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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/main/acm2/acm_generic_codec.h"
#include <assert.h>
#include <string.h>
#include <algorithm>
#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h"
#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
namespace {
static const int kInvalidPayloadType = 255;
std::map<int, int>::iterator FindSampleRateInMap(std::map<int, int>* cng_pt_map,
int sample_rate_hz) {
return find_if(cng_pt_map->begin(), cng_pt_map->end(),
[sample_rate_hz](decltype(*cng_pt_map->begin()) p) {
return p.second == sample_rate_hz;
});
}
void SetPtInMap(std::map<int, int>* pt_map,
int sample_rate_hz,
int payload_type) {
if (payload_type == kInvalidPayloadType)
return;
CHECK_GE(payload_type, 0);
CHECK_LT(payload_type, 128);
auto pt_iter = FindSampleRateInMap(pt_map, sample_rate_hz);
if (pt_iter != pt_map->end()) {
// Remove item in map with sample_rate_hz.
pt_map->erase(pt_iter);
}
(*pt_map)[payload_type] = sample_rate_hz;
}
} // namespace
namespace acm2 {
// Enum for CNG
enum {
kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER,
kNewCNGNumLPCParams = 8
};
// Interval for sending new CNG parameters (SID frames) is 100 msec.
enum {
kCngSidIntervalMsec = 100
};
// We set some of the variables to invalid values as a check point
// if a proper initialization has happened. Another approach is
// to initialize to a default codec that we are sure is always included.
ACMGenericCodec::ACMGenericCodec(const CodecInst& codec_inst,
int cng_pt_nb,
int cng_pt_wb,
int cng_pt_swb,
int cng_pt_fb,
bool enable_red,
int red_pt_nb)
: has_internal_fec_(false),
copy_red_enabled_(enable_red),
encoder_(NULL),
bitrate_bps_(0),
fec_enabled_(false),
loss_rate_(0),
max_playback_rate_hz_(48000),
max_payload_size_bytes_(-1),
max_rate_bps_(-1),
opus_dtx_enabled_(false),
is_opus_(false),
is_isac_(false),
opus_application_set_(false) {
acm_codec_params_.codec_inst = codec_inst;
acm_codec_params_.enable_dtx = false;
acm_codec_params_.enable_vad = false;
acm_codec_params_.vad_mode = VADNormal;
SetPtInMap(&red_pt_, 8000, red_pt_nb);
SetPtInMap(&cng_pt_, 8000, cng_pt_nb);
SetPtInMap(&cng_pt_, 16000, cng_pt_wb);
SetPtInMap(&cng_pt_, 32000, cng_pt_swb);
SetPtInMap(&cng_pt_, 48000, cng_pt_fb);
ResetAudioEncoder();
CHECK(encoder_);
}
ACMGenericCodec::~ACMGenericCodec() {
}
AudioDecoderProxy::AudioDecoderProxy()
: decoder_lock_(CriticalSectionWrapper::CreateCriticalSection()),
decoder_(nullptr) {
}
AudioDecoderProxy::~AudioDecoderProxy() = default;
void AudioDecoderProxy::SetDecoder(AudioDecoder* decoder) {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
decoder_ = decoder;
CHECK_EQ(decoder_->Init(), 0);
}
bool AudioDecoderProxy::IsSet() const {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return (decoder_ != nullptr);
}
int AudioDecoderProxy::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type) {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->Decode(encoded, encoded_len, sample_rate_hz,
max_decoded_bytes, decoded, speech_type);
}
int AudioDecoderProxy::DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type) {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->DecodeRedundant(encoded, encoded_len, sample_rate_hz,
max_decoded_bytes, decoded, speech_type);
}
bool AudioDecoderProxy::HasDecodePlc() const {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->HasDecodePlc();
}
int AudioDecoderProxy::DecodePlc(int num_frames, int16_t* decoded) {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->DecodePlc(num_frames, decoded);
}
int AudioDecoderProxy::Init() {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->Init();
}
int AudioDecoderProxy::IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->IncomingPacket(payload, payload_len, rtp_sequence_number,
rtp_timestamp, arrival_timestamp);
}
