error in channel file (values have to be either 1 or 0)\n");
+        exit(0);
+      }
+#ifdef PRINT_MIPS
+      printf("-1 -1\n");
+#endif
+
+      count++;
+    }
+
+    /* Runtime statistics */
+
+    runtime2 = (float)(clock()/(float)CLOCKS_PER_SEC-starttime2);
+
+    outtime = (float)((float)blockcount*
+                      (float)mode/1000.0);
+
+#ifndef PRINT_MIPS
+    printf("\nLength of speech file: %.1f s\n", outtime);
+    printf("Lost frames          : %.1f%%\n\n", 100*(float)noOfLostBlocks/(float)blockcount);
+
+    printf("Time to run iLBC_encode+iLBC_decode:");
+    printf(" %.1f s (%.1f%% of realtime)\n", runtime1+runtime2,
+           (100*(runtime1+runtime2)/outtime));
+
+    printf("Time in iLBC_encode                :");
+    printf(" %.1f s (%.1f%% of total runtime)\n",
+           runtime1, 100.0*runtime1/(runtime1+runtime2));
+
+    printf("Time in iLBC_decode                :");
+    printf(" %.1f s (%.1f%% of total runtime)\n\n",
+           runtime2, 100.0*runtime2/(runtime1+runtime2));
+#endif
+
+    /* Write data to files */
+    for (i=0; i<blockcount; i++) {
+      fwrite(&encodeddata[i*Enc_Inst.no_of_words], sizeof(short),
+             Enc_Inst.no_of_words, efileid);
+    }
+    for (i=0;i<blockcount;i++) {
+      fwrite(&decodeddata[i*Enc_Inst.blockl],sizeof(short),Enc_Inst.blockl,ofileid);
+    }
+
+    /* return memory and close files */
+
+    free(inputdata);
+    free(encodeddata);
+    free(decodeddata);
+    free(channeldata);
+    fclose(ifileid);  fclose(efileid); fclose(ofileid);
+    return(0);
+  }
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c
new file mode 100644
index 0000000000..a9a0147b9d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c
@@ -0,0 +1,241 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_UnpackBits.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
+    const uint16_t *bitstream,    /* (i) The packatized bitstream */
+    iLBC_bits *enc_bits,  /* (o) Paramerers from bitstream */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
+                                        ) {
+  const uint16_t *bitstreamPtr;
+  int i, k;
+  int16_t *tmpPtr;
+
+  bitstreamPtr=bitstream;
+
+  /* First int16_t */
+  enc_bits->lsf[0]  =  (*bitstreamPtr)>>10;       /* Bit 0..5  */
+  enc_bits->lsf[1]  = ((*bitstreamPtr)>>3)&0x7F;      /* Bit 6..12 */
+  enc_bits->lsf[2]  = ((*bitstreamPtr)&0x7)<<4;      /* Bit 13..15 */
+  bitstreamPtr++;
+  /* Second int16_t */
+  enc_bits->lsf[2] |= ((*bitstreamPtr)>>12)&0xF;      /* Bit 0..3  */
+
+  if (mode==20) {
+    enc_bits->startIdx             = ((*bitstreamPtr)>>10)&0x3;  /* Bit 4..5  */
+    enc_bits->state_first          = ((*bitstreamPtr)>>9)&0x1;  /* Bit 6  */
+    enc_bits->idxForMax            = ((*bitstreamPtr)>>3)&0x3F;  /* Bit 7..12 */
+    enc_bits->cb_index[0]          = ((*bitstreamPtr)&0x7)<<4;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>12)&0xE;  /* Bit 0..2  */
+    enc_bits->gain_index[0]        = ((*bitstreamPtr)>>8)&0x18;  /* Bit 3..4  */
+    enc_bits->gain_index[1]        = ((*bitstreamPtr)>>7)&0x8;  /* Bit 5  */
+    enc_bits->cb_index[3]          = ((*bitstreamPtr)>>2)&0xFE;  /* Bit 6..12 */
+    enc_bits->gain_index[3]        = ((*bitstreamPtr)<<2)&0x10;  /* Bit 13  */
+    enc_bits->gain_index[4]        = ((*bitstreamPtr)<<2)&0x8;  /* Bit 14  */
+    enc_bits->gain_index[6]        = ((*bitstreamPtr)<<4)&0x10;  /* Bit 15  */
+  } else { /* mode==30 */
+    enc_bits->lsf[3]               = ((*bitstreamPtr)>>6)&0x3F;  /* Bit 4..9  */
+    enc_bits->lsf[4]               = ((*bitstreamPtr)<<1)&0x7E;  /* Bit 10..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    enc_bits->lsf[4]              |= ((*bitstreamPtr)>>15)&0x1;  /* Bit 0  */
+    enc_bits->lsf[5]               = ((*bitstreamPtr)>>8)&0x7F;  /* Bit 1..7  */
+    enc_bits->startIdx             = ((*bitstreamPtr)>>5)&0x7;  /* Bit 8..10 */
+    enc_bits->state_first          = ((*bitstreamPtr)>>4)&0x1;  /* Bit 11  */
