| /* |
| * 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" |
| #ifdef WEBRTC_CODEC_CELT |
| #include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h" |
| #endif |
| #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/isacfix.h" |
| #endif |
| #ifdef WEBRTC_CODEC_ISAC |
| #include "webrtc/modules/audio_coding/codecs/isac/main/interface/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::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcG711_DecodeU( |
| reinterpret_cast<int16_t*>(const_cast<uint8_t*>(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) { |
| // One encoded byte per sample per channel. |
| return static_cast<int>(encoded_len / channels_); |
| } |
| |
| // PCMa |
| int AudioDecoderPcmA::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcG711_DecodeA( |
| reinterpret_cast<int16_t*>(const_cast<uint8_t*>(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) { |
| // One encoded byte per sample per channel. |
| return static_cast<int>(encoded_len / channels_); |
| } |
| |
| // PCM16B |
| #ifdef WEBRTC_CODEC_PCM16 |
| AudioDecoderPcm16B::AudioDecoderPcm16B() {} |
| |
| int AudioDecoderPcm16B::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcPcm16b_DecodeW16( |
| reinterpret_cast<int16_t*>(const_cast<uint8_t*>(encoded)), |
| static_cast<int16_t>(encoded_len), decoded, &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded, |
| size_t encoded_len) { |
| // Two encoded byte per sample per channel. |
| return static_cast<int>(encoded_len / (2 * channels_)); |
| } |
| |
| AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels) { |
| DCHECK(num_channels > 0); |
| channels_ = num_channels; |
| } |
| #endif |
| |
| // iLBC |
| #ifdef WEBRTC_CODEC_ILBC |
| AudioDecoderIlbc::AudioDecoderIlbc() { |
| WebRtcIlbcfix_DecoderCreate(&dec_state_); |
| } |
| |
| AudioDecoderIlbc::~AudioDecoderIlbc() { |
| WebRtcIlbcfix_DecoderFree(dec_state_); |
| } |
| |
| int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcIlbcfix_Decode(dec_state_, |
| reinterpret_cast<const int16_t*>(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_); |
| } |
| #endif |
| |
| // iSAC float |
| #ifdef WEBRTC_CODEC_ISAC |
| AudioDecoderIsac::AudioDecoderIsac(int decode_sample_rate_hz) { |
| DCHECK(decode_sample_rate_hz == 16000 || decode_sample_rate_hz == 32000); |
| WebRtcIsac_Create(&isac_state_); |
| WebRtcIsac_SetDecSampRate(isac_state_, decode_sample_rate_hz); |
| } |
| |
| AudioDecoderIsac::~AudioDecoderIsac() { |
| WebRtcIsac_Free(isac_state_); |
| } |
| |
| int AudioDecoderIsac::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcIsac_Decode(isac_state_, |
| encoded, |
| static_cast<int16_t>(encoded_len), decoded, |
| &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderIsac::DecodeRedundant(const uint8_t* encoded, |
| size_t encoded_len, int16_t* decoded, |
| SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcIsac_DecodeRcu(isac_state_, |
| encoded, |
| static_cast<int16_t>(encoded_len), decoded, |
| &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderIsac::DecodePlc(int num_frames, int16_t* decoded) { |
| return WebRtcIsac_DecodePlc(isac_state_, decoded, num_frames); |
| } |
| |
| int AudioDecoderIsac::Init() { |
| return WebRtcIsac_DecoderInit(isac_state_); |
| } |
| |
| int AudioDecoderIsac::IncomingPacket(const uint8_t* payload, |
| size_t payload_len, |
| uint16_t rtp_sequence_number, |
| uint32_t rtp_timestamp, |
| uint32_t arrival_timestamp) { |
| return WebRtcIsac_UpdateBwEstimate(isac_state_, |
| payload, |
| static_cast<int32_t>(payload_len), |
| rtp_sequence_number, |
| rtp_timestamp, |
| arrival_timestamp); |
| } |
| |
| int AudioDecoderIsac::ErrorCode() { |
| return WebRtcIsac_GetErrorCode(isac_state_); |
| } |
| #endif |
| |
| // iSAC fix |
| #ifdef WEBRTC_CODEC_ISACFX |
| AudioDecoderIsacFix::AudioDecoderIsacFix() { |
| WebRtcIsacfix_Create(&isac_state_); |
| } |
| |
| AudioDecoderIsacFix::~AudioDecoderIsacFix() { |
| WebRtcIsacfix_Free(isac_state_); |
| } |
| |
| int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcIsacfix_Decode(isac_state_, |
| encoded, |
| static_cast<int16_t>(encoded_len), decoded, |
| &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| return ret; |
| } |
| |
| int AudioDecoderIsacFix::Init() { |
| return WebRtcIsacfix_DecoderInit(isac_state_); |
| } |
| |
| int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload, |
| size_t payload_len, |
| uint16_t rtp_sequence_number, |
| uint32_t rtp_timestamp, |
