Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

7 files changed, 800 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
new file mode 100644
index 0000000000..46ac671b30
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -0,0 +1,102 @@
+/*
+ *  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/g711/audio_decoder_pcm.h"
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+namespace webrtc {
+
+void AudioDecoderPcmU::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmU::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmU::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted =
+      PacketDuration(encoded, encoded_len) *
+      Channels();         // 1 byte per sample per channel
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG711_DecodeU(encoded, encoded_len_adjusted, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+void AudioDecoderPcmA::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmA::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmA::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted =
+      PacketDuration(encoded, encoded_len) *
+      Channels();         // 1 byte per sample per channel
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG711_DecodeA(encoded, encoded_len_adjusted, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
new file mode 100644
index 0000000000..3fa42cba30
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -0,0 +1,81 @@
+/*
+ *  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_G711_AUDIO_DECODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class AudioDecoderPcmU final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1);
+  }
+
+  AudioDecoderPcmU(const AudioDecoderPcmU&) = delete;
+  AudioDecoderPcmU& operator=(const AudioDecoderPcmU&) = delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const 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:
+  const size_t num_channels_;
+};
+
+class AudioDecoderPcmA final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1);
+  }
+
+  AudioDecoderPcmA(const AudioDecoderPcmA&) = delete;
+  AudioDecoderPcmA& operator=(const AudioDecoderPcmA&) = delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const 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:
+  const size_t num_channels_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
new file mode 100644
index 0000000000..65e2da479d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -0,0 +1,126 @@
+/*
+ *  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/g711/audio_encoder_pcm.h"
+
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+bool AudioEncoderPcm::Config::IsOk() const {
+  return (frame_size_ms % 10 == 0) && (num_channels >= 1);
+}
+
+AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
+    : sample_rate_hz_(sample_rate_hz),
+      num_channels_(config.num_channels),
+      payload_type_(config.payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      full_frame_samples_(config.num_channels * config.frame_size_ms *
+                          sample_rate_hz / 1000),
+      first_timestamp_in_buffer_(0) {
+  RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
+  RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
+      << "Frame size must be an integer multiple of 10 ms.";
+  speech_buffer_.reserve(full_frame_samples_);
+}
+
+AudioEncoderPcm::~AudioEncoderPcm() = default;
+
+int AudioEncoderPcm::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioEncoderPcm::NumChannels() const {
+  return num_channels_;
+}
+
+size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderPcm::GetTargetBitrate() const {
+  return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
+                          NumChannels());
+}
+
+AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (speech_buffer_.empty()) {
+    first_timestamp_in_buffer_ = rtp_timestamp;
+  }
+  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
+  if (speech_buffer_.size() < full_frame_samples_) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
+  EncodedInfo info;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoded_bytes = encoded->AppendData(
+      full_frame_samples_ * BytesPerSample(),
+      [&](rtc::ArrayView<uint8_t> encoded) {
+        return EncodeCall(&speech_buffer_[0], full_frame_samples_,
+                          encoded.data());
+      });
+  speech_buffer_.clear();
+  info.encoder_type = GetCodecType();
+  return info;
+}
+
+void AudioEncoderPcm::Reset() {
+  speech_buffer_.clear();
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderPcm::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeA(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmA::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmA;
+}
+
+size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeU(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmU::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmU;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
new file mode 100644
index 0000000000..d50be4b457
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -0,0 +1,128 @@
+/*
+ *  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_G711_AUDIO_ENCODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+class AudioEncoderPcm : public AudioEncoder {
+ public:
+  struct Config {
+   public:
+    bool IsOk() const;
+
+    int frame_size_ms;
+    size_t num_channels;
+    int payload_type;
+
+   protected:
+    explicit Config(int pt)
+        : frame_size_ms(20), num_channels(1), payload_type(pt) {}
+  };
+
+  ~AudioEncoderPcm() override;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  AudioEncoderPcm(const Config& config, int sample_rate_hz);
+
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+  virtual size_t EncodeCall(const int16_t* audio,
+                            size_t input_len,
+                            uint8_t* encoded) = 0;
+
+  virtual size_t BytesPerSample() const = 0;
+
+  // Used to set EncodedInfoLeaf::encoder_type in
+  // AudioEncoderPcm::EncodeImpl
+  virtual AudioEncoder::CodecType GetCodecType() const = 0;
+
+ private:
+  const int sample_rate_hz_;
+  const size_t num_channels_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  const size_t full_frame_samples_;
+  std::vector<int16_t> speech_buffer_;
+  uint32_t first_timestamp_in_buffer_;
+};
+
+class AudioEncoderPcmA final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+    Config() : AudioEncoderPcm::Config(8) {}
+  };
+
+  explicit AudioEncoderPcmA(const Config& config)
+      : AudioEncoderPcm(config, kSampleRateHz) {}
+
+  AudioEncoderPcmA(const AudioEncoderPcmA&) = delete;
+  AudioEncoderPcmA& operator=(const AudioEncoderPcmA&) = delete;
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+
+ private:
+  static const int kSampleRateHz = 8000;
+};
+
+class AudioEncoderPcmU final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+    Config() : AudioEncoderPcm::Config(0) {}
+  };
+
+  explicit AudioEncoderPcmU(const Config& config)
+      : AudioEncoderPcm(config, kSampleRateHz) {}
+
+  AudioEncoderPcmU(const AudioEncoderPcmU&) = delete;
+  AudioEncoderPcmU& operator=(const AudioEncoderPcmU&) = delete;
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+
+ private:
+  static const int kSampleRateHz = 8000;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c
new file mode 100644
index 0000000000..5fe1692ccb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c
@@ -0,0 +1,59 @@
+/*
+ *  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.