+    enc_bits->idxForMax            = ((*bitstreamPtr)<<2)&0x3C;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 4:th int16_t */
+    enc_bits->idxForMax           |= ((*bitstreamPtr)>>14)&0x3;  /* Bit 0..1  */
+    enc_bits->cb_index[0]        = ((*bitstreamPtr)>>7)&0x78;  /* Bit 2..5  */
+    enc_bits->gain_index[0]        = ((*bitstreamPtr)>>5)&0x10;  /* Bit 6  */
+    enc_bits->gain_index[1]        = ((*bitstreamPtr)>>5)&0x8;  /* Bit 7  */
+    enc_bits->cb_index[3]          = ((*bitstreamPtr))&0xFC;  /* Bit 8..13 */
+    enc_bits->gain_index[3]        = ((*bitstreamPtr)<<3)&0x10;  /* Bit 14  */
+    enc_bits->gain_index[4]        = ((*bitstreamPtr)<<3)&0x8;  /* Bit 15  */
+  }
+  /* Class 2 bits of ULP */
+  /* 4:th to 6:th int16_t for 20 ms case
+     5:th to 7:th int16_t for 30 ms case */
+  bitstreamPtr++;
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<3; k++) {
+    for (i=15; i>=0; i--) {
+      (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 7:th int16_t */
+    for (i=15; i>6; i--) {
+      (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>4)&0x4; /* Bit 9  */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>1)&0x4; /* Bit 12  */
+    enc_bits->gain_index[6]       |= ((*bitstreamPtr)<<1)&0x8; /* Bit 13  */
+    enc_bits->gain_index[7]        = ((*bitstreamPtr)<<2)&0xC; /* Bit 14..15 */
+
+  } else { /* mode==30 */
+    /* 8:th int16_t */
+    for (i=15; i>5; i--) {
+      (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>3)&0x6; /* Bit 10..11 */
+    enc_bits->gain_index[0]       |= ((*bitstreamPtr))&0x8;  /* Bit 12  */
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr))&0x4;  /* Bit 13  */
+    enc_bits->cb_index[3]         |= ((*bitstreamPtr))&0x2;  /* Bit 14  */
+    enc_bits->cb_index[6]          = ((*bitstreamPtr)<<7)&0x80; /* Bit 15  */
+    bitstreamPtr++;
+    /* 9:th int16_t */
+    enc_bits->cb_index[6]         |= ((*bitstreamPtr)>>9)&0x7E; /* Bit 0..5  */
+    enc_bits->cb_index[9]          = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */
+    enc_bits->cb_index[12]         = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 10:th int16_t */
+    enc_bits->cb_index[12]         |= ((*bitstreamPtr)>>11)&0x1E;/* Bit 0..3 */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>8)&0xC; /* Bit 4..5  */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>7)&0x6; /* Bit 6..7  */
+    enc_bits->gain_index[6]        = ((*bitstreamPtr)>>3)&0x18; /* Bit 8..9  */
+    enc_bits->gain_index[7]        = ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
+    enc_bits->gain_index[9]        = ((*bitstreamPtr)<<1)&0x10; /* Bit 12  */
+    enc_bits->gain_index[10]       = ((*bitstreamPtr)<<1)&0x8; /* Bit 13  */
+    enc_bits->gain_index[12]       = ((*bitstreamPtr)<<3)&0x10; /* Bit 14  */
+    enc_bits->gain_index[13]       = ((*bitstreamPtr)<<3)&0x8; /* Bit 15  */
+  }
+  bitstreamPtr++;
+  /* Class 3 bits of ULP */
+  /*  8:th to 14:th int16_t for 20 ms case
+      11:th to 17:th int16_t for 30 ms case */
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<7; k++) {
+    for (i=14; i>=0; i-=2) {
+      (*tmpPtr)                 |= ((*bitstreamPtr)>>i)&0x3; /* Bit 15-i..14-i*/
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 15:th int16_t */
+    enc_bits->idxVec[56]          |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>13)&0x1; /* Bit 2  */
+    enc_bits->cb_index[1]          = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9  */
+    enc_bits->cb_index[2]          = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 16:th int16_t */
+    enc_bits->cb_index[2]         |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0  */
+    enc_bits->gain_index[0]       |= ((*bitstreamPtr)>>12)&0x7; /* Bit 1..3  */
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5  */
+    enc_bits->gain_index[2]        = ((*bitstreamPtr)>>7)&0x7; /* Bit 6..8  */
+    enc_bits->cb_index[3]         |= ((*bitstreamPtr)>>6)&0x1; /* Bit 9  */