int AudioDecoderProxy::ErrorCode() {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->ErrorCode();
}
int AudioDecoderProxy::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->PacketDuration(encoded, encoded_len);
}
int AudioDecoderProxy::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->PacketDurationRedundant(encoded, encoded_len);
}
bool AudioDecoderProxy::PacketHasFec(const uint8_t* encoded,
size_t encoded_len) const {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->PacketHasFec(encoded, encoded_len);
}
CNG_dec_inst* AudioDecoderProxy::CngDecoderInstance() {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->CngDecoderInstance();
}
size_t AudioDecoderProxy::Channels() const {
CriticalSectionScoped decoder_lock(decoder_lock_.get());
return decoder_->Channels();
}
int16_t ACMGenericCodec::EncoderParams(WebRtcACMCodecParams* enc_params) const {
*enc_params = acm_codec_params_;
return 0;
}
int16_t ACMGenericCodec::InitEncoder(WebRtcACMCodecParams* codec_params,
bool force_initialization) {
bitrate_bps_ = 0;
loss_rate_ = 0;
opus_dtx_enabled_ = false;
acm_codec_params_ = *codec_params;
if (force_initialization)
opus_application_set_ = false;
opus_application_ = GetOpusApplication(codec_params->codec_inst.channels,
opus_dtx_enabled_);
opus_application_set_ = true;
ResetAudioEncoder();
return 0;
}
void ACMGenericCodec::ResetAudioEncoder() {
const CodecInst& codec_inst = acm_codec_params_.codec_inst;
if (!STR_CASE_CMP(codec_inst.plname, "PCMU")) {
AudioEncoderPcmU::Config config;
config.num_channels = codec_inst.channels;
config.frame_size_ms = codec_inst.pacsize / 8;
config.payload_type = codec_inst.pltype;
audio_encoder_.reset(new AudioEncoderPcmU(config));
} else if (!STR_CASE_CMP(codec_inst.plname, "PCMA")) {
AudioEncoderPcmA::Config config;
config.num_channels = codec_inst.channels;
config.frame_size_ms = codec_inst.pacsize / 8;
config.payload_type = codec_inst.pltype;
audio_encoder_.reset(new AudioEncoderPcmA(config));
#ifdef WEBRTC_CODEC_PCM16
} else if (!STR_CASE_CMP(codec_inst.plname, "L16")) {
AudioEncoderPcm16B::Config config;
config.num_channels = codec_inst.channels;
config.sample_rate_hz = codec_inst.plfreq;
config.frame_size_ms = codec_inst.pacsize / (config.sample_rate_hz / 1000);
config.payload_type = codec_inst.pltype;
audio_encoder_.reset(new AudioEncoderPcm16B(config));
#endif
#ifdef WEBRTC_CODEC_ILBC
} else if (!STR_CASE_CMP(codec_inst.plname, "ILBC")) {
AudioEncoderIlbc::Config config;
config.frame_size_ms = codec_inst.pacsize / 8;
config.payload_type = codec_inst.pltype;
audio_encoder_.reset(new AudioEncoderIlbc(config));
#endif
#ifdef WEBRTC_CODEC_OPUS
} else if (!STR_CASE_CMP(codec_inst.plname, "opus")) {
is_opus_ = true;
has_internal_fec_ = true;
AudioEncoderOpus::Config config;
config.frame_size_ms = codec_inst.pacsize / 48;
config.num_channels = codec_inst.channels;
config.fec_enabled = fec_enabled_;
config.bitrate_bps = codec_inst.rate;
config.max_playback_rate_hz = max_playback_rate_hz_;
config.dtx_enabled = opus_dtx_enabled_;
config.payload_type = codec_inst.pltype;
switch (GetOpusApplication(config.num_channels, config.dtx_enabled)) {
case kVoip:
config.application = AudioEncoderOpus::ApplicationMode::kVoip;
break;
case kAudio:
config.application = AudioEncoderOpus::ApplicationMode::kAudio;
break;
}
audio_encoder_.reset(new AudioEncoderOpus(config));
#endif
#ifdef WEBRTC_CODEC_G722
} else if (!STR_CASE_CMP(codec_inst.plname, "G722")) {
AudioEncoderG722::Config config;
config.num_channels = codec_inst.channels;
config.frame_size_ms = codec_inst.pacsize / 16;
config.payload_type = codec_inst.pltype;
audio_encoder_.reset(new AudioEncoderG722(config));
#endif
#ifdef WEBRTC_CODEC_ISACFX
} else if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
DCHECK_EQ(codec_inst.plfreq, 16000);
is_isac_ = true;
AudioEncoderDecoderIsacFix* enc_dec;
if (codec_inst.rate == -1) {
// Adaptive mode.