| uint32_t arrival_timestamp) { |
| return WebRtcIsacfix_UpdateBwEstimate( |
| isac_state_, |
| payload, |
| static_cast<int32_t>(payload_len), |
| rtp_sequence_number, rtp_timestamp, arrival_timestamp); |
| } |
| |
| int AudioDecoderIsacFix::ErrorCode() { |
| return WebRtcIsacfix_GetErrorCode(isac_state_); |
| } |
| #endif |
| |
| // G.722 |
| #ifdef WEBRTC_CODEC_G722 |
| AudioDecoderG722::AudioDecoderG722() { |
| WebRtcG722_CreateDecoder(&dec_state_); |
| } |
| |
| AudioDecoderG722::~AudioDecoderG722() { |
| WebRtcG722_FreeDecoder(dec_state_); |
| } |
| |
| int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcG722_Decode( |
| dec_state_, |
| const_cast<int16_t*>(reinterpret_cast<const int16_t*>(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) { |
| // 1/2 encoded byte per sample per channel. |
| return static_cast<int>(2 * encoded_len / channels_); |
| } |
| |
| AudioDecoderG722Stereo::AudioDecoderG722Stereo() { |
| channels_ = 2; |
| WebRtcG722_CreateDecoder(&dec_state_left_); |
| WebRtcG722_CreateDecoder(&dec_state_right_); |
| } |
| |
| AudioDecoderG722Stereo::~AudioDecoderG722Stereo() { |
| WebRtcG722_FreeDecoder(dec_state_left_); |
| WebRtcG722_FreeDecoder(dec_state_right_); |
| } |
| |
| int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| 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_, |
| reinterpret_cast<int16_t*>(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_, |
| reinterpret_cast<int16_t*>(&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; |
| } |
| |
| 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 |
| |
| // CELT |
| #ifdef WEBRTC_CODEC_CELT |
| AudioDecoderCelt::AudioDecoderCelt(int num_channels) { |
| DCHECK(num_channels == 1 || num_channels == 2); |
| channels_ = num_channels; |
| WebRtcCelt_CreateDec(reinterpret_cast<CELT_decinst_t**>(&state_), |
| static_cast<int>(channels_)); |
| } |
| |
| AudioDecoderCelt::~AudioDecoderCelt() { |
| WebRtcCelt_FreeDec(static_cast<CELT_decinst_t*>(state_)); |
| } |
| |
| int AudioDecoderCelt::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default to speech. |
| int ret = WebRtcCelt_DecodeUniversal(static_cast<CELT_decinst_t*>(state_), |
| encoded, static_cast<int>(encoded_len), |
| decoded, &temp_type); |
| *speech_type = ConvertSpeechType(temp_type); |
| if (ret < 0) { |
| return -1; |
| } |
| // Return the total number of samples. |
| return ret * static_cast<int>(channels_); |
| } |
| |
| int AudioDecoderCelt::Init() { |
| return WebRtcCelt_DecoderInit(static_cast<CELT_decinst_t*>(state_)); |
| } |
| |
| bool AudioDecoderCelt::HasDecodePlc() const { return true; } |
| |
| int AudioDecoderCelt::DecodePlc(int num_frames, int16_t* decoded) { |
| int ret = WebRtcCelt_DecodePlc(static_cast<CELT_decinst_t*>(state_), |
| decoded, num_frames); |
| if (ret < 0) { |
| return -1; |
| } |
| // Return the total number of samples. |
| return ret * static_cast<int>(channels_); |
| } |
| #endif |
| |
| // Opus |
| #ifdef WEBRTC_CODEC_OPUS |
| AudioDecoderOpus::AudioDecoderOpus(int num_channels) { |
| DCHECK(num_channels == 1 || num_channels == 2); |
| channels_ = num_channels; |
| WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_)); |
| } |
| |
| AudioDecoderOpus::~AudioDecoderOpus() { |
| WebRtcOpus_DecoderFree(dec_state_); |
| } |
| |
| int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len, |
| int16_t* decoded, SpeechType* speech_type) { |
| int16_t temp_type = 1; // Default is speech. |
| int16_t ret = WebRtcOpus_DecodeNew(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::DecodeRedundant(const uint8_t* encoded, |
| size_t encoded_len, int16_t* decoded, |
| SpeechType* speech_type) { |
| 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_DecoderInitNew(dec_state_); |
| } |
| |
| int AudioDecoderOpus::PacketDuration(const uint8_t* encoded, |
| size_t encoded_len) { |
| return WebRtcOpus_DurationEst(dec_state_, |
| encoded, static_cast<int>(encoded_len)); |
| } |
| |
| int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded, |
| size_t encoded_len) const { |
| 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); |
| } |
| #endif |
| |
| AudioDecoderCng::AudioDecoderCng() { |
| CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_)); |
| } |
| |
| AudioDecoderCng::~AudioDecoderCng() { |
| WebRtcCng_FreeDec(dec_state_); |
| } |
| |
| int AudioDecoderCng::Init() { |
| return WebRtcCng_InitDec(dec_state_); |
| } |
| |
| } // namespace webrtc |