+    enc_bits->cb_index[4]          = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 17:th int16_t */
+    enc_bits->cb_index[4]         |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0  */
+    enc_bits->cb_index[5]          = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7  */
+    enc_bits->cb_index[6]          = ((*bitstreamPtr))&0xFF; /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 18:th int16_t */
+    enc_bits->cb_index[7]          = (*bitstreamPtr)>>8;  /* Bit 0..7  */
+    enc_bits->cb_index[8]          = (*bitstreamPtr)&0xFF;  /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3  */
+    enc_bits->gain_index[5]        = ((*bitstreamPtr)>>9)&0x7; /* Bit 4..6  */
+    enc_bits->gain_index[6]       |= ((*bitstreamPtr)>>6)&0x7; /* Bit 7..9  */
+    enc_bits->gain_index[7]       |= ((*bitstreamPtr)>>4)&0x3; /* Bit 10..11 */
+    enc_bits->gain_index[8]        = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
+  } else { /* mode==30 */
+    /* 18:th int16_t */
+    enc_bits->idxVec[56]          |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->idxVec[57]          |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3  */
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>11)&1; /* Bit 4  */
+    enc_bits->cb_index[1]          = ((*bitstreamPtr)>>4)&0x7F; /* Bit 5..11 */
+    enc_bits->cb_index[2]          = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    enc_bits->cb_index[2]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->gain_index[0]       |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5  */
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7  */
+    enc_bits->gain_index[2]        = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
+    enc_bits->cb_index[3]         |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11  */
+    enc_bits->cb_index[4]          = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 20:th int16_t */
+    enc_bits->cb_index[4]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->cb_index[5]          = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9  */
+    enc_bits->cb_index[6]         |= ((*bitstreamPtr)>>5)&0x1; /* Bit 10  */
+    enc_bits->cb_index[7]          = ((*bitstreamPtr)<<3)&0xF8; /* Bit 11..15 */
+    bitstreamPtr++;
+    /* 21:st int16_t */
+    enc_bits->cb_index[7]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->cb_index[8]          = ((*bitstreamPtr)>>5)&0xFF; /* Bit 3..10 */
+    enc_bits->cb_index[9]         |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11  */
+    enc_bits->cb_index[10]         = ((*bitstreamPtr)<<4)&0xF0; /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 22:nd int16_t */
+    enc_bits->cb_index[10]        |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3  */
+    enc_bits->cb_index[11]         = ((*bitstreamPtr)>>4)&0xFF; /* Bit 4..11 */
+    enc_bits->cb_index[12]        |= ((*bitstreamPtr)>>3)&0x1; /* Bit 12  */
+    enc_bits->cb_index[13]         = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 23:rd int16_t */
+    enc_bits->cb_index[13]        |= ((*bitstreamPtr)>>11)&0x1F;/* Bit 0..4  */
+    enc_bits->cb_index[14]         = ((*bitstreamPtr)>>3)&0xFF; /* Bit 5..12 */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>1)&0x3; /* Bit 13..14 */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)&0x1);  /* Bit 15  */
+    bitstreamPtr++;
+    /* 24:rd int16_t */
+    enc_bits->gain_index[5]        = ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->gain_index[6]       |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5  */
+    enc_bits->gain_index[7]       |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7  */
+    enc_bits->gain_index[8]        = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
+    enc_bits->gain_index[9]       |= ((*bitstreamPtr)>>1)&0xF; /* Bit 11..14 */
+    enc_bits->gain_index[10]      |= ((*bitstreamPtr)<<2)&0x4; /* Bit 15  */
+    bitstreamPtr++;
+    /* 25:rd int16_t */
