webrtc_m130/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc

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/*
* 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 "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/audio_decoder_g722.h"
#endif
#ifdef WEBRTC_CODEC_ILBC
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_decoder_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/audio_decoder_opus.h"
#endif
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_decoder_pcm16b.h"
namespace webrtc {
// PCMu
void AudioDecoderPcmU::Reset() {
}
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) {
RTC_DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
return static_cast<int>(ret);
}
int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// One encoded byte per sample per channel.
return static_cast<int>(encoded_len / Channels());
}
size_t AudioDecoderPcmUMultiCh::Channels() const {
return channels_;
}
// PCMa
void AudioDecoderPcmA::Reset() {
}
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) {
RTC_DCHECK_EQ(sample_rate_hz, 8000);
int16_t temp_type = 1; // Default is speech.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
return static_cast<int>(ret);
}
int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
// One encoded byte per sample per channel.
return static_cast<int>(encoded_len / Channels());
}
size_t AudioDecoderPcmAMultiCh::Channels() const {
return channels_;
}
AudioDecoderCng::AudioDecoderCng() {
RTC_CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
WebRtcCng_InitDec(dec_state_);
}
AudioDecoderCng::~AudioDecoderCng() {
WebRtcCng_FreeDec(dec_state_);
}
void AudioDecoderCng::Reset() {
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
case kDecoderPCM16B:
case kDecoderPCM16Bwb:
case kDecoderPCM16Bswb32kHz:
case kDecoderPCM16Bswb48kHz:
case kDecoderPCM16B_2ch:
case kDecoderPCM16Bwb_2ch:
case kDecoderPCM16Bswb32kHz_2ch:
case kDecoderPCM16Bswb48kHz_2ch:
case kDecoderPCM16B_5ch:
#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
case kDecoderPCM16B:
case kDecoderPCM16B_2ch:
case kDecoderPCM16B_5ch:
case kDecoderCNGnb: {
return 8000;
}
#if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
case kDecoderISAC:
#endif
case kDecoderPCM16Bwb:
case kDecoderPCM16Bwb_2ch:
#ifdef WEBRTC_CODEC_G722
case kDecoderG722:
case kDecoderG722_2ch:
#endif
case kDecoderCNGwb: {
return 16000;
}
#ifdef WEBRTC_CODEC_ISAC
case kDecoderISACswb:
case kDecoderISACfb:
#endif
case kDecoderPCM16Bswb32kHz:
case kDecoderPCM16Bswb32kHz_2ch:
case kDecoderCNGswb32kHz: {
return 32000;
}
case kDecoderPCM16Bswb48kHz:
case kDecoderPCM16Bswb48kHz_2ch: {
return 48000;
}
#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:
return new AudioDecoderIsacFix();
#elif defined(WEBRTC_CODEC_ISAC)
case kDecoderISAC:
case kDecoderISACswb:
case kDecoderISACfb:
return new AudioDecoderIsac();
#endif
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);
#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