AudioEncoderDecoderIsacFix::ConfigAdaptive config;
config.payload_type = codec_inst.pltype;
enc_dec = new AudioEncoderDecoderIsacFix(config);
} else {
// Channel independent mode.
AudioEncoderDecoderIsacFix::Config config;
config.bit_rate = codec_inst.rate;
config.frame_size_ms = codec_inst.pacsize / 16;
config.payload_type = codec_inst.pltype;
enc_dec = new AudioEncoderDecoderIsacFix(config);
}
decoder_proxy_.SetDecoder(enc_dec);
audio_encoder_.reset(enc_dec);
#endif
#ifdef WEBRTC_CODEC_ISAC
} else if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
is_isac_ = true;
AudioEncoderDecoderIsac* enc_dec;
if (codec_inst.rate == -1) {
// Adaptive mode.
AudioEncoderDecoderIsac::ConfigAdaptive config;
config.sample_rate_hz = codec_inst.plfreq;
config.initial_frame_size_ms = rtc::CheckedDivExact(
1000 * codec_inst.pacsize, config.sample_rate_hz);
config.max_payload_size_bytes = max_payload_size_bytes_;
config.max_bit_rate = max_rate_bps_;
config.payload_type = codec_inst.pltype;
enc_dec = new AudioEncoderDecoderIsac(config);
} else {
// Channel independent mode.
AudioEncoderDecoderIsac::Config config;
config.sample_rate_hz = codec_inst.plfreq;
config.bit_rate = codec_inst.rate;
config.frame_size_ms = rtc::CheckedDivExact(1000 * codec_inst.pacsize,
config.sample_rate_hz);
config.max_payload_size_bytes = max_payload_size_bytes_;
config.max_bit_rate = max_rate_bps_;
config.payload_type = codec_inst.pltype;
enc_dec = new AudioEncoderDecoderIsac(config);
}
decoder_proxy_.SetDecoder(enc_dec);
audio_encoder_.reset(enc_dec);
#endif
} else {
FATAL();
}
if (bitrate_bps_ != 0)
audio_encoder_->SetTargetBitrate(bitrate_bps_);
audio_encoder_->SetProjectedPacketLossRate(loss_rate_ / 100.0);
encoder_ = audio_encoder_.get();
// Attach RED if needed.
auto pt_iter =
FindSampleRateInMap(&red_pt_, audio_encoder_->SampleRateHz());
if (copy_red_enabled_ && pt_iter != red_pt_.end()) {
CHECK_NE(pt_iter->first, kInvalidPayloadType);
AudioEncoderCopyRed::Config config;
config.payload_type = pt_iter->first;
config.speech_encoder = encoder_;
red_encoder_.reset(new AudioEncoderCopyRed(config));
encoder_ = red_encoder_.get();
} else {
red_encoder_.reset();
copy_red_enabled_ = false;
}
// Attach CNG if needed.
// Reverse-lookup from sample rate to complete key-value pair.
pt_iter =
FindSampleRateInMap(&cng_pt_, audio_encoder_->SampleRateHz());
if (acm_codec_params_.enable_dtx && pt_iter != cng_pt_.end()) {
AudioEncoderCng::Config config;
config.num_channels = acm_codec_params_.codec_inst.channels;
config.payload_type = pt_iter->first;
config.speech_encoder = encoder_;
switch (acm_codec_params_.vad_mode) {
case VADNormal:
config.vad_mode = Vad::kVadNormal;
break;
case VADLowBitrate:
config.vad_mode = Vad::kVadLowBitrate;
break;
case VADAggr:
config.vad_mode = Vad::kVadAggressive;
break;
case VADVeryAggr:
config.vad_mode = Vad::kVadVeryAggressive;
break;
default:
FATAL();
}
cng_encoder_.reset(new AudioEncoderCng(config));
encoder_ = cng_encoder_.get();
} else {
cng_encoder_.reset();
}
}
OpusApplicationMode ACMGenericCodec::GetOpusApplication(
int num_channels, bool enable_dtx) const {
if (opus_application_set_)
return opus_application_;
return num_channels == 1 || enable_dtx ? kVoip : kAudio;
}
int16_t ACMGenericCodec::SetBitRate(const int32_t bitrate_bps) {
encoder_->SetTargetBitrate(bitrate_bps);
bitrate_bps_ = bitrate_bps;
return 0;
}
int16_t ACMGenericCodec::SetVAD(bool* enable_dtx,
bool* enable_vad,
ACMVADMode* mode) {
if (is_opus_) {
*enable_dtx = false;
*enable_vad = false;
return 0;
}
// Note: |enable_vad| is not used; VAD is enabled based on the DTX setting and
// the |enable_vad| is set equal to |enable_dtx|.