+ */
+
+#include <string.h>
+
+#include "modules/third_party/g711/g711.h"
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    encoded[n] = linear_to_alaw(speechIn[n]);
+  return len;
+}
+
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    encoded[n] = linear_to_ulaw(speechIn[n]);
+  return len;
+}
+
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    decoded[n] = alaw_to_linear(encoded[n]);
+  *speechType = 1;
+  return len;
+}
+
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    decoded[n] = ulaw_to_linear(encoded[n]);
+  *speechType = 1;
+  return len;
+}
+
+int16_t WebRtcG711_Version(char* version, int16_t lenBytes) {
+  strncpy(version, "2.0.0", lenBytes);
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h
new file mode 100644
index 0000000000..c92e6cc1c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h
@@ -0,0 +1,136 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+// Comfort noise constants
+#define G711_WEBRTC_SPEECH 1
+#define G711_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcG711_EncodeA(...)
+ *
+ * This function encodes a G711 A-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ *      - speechIn           : Input speech vector
+ *      - len                : Samples in speechIn
+ *
+ * Output:
+ *      - encoded            : The encoded data vector
+ *
+ * Return value              : Length (in bytes) of coded data.
+ *                             Always equal to len input parameter.
+ */
+
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG711_EncodeU(...)
+ *
+ * This function encodes a G711 U-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ *      - speechIn           : Input speech vector
+ *      - len                : Samples in speechIn
+ *
+ * Output:
+ *      - encoded            : The encoded data vector
+ *
+ * Return value              : Length (in bytes) of coded data.
+ *                             Always equal to len input parameter.
+ */
+
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG711_DecodeA(...)
+ *
+ * This function decodes a packet G711 A-law frame.
+ *
+ * Input:
+ *      - encoded            : Encoded data
+ *      - len                : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - speechType         : 1 normal, 2 CNG (for G711 it should
+ *                             always return 1 since G711 does not have a
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples in decoded vector
+ *                             -1 - Error
+ */
+
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcG711_DecodeU(...)
+ *
+ * This function decodes a packet G711 U-law frame.
+ *
+ * Input:
+ *      - encoded            : Encoded data
+ *      - len                : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - speechType         : 1 normal, 2 CNG (for G711 it should
+ *                             always return 1 since G711 does not have a
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples in decoded vector
+ *                             -1 - Error
+ */
+
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
+
+/**********************************************************************
+ * WebRtcG711_Version(...)
+ *
+ * This function gives the version string of the G.711 codec.
+ *
+ * Input:
+ *      - lenBytes:     the size of Allocated space (in Bytes) where
+ *                      the version number is written to (in string format).
+ *
+ * Output:
+ *      - version:      Pointer to a buffer where the version number is
+ *                      written to.
+ *
+ */
+
+int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc
new file mode 100644
index 0000000000..f3a42f5d79
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc
@@ -0,0 +1,168 @@
+/*
+ *  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.