+    enc_bits->gain_index[10]      |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->gain_index[11]       = ((*bitstreamPtr)>>11)&0x7; /* Bit 2..4  */
+    enc_bits->gain_index[12]      |= ((*bitstreamPtr)>>7)&0xF; /* Bit 5..8  */
+    enc_bits->gain_index[13]      |= ((*bitstreamPtr)>>4)&0x7; /* Bit 9..11 */
+    enc_bits->gain_index[14]       = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
+  }
+  /* Last bit should be zero, otherwise it's an "empty" frame */
+  if (((*bitstreamPtr)&0x1) == 1) {
+    return(1);
+  } else {
+    return(0);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h
new file mode 100644
index 0000000000..1a63280e6b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_UnpackBits.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+int16_t
+WebRtcIlbcfix_UnpackBits(/* (o) "Empty" frame indicator */
+                         const uint16_t*
+                             bitstream, /* (i) The packatized bitstream */
+                         iLBC_bits*
+                             enc_bits, /* (o) Paramerers from bitstream */
+                         int16_t mode  /* (i) Codec mode (20 or 30) */
+                         );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c
new file mode 100644
index 0000000000..d9375fb995
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq3.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/vq3.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+/*----------------------------------------------------------------*
+ *  vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq3(
+    int16_t *Xq, /* quantized vector (Q13) */
+    int16_t *index,
+    int16_t *CB, /* codebook in Q13 */
+    int16_t *X,  /* vector to quantize (Q13) */
+    int16_t n_cb
+                       ){
+  int16_t i, j;
+  int16_t pos, minindex=0;
+  int16_t tmp;
+  int32_t dist, mindist;
+
+  pos = 0;
+  mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
+
+  /* Find the codebook with the lowest square distance */
+  for (j = 0; j < n_cb; j++) {
+    tmp = X[0] - CB[pos];
+    dist = tmp * tmp;
+    for (i = 1; i < 3; i++) {
+      tmp = X[i] - CB[pos + i];
+      dist += tmp * tmp;
+    }
+
+    if (dist < mindist) {
+      mindist = dist;
+      minindex = j;
+    }
+    pos += 3;
+  }
+
+  /* Store the quantized codebook and the index */
+  for (i = 0; i < 3; i++) {
+    Xq[i] = CB[minindex*3 + i];
+  }
+  *index = minindex;
+
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.h
new file mode 100644
index 0000000000..c946478a1a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq3.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Vector quantization of order 3 (based on MSE)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq3(
+    int16_t* Xq,    /* (o) the quantized vector (Q13) */
+    int16_t* index, /* (o) the quantization index */
+    int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
+    int16_t* X,     /* (i) the vector to quantize (Q13) */
+    int16_t n_cb    /* (i) the number of vectors in the codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c
new file mode 100644
index 0000000000..c9a65aec2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq4.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/vq4.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+/*----------------------------------------------------------------*
+ *  vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq4(
+    int16_t *Xq, /* quantized vector (Q13) */
+    int16_t *index,
+    int16_t *CB, /* codebook in Q13 */
+    int16_t *X,  /* vector to quantize (Q13) */
+    int16_t n_cb
+                       ){
+  int16_t i, j;
+  int16_t pos, minindex=0;
+  int16_t tmp;
+  int32_t dist, mindist;
+
+  pos = 0;
+  mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
+
+  /* Find the codebook with the lowest square distance */
+  for (j = 0; j < n_cb; j++) {
+    tmp = X[0] - CB[pos];
+    dist = tmp * tmp;
+    for (i = 1; i < 4; i++) {