// The case when VAD is enabled but DTX is disabled may result in a
// kPassiveNormalEncoded frame type, but this is not a case that VoE
// distinguishes from the cases where DTX is in fact used. In the case where
// DTX is enabled but VAD is disabled, the comment in the ACM interface states
// that VAD will be enabled anyway.
DCHECK_EQ(*enable_dtx, *enable_vad);
*enable_vad = *enable_dtx;
acm_codec_params_.enable_dtx = *enable_dtx;
acm_codec_params_.enable_vad = *enable_vad;
acm_codec_params_.vad_mode = *mode;
if (acm_codec_params_.enable_dtx && !cng_encoder_) {
ResetAudioEncoder();
} else if (!acm_codec_params_.enable_dtx && cng_encoder_) {
cng_encoder_.reset();
encoder_ = audio_encoder_.get();
}
return 0;
}
void ACMGenericCodec::SetCngPt(int sample_rate_hz, int payload_type) {
SetPtInMap(&cng_pt_, sample_rate_hz, payload_type);
ResetAudioEncoder();
}
void ACMGenericCodec::SetRedPt(int sample_rate_hz, int payload_type) {
SetPtInMap(&red_pt_, sample_rate_hz, payload_type);
ResetAudioEncoder();
}
int32_t ACMGenericCodec::SetISACMaxPayloadSize(
const uint16_t max_payload_len_bytes) {
if (!is_isac_)
return -1; // Needed for tests to pass.
max_payload_size_bytes_ = max_payload_len_bytes;
ResetAudioEncoder();
return 0;
}
int32_t ACMGenericCodec::SetISACMaxRate(const uint32_t max_rate_bps) {
if (!is_isac_)
return -1; // Needed for tests to pass.
max_rate_bps_ = max_rate_bps;
ResetAudioEncoder();
return 0;
}
int ACMGenericCodec::SetOpusMaxPlaybackRate(int frequency_hz) {
if (!is_opus_)
return -1; // Needed for tests to pass.
max_playback_rate_hz_ = frequency_hz;
ResetAudioEncoder();
return 0;
}
AudioDecoder* ACMGenericCodec::Decoder() {
return decoder_proxy_.IsSet() ? &decoder_proxy_ : nullptr;
}
int ACMGenericCodec::EnableOpusDtx(bool force_voip) {
if (!is_opus_)
return -1; // Needed for tests to pass.
if (!force_voip &&
GetOpusApplication(encoder_->NumChannels(), true) != kVoip) {
// Opus DTX can only be enabled when application mode is KVoip.
return -1;
}
opus_application_ = kVoip;
opus_application_set_ = true;
opus_dtx_enabled_ = true;
ResetAudioEncoder();
return 0;
}
int ACMGenericCodec::DisableOpusDtx() {
if (!is_opus_)
return -1; // Needed for tests to pass.
opus_dtx_enabled_ = false;
ResetAudioEncoder();
return 0;
}
int ACMGenericCodec::SetFEC(bool enable_fec) {
if (!HasInternalFEC())
return enable_fec ? -1 : 0;
if (fec_enabled_ != enable_fec) {
fec_enabled_ = enable_fec;
ResetAudioEncoder();
}
return 0;
}
int ACMGenericCodec::SetOpusApplication(OpusApplicationMode application,
bool disable_dtx_if_needed) {
if (opus_dtx_enabled_ && application == kAudio) {
if (disable_dtx_if_needed) {
opus_dtx_enabled_ = false;
} else {
// Opus can only be set to kAudio when DTX is off.
return -1;
}
}
opus_application_ = application;
opus_application_set_ = true;
ResetAudioEncoder();
return 0;
}
int ACMGenericCodec::SetPacketLossRate(int loss_rate) {
encoder_->SetProjectedPacketLossRate(loss_rate / 100.0);
loss_rate_ = loss_rate;
return 0;
}
int ACMGenericCodec::SetCopyRed(bool enable) {
copy_red_enabled_ = enable;
ResetAudioEncoder();
return copy_red_enabled_ == enable ? 0 : -1;
}
AudioEncoder* ACMGenericCodec::GetAudioEncoder() {
return encoder_;
}
const AudioEncoder* ACMGenericCodec::GetAudioEncoder() const {
return encoder_;
}
} // namespace acm2
} // namespace webrtc