| /* |
| * 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. |
| */ |
| |
| #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" |
| |
| #include <assert.h> |
| #include <string.h> // memmove |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" |
| #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" |
| #ifdef WEBRTC_CODEC_G722 |
| #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" |
| #endif |
| #ifdef WEBRTC_CODEC_ILBC |
| #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h" |
| #endif |
| #ifdef WEBRTC_CODEC_ISACFX |
| #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h" |
| #endif |
| #ifdef WEBRTC_CODEC_ISAC |
| #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h" |
| #endif |
| #ifdef WEBRTC_CODEC_OPUS |
| #include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" |
| #endif |
| #ifdef WEBRTC_CODEC_PCM16 |
| #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" |
| #endif |
| |
| namespace webrtc { |
| |
| // PCMu |
| |
| int AudioDecoderPcmU::Init() { |
| return 0; |
| } |
| size_t AudioDecoderPcmU::Channels() const { |
| return 1; |
| } |
| |
| int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| DCHECK_EQ(sample_rate_hz, 8000); |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcG711_DecodeU(encoded, static_cast<int16_t>(encoded_len), |
| decoded, &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded, |
| size_t encoded_len) const { |
| // One encoded byte per sample per channel. |
| return static_cast<int>(encoded_len / Channels()); |
| } |
| |
| size_t AudioDecoderPcmUMultiCh::Channels() const { |
| return channels_; |
| } |
| |
| // PCMa |
| |
| int AudioDecoderPcmA::Init() { |
| return 0; |
| } |
| size_t AudioDecoderPcmA::Channels() const { |
| return 1; |
| } |
| |
| int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| DCHECK_EQ(sample_rate_hz, 8000); |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcG711_DecodeA(encoded, static_cast<int16_t>(encoded_len), |
| decoded, &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded, |
| size_t encoded_len) const { |
| // One encoded byte per sample per channel. |
| return static_cast<int>(encoded_len / Channels()); |
| } |
| |
| size_t AudioDecoderPcmAMultiCh::Channels() const { |
| return channels_; |
| } |
| |
| // PCM16B |
| #ifdef WEBRTC_CODEC_PCM16 |
| AudioDecoderPcm16B::AudioDecoderPcm16B() {} |
| |
| int AudioDecoderPcm16B::Init() { |
| return 0; |
| } |
| size_t AudioDecoderPcm16B::Channels() const { |
| return 1; |
| } |
| |
| int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 || |
| sample_rate_hz == 32000 || sample_rate_hz == 48000) |
| << "Unsupported sample rate " << sample_rate_hz; |
| int16_t ret = |
| WebRtcPcm16b_Decode(encoded, static_cast<int16_t>(encoded_len), decoded); |
| *speech_type = ConvertSpeechType(1); |
| return ret; |
| } |
| |
| int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded, |
| size_t encoded_len) const { |
| // Two encoded byte per sample per channel. |
| return static_cast<int>(encoded_len / (2 * Channels())); |
| } |
| |
| AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels) |
| : channels_(num_channels) { |
| DCHECK(num_channels > 0); |
| } |
| |
| size_t AudioDecoderPcm16BMultiCh::Channels() const { |
| return channels_; |
| } |
| #endif |
| |
| // iLBC |
| #ifdef WEBRTC_CODEC_ILBC |
| AudioDecoderIlbc::AudioDecoderIlbc() { |
| WebRtcIlbcfix_DecoderCreate(&dec_state_); |
| } |
| |
| AudioDecoderIlbc::~AudioDecoderIlbc() { |
| WebRtcIlbcfix_DecoderFree(dec_state_); |
| } |
| |
| bool AudioDecoderIlbc::HasDecodePlc() const { |
| return true; |
| } |
| |
| int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| DCHECK_EQ(sample_rate_hz, 8000); |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcIlbcfix_Decode(dec_state_, encoded, |
| static_cast<int16_t>(encoded_len), decoded, |
| &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) { |
| return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames); |
| } |
| |