+ */
+
+/*
+ * testG711.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+/* include API */
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G711 1000
+
+/* function for reading audio data from PCM file */
+bool readframe(int16_t* data, FILE* inp, size_t length) {
+  size_t rlen = fread(data, sizeof(int16_t), length, inp);
+  if (rlen >= length)
+    return false;
+  memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+  return true;
+}
+
+int main(int argc, char* argv[]) {
+  char inname[80], outname[40], bitname[40];
+  FILE* inp;
+  FILE* outp;
+  FILE* bitp = NULL;
+  int framecnt;
+  bool endfile;
+
+  size_t framelength = 80;
+
+  /* Runtime statistics */
+  double starttime;
+  double runtime;
+  double length_file;
+
+  size_t stream_len = 0;
+  int16_t shortdata[480];
+  int16_t decoded[480];
+  uint8_t streamdata[1000];
+  int16_t speechType[1];
+  char law[2];
+  char versionNumber[40];
+
+  /* handling wrong input arguments in the command line */
+  if ((argc != 5) && (argc != 6)) {
+    printf("\n\nWrong number of arguments or flag values.\n\n");
+
+    printf("\n");
+    printf("\nG.711 test application\n\n");
+    printf("Usage:\n\n");
+    printf("./testG711.exe framelength law infile outfile \n\n");
+    printf("framelength: Framelength in samples.\n");
+    printf("law        : Coding law, A och u.\n");
+    printf("infile     : Normal speech input file\n");
+    printf("outfile    : Speech output file\n\n");
+    printf("outbits    : Output bitstream file [optional]\n\n");
+    exit(0);
+  }
+
+  /* Get version and print */
+  WebRtcG711_Version(versionNumber, 40);
+
+  printf("-----------------------------------\n");
+  printf("G.711 version: %s\n\n", versionNumber);
+  /* Get frame length */
+  int framelength_int = atoi(argv[1]);
+  if (framelength_int < 0) {
+    printf("  G.722: Invalid framelength %d.\n", framelength_int);
+    exit(1);
+  }
+  framelength = static_cast<size_t>(framelength_int);
+
+  /* Get compression law */
+  strcpy(law, argv[2]);
+
+  /* Get Input and Output files */
+  sscanf(argv[3], "%s", inname);
+  sscanf(argv[4], "%s", outname);
+  if (argc == 6) {
+    sscanf(argv[5], "%s", bitname);
+    if ((bitp = fopen(bitname, "wb")) == NULL) {
+      printf("  G.711: Cannot read file %s.\n", bitname);
+      exit(1);
+    }
+  }
+
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  G.711: Cannot read file %s.\n", inname);
+    exit(1);
+  }
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  G.711: Cannot write file %s.\n", outname);
+    exit(1);
+  }
+  printf("\nInput:  %s\nOutput: %s\n", inname, outname);
+  if (argc == 6) {
+    printf("\nBitfile:  %s\n", bitname);
+  }
+
+  starttime = clock() / (double)CLOCKS_PER_SEC_G711; /* Runtime statistics */
+
+  /* Initialize encoder and decoder */
+  framecnt = 0;
+  endfile = false;
+  while (!endfile) {
+    framecnt++;
+    /* Read speech block */
+    endfile = readframe(shortdata, inp, framelength);
+
+    /* G.711 encoding */
+    if (!strcmp(law, "A")) {
+      /* A-law encoding */
+      stream_len = WebRtcG711_EncodeA(shortdata, framelength, streamdata);
+      if (argc == 6) {
+        /* Write bits to file */
+        if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
+            stream_len) {
+          return -1;
+        }
+      }
+      WebRtcG711_DecodeA(streamdata, stream_len, decoded, speechType);
+    } else if (!strcmp(law, "u")) {
+      /* u-law encoding */
+      stream_len = WebRtcG711_EncodeU(shortdata, framelength, streamdata);
+      if (argc == 6) {
+        /* Write bits to file */
+        if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
+            stream_len) {
+          return -1;
+        }
+      }
+      WebRtcG711_DecodeU(streamdata, stream_len, decoded, speechType);
+    } else {
+      printf("Wrong law mode\n");
+      exit(1);
+    }
+    /* Write coded speech to file */
+    if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) {
+      return -1;
+    }
+  }
+
+  runtime = (double)(clock() / (double)CLOCKS_PER_SEC_G711 - starttime);
+  length_file = ((double)framecnt * (double)framelength / 8000);
+  printf("\n\nLength of speech file: %.1f s\n", length_file);
+  printf("Time to run G.711:      %.2f s (%.2f %% of realtime)\n\n", runtime,
+         (100 * runtime / length_file));
+  printf("---------------------END----------------------\n");
+
+  fclose(inp);
+  fclose(outp);
+
+  return 0;
+}