+      tmp = X[i] - CB[pos + i];
+      dist += tmp * tmp;
+    }
+
+    if (dist < mindist) {
+      mindist = dist;
+      minindex = j;
+    }
+    pos += 4;
+  }
+
+  /* Store the quantized codebook and the index */
+  for (i = 0; i < 4; i++) {
+    Xq[i] = CB[minindex*4 + i];
+  }
+  *index = minindex;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.h
new file mode 100644
index 0000000000..6d14830c03
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq4.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Vector quantization of order 4 (based on MSE)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq4(
+    int16_t* Xq,    /* (o) the quantized vector (Q13) */
+    int16_t* index, /* (o) the quantization index */
+    int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
+    int16_t* X,     /* (i) the vector to quantize (Q13) */
+    int16_t n_cb    /* (i) the number of vectors in the codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
new file mode 100644
index 0000000000..e82d167220
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Window32W32.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/window32_w32.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  window multiplication
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Window32W32(
+    int32_t *z,    /* Output */
+    int32_t *x,    /* Input (same domain as Output)*/
+    const int32_t  *y,  /* Q31 Window */
+    size_t N     /* length to process */
+                               ) {
+  size_t i;
+  int16_t x_low, x_hi, y_low, y_hi;
+  int16_t left_shifts;
+  int32_t temp;
+
+  left_shifts = (int16_t)WebRtcSpl_NormW32(x[0]);
+  WebRtcSpl_VectorBitShiftW32(x, N, x, (int16_t)(-left_shifts));
+
+
+  /* The double precision numbers use a special representation:
+   * w32 = hi<<16 + lo<<1
+   */
+  for (i = 0; i < N; i++) {
+    /* Extract higher bytes */
+    x_hi = (int16_t)(x[i] >> 16);
+    y_hi = (int16_t)(y[i] >> 16);
+
+    /* Extract lower bytes, defined as (w32 - hi<<16)>>1 */
+    x_low = (int16_t)((x[i] - (x_hi << 16)) >> 1);
+
+    y_low = (int16_t)((y[i] - (y_hi << 16)) >> 1);
+
+    /* Calculate z by a 32 bit multiplication using both low and high from x and y */
+    temp = ((x_hi * y_hi) << 1) + ((x_hi * y_low) >> 14);
+
+    z[i] = temp + ((x_low * y_hi) >> 14);
+  }
+
+  WebRtcSpl_VectorBitShiftW32(z, N, z, left_shifts);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
new file mode 100644
index 0000000000..15d72c5ba2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Window32W32.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  window multiplication
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Window32W32(int32_t* z, /* Output */
+                               int32_t* x, /* Input (same domain as Output)*/
+                               const int32_t* y, /* Q31 Window */
+                               size_t N          /* length to process */
+                               );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
new file mode 100644
index 0000000000..9dc880b37e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
@@ -0,0 +1,142 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_XcorrCoef.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * cross correlation which finds the optimal lag for the
+ * crossCorr*crossCorr/(energy) criteria
+ *---------------------------------------------------------------*/
+
+size_t WebRtcIlbcfix_XcorrCoef(
+    int16_t *target,  /* (i) first array */
+    int16_t *regressor, /* (i) second array */
+    size_t subl,  /* (i) dimension arrays */
+    size_t searchLen, /* (i) the search lenght */
+    size_t offset,  /* (i) samples offset between arrays */
+    int16_t step   /* (i) +1 or -1 */
+                            ){
+  size_t k;
+  size_t maxlag;
+  int16_t pos;
+  int16_t max;
+  int16_t crossCorrScale, Energyscale;
+  int16_t crossCorrSqMod, crossCorrSqMod_Max;
+  int32_t crossCorr, Energy;
+  int16_t crossCorrmod, EnergyMod, EnergyMod_Max;
