diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/ilbc')
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..4a3f004ed3 --- /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..022c113dda --- /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..15dc884f2a --- /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..c489ab54f9 --- /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..4b3703182e --- /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..11856649e7 --- /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..5a3b13e446 --- /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..8ba82927b8 --- /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..d9df9a78f8 --- /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..5bed469a12 --- /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..40510007a9 --- /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..5e3bcc6d3c --- /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..5290420bbf --- /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..3a11488056 --- /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..20c85fb20e --- /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..0c631bcb86 --- /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(®ressor[-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..d0f5f1a4ed --- /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..dee67cc5f9 --- /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..b5e6cef97b --- /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..fab9718a75 --- /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..87030e568f --- /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..4d3f733355 --- /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..0172ac416b --- /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..92f9ad68e7 --- /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..4a233fb946 --- /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..f4fa97d477 --- /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..ca050b02cc --- /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..a0ccfa96ac --- /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..08d1e8325a --- /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..fccc3c2b1c --- /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..06f292f038 --- /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..a0dfb8e8eb --- /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..bc29b44393 --- /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..6db30b3e15 --- /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..d2ebeeeda9 --- /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..d10f84126e --- /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..d95173689a --- /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..29be89e35a --- /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..7e7e10e62a --- /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..90e0c4a3ba --- /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..00b126af7e --- /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..38dcdfa59d --- /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..12da5cdea5 --- /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..a40661fb80 --- /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..79d3cd12ee --- /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..0590329b08 --- /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..7a215e43d3 --- /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..2e517743ce --- /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..1ef5e1a7db --- /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..33d06b8ad0 --- /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..0337368bcb --- /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..93bb72e998 --- /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 = ®ressor[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..3fcce25147 --- /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 |