| int AudioDecoderIlbc::Init() { |
| return WebRtcIlbcfix_Decoderinit30Ms(dec_state_); |
| } |
| |
| size_t AudioDecoderIlbc::Channels() const { |
| return 1; |
| } |
| #endif |
| |
| // G.722 |
| #ifdef WEBRTC_CODEC_G722 |
| AudioDecoderG722::AudioDecoderG722() { |
| WebRtcG722_CreateDecoder(&dec_state_); |
| } |
| |
| AudioDecoderG722::~AudioDecoderG722() { |
| WebRtcG722_FreeDecoder(dec_state_); |
| } |
| |
| bool AudioDecoderG722::HasDecodePlc() const { |
| return false; |
| } |
| |
| int AudioDecoderG722::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| DCHECK_EQ(sample_rate_hz, 16000); |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = |
| WebRtcG722_Decode(dec_state_, encoded, static_cast<int16_t>(encoded_len), |
| decoded, &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderG722::Init() { |
| return WebRtcG722_DecoderInit(dec_state_); |
| } |
| |
| int AudioDecoderG722::PacketDuration(const uint8_t* encoded, |
| size_t encoded_len) const { |
| // 1/2 encoded byte per sample per channel. |
| return static_cast<int>(2 * encoded_len / Channels()); |
| } |
| |
| size_t AudioDecoderG722::Channels() const { |
| return 1; |
| } |
| |
| AudioDecoderG722Stereo::AudioDecoderG722Stereo() { |
| WebRtcG722_CreateDecoder(&dec_state_left_); |
| WebRtcG722_CreateDecoder(&dec_state_right_); |
| } |
| |
| AudioDecoderG722Stereo::~AudioDecoderG722Stereo() { |
| WebRtcG722_FreeDecoder(dec_state_left_); |
| WebRtcG722_FreeDecoder(dec_state_right_); |
| } |
| |
| int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| DCHECK_EQ(sample_rate_hz, 16000); |
| int16_t temp_type = 1; // Default is speech. |
| // De-interleave the bit-stream into two separate payloads. |
| uint8_t* encoded_deinterleaved = new uint8_t[encoded_len]; |
| SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved); |
| // Decode left and right. |
| int16_t ret = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved, |
| static_cast<int16_t>(encoded_len / 2), |
| decoded, &temp_type); |
| if (ret >= 0) { |
| int decoded_len = ret; |
| ret = WebRtcG722_Decode(dec_state_right_, |
| &encoded_deinterleaved[encoded_len / 2], |
| static_cast<int16_t>(encoded_len / 2), |
| &decoded[decoded_len], &temp_type); |
| if (ret == decoded_len) { |
| decoded_len += ret; |
| // Interleave output. |
| for (int k = decoded_len / 2; k < decoded_len; k++) { |
| int16_t temp = decoded[k]; |
| memmove(&decoded[2 * k - decoded_len + 2], |
| &decoded[2 * k - decoded_len + 1], |
| (decoded_len - k - 1) * sizeof(int16_t)); |
| decoded[2 * k - decoded_len + 1] = temp; |
| } |
| ret = decoded_len; // Return total number of samples. |
| } |
| } |
| *speech_type = ConvertSpeechType(temp_type); |
| delete [] encoded_deinterleaved; |
| return ret; |
| } |
| |
| size_t AudioDecoderG722Stereo::Channels() const { |
| return 2; |
| } |
| |
| int AudioDecoderG722Stereo::Init() { |
| int r = WebRtcG722_DecoderInit(dec_state_left_); |
| if (r != 0) |
| return r; |
| return WebRtcG722_DecoderInit(dec_state_right_); |
| } |
| |
| // Split the stereo packet and place left and right channel after each other |
| // in the output array. |
| void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded, |
| size_t encoded_len, |
| uint8_t* encoded_deinterleaved) { |
| assert(encoded); |
| // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ..., |
| // where "lx" is 4 bits representing left sample number x, and "rx" right |
| // sample. Two samples fit in one byte, represented with |...|. |
| for (size_t i = 0; i + 1 < encoded_len; i += 2) { |
| uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F); |
| encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4); |
| encoded_deinterleaved[i + 1] = right_byte; |
| } |
| |
| // Move one byte representing right channel each loop, and place it at the |
| // end of the bytestream vector. After looping the data is reordered to: |
| // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|, |
| // where N is the total number of samples. |