+  int16_t *tp, *rp;
+  int16_t *rp_beg, *rp_end;
+  int16_t totscale, totscale_max;
+  int16_t scalediff;
+  int32_t newCrit, maxCrit;
+  int shifts;
+
+  /* Initializations, to make sure that the first one is selected */
+  crossCorrSqMod_Max=0;
+  EnergyMod_Max=WEBRTC_SPL_WORD16_MAX;
+  totscale_max=-500;
+  maxlag=0;
+  pos=0;
+
+  /* Find scale value and start position */
+  if (step==1) {
+    max=WebRtcSpl_MaxAbsValueW16(regressor, subl + searchLen - 1);
+    rp_beg = regressor;
+    rp_end = regressor + subl;
+  } else { /* step==-1 */
+    max = WebRtcSpl_MaxAbsValueW16(regressor - searchLen, subl + searchLen - 1);
+    rp_beg = regressor - 1;
+    rp_end = regressor + subl - 1;
+  }
+
+  /* Introduce a scale factor on the Energy in int32_t in
+     order to make sure that the calculation does not
+     overflow */
+
+  if (max>5000) {
+    shifts=2;
+  } else {
+    shifts=0;
+  }
+
+  /* Calculate the first energy, then do a +/- to get the other energies */
+  Energy=WebRtcSpl_DotProductWithScale(regressor, regressor, subl, shifts);
+
+  for (k=0;k<searchLen;k++) {
+    tp = target;
+    rp = &regressor[pos];
+
+    crossCorr=WebRtcSpl_DotProductWithScale(tp, rp, subl, shifts);
+
+    if ((Energy>0)&&(crossCorr>0)) {
+
+      /* Put cross correlation and energy on 16 bit word */
+      crossCorrScale=(int16_t)WebRtcSpl_NormW32(crossCorr)-16;
+      crossCorrmod=(int16_t)WEBRTC_SPL_SHIFT_W32(crossCorr, crossCorrScale);
+      Energyscale=(int16_t)WebRtcSpl_NormW32(Energy)-16;
+      EnergyMod=(int16_t)WEBRTC_SPL_SHIFT_W32(Energy, Energyscale);
+
+      /* Square cross correlation and store upper int16_t */
+      crossCorrSqMod = (int16_t)((crossCorrmod * crossCorrmod) >> 16);
+
+      /* Calculate the total number of (dynamic) right shifts that have
+         been performed on (crossCorr*crossCorr)/energy
+      */
+      totscale=Energyscale-(crossCorrScale<<1);
+
+      /* Calculate the shift difference in order to be able to compare the two
+         (crossCorr*crossCorr)/energy in the same domain
+      */
+      scalediff=totscale-totscale_max;
+      scalediff=WEBRTC_SPL_MIN(scalediff,31);
+      scalediff=WEBRTC_SPL_MAX(scalediff,-31);
+
+      /* Compute the cross multiplication between the old best criteria
+         and the new one to be able to compare them without using a
+         division */
+
+      if (scalediff<0) {
+        newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max)>>(-scalediff);
+        maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod);
+      } else {
+        newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max);
+        maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod)>>scalediff;
+      }
+
+      /* Store the new lag value if the new criteria is larger
+         than previous largest criteria */
+
+      if (newCrit > maxCrit) {
+        crossCorrSqMod_Max = crossCorrSqMod;
+        EnergyMod_Max = EnergyMod;
+        totscale_max = totscale;
+        maxlag = k;
+      }
+    }
+    pos+=step;
+
+    /* Do a +/- to get the next energy */
+    Energy += step * ((*rp_end * *rp_end - *rp_beg * *rp_beg) >> shifts);
+    rp_beg+=step;
+    rp_end+=step;
+  }
+
+  return(maxlag+offset);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
new file mode 100644
index 0000000000..3be5a296b5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_XcorrCoef.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * cross correlation which finds the optimal lag for the
+ * crossCorr*crossCorr/(energy) criteria
+ *---------------------------------------------------------------*/
+
+size_t WebRtcIlbcfix_XcorrCoef(
+    int16_t* target,    /* (i) first array */
+    int16_t* regressor, /* (i) second array */
+    size_t subl,        /* (i) dimension arrays */
+    size_t searchLen,   /* (i) the search lenght */
+    size_t offset,      /* (i) samples offset between arrays */
+    int16_t step        /* (i) +1 or -1 */
+    );
+
+#endif