| for (size_t i = 0; i < encoded_len / 2; i++) { |
| uint8_t right_byte = encoded_deinterleaved[i + 1]; |
| memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2], |
| encoded_len - i - 2); |
| encoded_deinterleaved[encoded_len - 1] = right_byte; |
| } |
| } |
| #endif |
| |
| // Opus |
| #ifdef WEBRTC_CODEC_OPUS |
| AudioDecoderOpus::AudioDecoderOpus(int num_channels) : channels_(num_channels) { |
| DCHECK(num_channels == 1 || num_channels == 2); |
| WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_)); |
| } |
| |
| AudioDecoderOpus::~AudioDecoderOpus() { |
| WebRtcOpus_DecoderFree(dec_state_); |
| } |
| |
| int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| DCHECK_EQ(sample_rate_hz, 48000); |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcOpus_Decode(dec_state_, encoded, |
| static_cast<int16_t>(encoded_len), decoded, |
| &temp_type); |
| if (ret > 0) |
| ret *= static_cast<int16_t>(channels_); // Return total number of samples. |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| if (!PacketHasFec(encoded, encoded_len)) { |
| // This packet is a RED packet. |
| return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, |
| speech_type); |
| } |
| |
| DCHECK_EQ(sample_rate_hz, 48000); |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcOpus_DecodeFec(dec_state_, encoded, |
| static_cast<int16_t>(encoded_len), decoded, |
| &temp_type); |
| if (ret > 0) |
| ret *= static_cast<int16_t>(channels_); // Return total number of samples. |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderOpus::Init() { |
| return WebRtcOpus_DecoderInit(dec_state_); |
| } |
| |
| int AudioDecoderOpus::PacketDuration(const uint8_t* encoded, |
| size_t encoded_len) const { |
| return WebRtcOpus_DurationEst(dec_state_, |
| encoded, static_cast<int>(encoded_len)); |
| } |
| |
| int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded, |
| size_t encoded_len) const { |
| if (!PacketHasFec(encoded, encoded_len)) { |
| // This packet is a RED packet. |
| return PacketDuration(encoded, encoded_len); |
| } |
| |
| return WebRtcOpus_FecDurationEst(encoded, static_cast<int>(encoded_len)); |
| } |
| |
| bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded, |
| size_t encoded_len) const { |
| int fec; |
| fec = WebRtcOpus_PacketHasFec(encoded, static_cast<int>(encoded_len)); |
| return (fec == 1); |
| } |
| |
| size_t AudioDecoderOpus::Channels() const { |
| return channels_; |
| } |
| #endif |
| |
| AudioDecoderCng::AudioDecoderCng() { |
| CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_)); |
| } |
| |
| AudioDecoderCng::~AudioDecoderCng() { |
| WebRtcCng_FreeDec(dec_state_); |
| } |
| |
| int AudioDecoderCng::Init() { |
| return WebRtcCng_InitDec(dec_state_); |
| } |
| |
| int AudioDecoderCng::IncomingPacket(const uint8_t* payload, |
| size_t payload_len, |
| uint16_t rtp_sequence_number, |
| uint32_t rtp_timestamp, |
| uint32_t arrival_timestamp) { |
| return -1; |
| } |
| |
| CNG_dec_inst* AudioDecoderCng::CngDecoderInstance() { |
| return dec_state_; |
| } |
| |
| size_t AudioDecoderCng::Channels() const { |
| return 1; |
| } |
| |
| int AudioDecoderCng::DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) { |
| return -1; |
| } |
| |
| bool CodecSupported(NetEqDecoder codec_type) { |
| switch (codec_type) { |
| case kDecoderPCMu: |
| case kDecoderPCMa: |
| case kDecoderPCMu_2ch: |
| case kDecoderPCMa_2ch: |
| #ifdef WEBRTC_CODEC_ILBC |
| case kDecoderILBC: |
| #endif |
| #if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC) |
| case kDecoderISAC: |
| #endif |
| #ifdef WEBRTC_CODEC_ISAC |
| case kDecoderISACswb: |
| case kDecoderISACfb: |
| #endif |
| #ifdef WEBRTC_CODEC_PCM16 |
| case kDecoderPCM16B: |
| case kDecoderPCM16Bwb: |
| case kDecoderPCM16Bswb32kHz: |
| case kDecoderPCM16Bswb48kHz: |
| case kDecoderPCM16B_2ch: |
| case kDecoderPCM16Bwb_2ch: |
| case kDecoderPCM16Bswb32kHz_2ch: |
| case kDecoderPCM16Bswb48kHz_2ch: |
| case kDecoderPCM16B_5ch: |
| #endif |
| #ifdef WEBRTC_CODEC_G722 |
| case kDecoderG722: |
| case kDecoderG722_2ch: |
| #endif |
| #ifdef WEBRTC_CODEC_OPUS |
| case kDecoderOpus: |
| case kDecoderOpus_2ch: |
| #endif |
| case kDecoderRED: |
| case kDecoderAVT: |
| case kDecoderCNGnb: |
| case kDecoderCNGwb: |
| case kDecoderCNGswb32kHz: |
| case kDecoderCNGswb48kHz: |
| case kDecoderArbitrary: { |
| return true; |
| } |
| default: { |
| return false; |
| } |
| } |
| } |
| |
| int CodecSampleRateHz(NetEqDecoder codec_type) { |
| switch (codec_type) { |
| case kDecoderPCMu: |
| case kDecoderPCMa: |
| case kDecoderPCMu_2ch: |
| case kDecoderPCMa_2ch: |
| #ifdef WEBRTC_CODEC_ILBC |
| case kDecoderILBC: |
| #endif |
| #ifdef WEBRTC_CODEC_PCM16 |
| case kDecoderPCM16B: |
| case kDecoderPCM16B_2ch: |
| case kDecoderPCM16B_5ch: |
| #endif |
| case kDecoderCNGnb: { |
| return 8000; |
| } |
| #if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC) |
| case kDecoderISAC: |
| #endif |
| #ifdef WEBRTC_CODEC_PCM16 |
| case kDecoderPCM16Bwb: |
| case kDecoderPCM16Bwb_2ch: |
| #endif |
| #ifdef WEBRTC_CODEC_G722 |
| case kDecoderG722: |
| case kDecoderG722_2ch: |
| #endif |
| case kDecoderCNGwb: { |
| return 16000; |
| } |
| #ifdef WEBRTC_CODEC_ISAC |
| case kDecoderISACswb: |
| case kDecoderISACfb: |
| #endif |
| #ifdef WEBRTC_CODEC_PCM16 |
| case kDecoderPCM16Bswb32kHz: |
| case kDecoderPCM16Bswb32kHz_2ch: |
| #endif |
| case kDecoderCNGswb32kHz: { |
| return 32000; |
| } |
| #ifdef WEBRTC_CODEC_PCM16 |
| case kDecoderPCM16Bswb48kHz: |
| case kDecoderPCM16Bswb48kHz_2ch: { |
| return 48000; |
| } |
| #endif |
| #ifdef WEBRTC_CODEC_OPUS |
| case kDecoderOpus: |
| case kDecoderOpus_2ch: { |
| return 48000; |
| } |
| #endif |
| case kDecoderCNGswb48kHz: { |
| // TODO(tlegrand): Remove limitation once ACM has full 48 kHz support. |
| return 32000; |
| } |
| default: { |
| return -1; // Undefined sample rate. |
| } |
| } |
| } |
| |
| AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type) { |
| if (!CodecSupported(codec_type)) { |
| return NULL; |
| } |
| switch (codec_type) { |
| case kDecoderPCMu: |
| return new AudioDecoderPcmU; |
| case kDecoderPCMa: |
| return new AudioDecoderPcmA; |
| case kDecoderPCMu_2ch: |
| return new AudioDecoderPcmUMultiCh(2); |
| case kDecoderPCMa_2ch: |
| return new AudioDecoderPcmAMultiCh(2); |
| #ifdef WEBRTC_CODEC_ILBC |
| case kDecoderILBC: |
| return new AudioDecoderIlbc; |
| #endif |
| #if defined(WEBRTC_CODEC_ISACFX) |
| case kDecoderISAC: { |
| AudioEncoderDecoderIsacFix::Config config; |
| return new AudioEncoderDecoderIsacFix(config); |
| } |
| #elif defined(WEBRTC_CODEC_ISAC) |
| case kDecoderISAC: { |
| AudioEncoderDecoderIsac::Config config; |
| config.sample_rate_hz = 16000; |
| return new AudioEncoderDecoderIsac(config); |
| } |
| case kDecoderISACswb: |
| case kDecoderISACfb: { |
| AudioEncoderDecoderIsac::Config config; |
| config.sample_rate_hz = 32000; |
| return new AudioEncoderDecoderIsac(config); |
| } |
| #endif |
| #ifdef WEBRTC_CODEC_PCM16 |
| case kDecoderPCM16B: |
| case kDecoderPCM16Bwb: |
| case kDecoderPCM16Bswb32kHz: |
| case kDecoderPCM16Bswb48kHz: |
| return new AudioDecoderPcm16B; |
| case kDecoderPCM16B_2ch: |
| case kDecoderPCM16Bwb_2ch: |
| case kDecoderPCM16Bswb32kHz_2ch: |
| case kDecoderPCM16Bswb48kHz_2ch: |
| return new AudioDecoderPcm16BMultiCh(2); |
| case kDecoderPCM16B_5ch: |
| return new AudioDecoderPcm16BMultiCh(5); |
| #endif |
| #ifdef WEBRTC_CODEC_G722 |
| case kDecoderG722: |
| return new AudioDecoderG722; |
| case kDecoderG722_2ch: |
| return new AudioDecoderG722Stereo; |
| #endif |
| #ifdef WEBRTC_CODEC_OPUS |
| case kDecoderOpus: |
| return new AudioDecoderOpus(1); |
| case kDecoderOpus_2ch: |
| return new AudioDecoderOpus(2); |
| #endif |
| case kDecoderCNGnb: |
| case kDecoderCNGwb: |
| case kDecoderCNGswb32kHz: |
| case kDecoderCNGswb48kHz: |
| return new AudioDecoderCng; |
| case kDecoderRED: |
| case kDecoderAVT: |
| case kDecoderArbitrary: |
| default: { |
| return NULL; |
| } |
| } |
| } |
| |
| } // namespace webrtc |
