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webrtc / src / webrtc / 0ab923a94a7f33e3242ac549c4bf16419c1994bc / . / modules / audio_coding / main / acm2 / acm_receiver.cc
blob: bbe5a16008de1807c55912c8ee88253ebab28960 [file] [log] [blame]
/*
* Copyright (c) 2013 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_receiver.h"
#include <stdlib.h> // malloc
#include <algorithm> // sort
#include <vector>
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/main/acm2/nack.h"
#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
namespace acm2 {
namespace {
const int kNackThresholdPackets = 2;
// |vad_activity_| field of |audio_frame| is set to |previous_audio_activity_|
// before the call to this function.
void SetAudioFrameActivityAndType(bool vad_enabled,
NetEqOutputType type,
AudioFrame* audio_frame) {
if (vad_enabled) {
switch (type) {
case kOutputNormal: {
audio_frame->vad_activity_ = AudioFrame::kVadActive;
audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
break;
}
case kOutputVADPassive: {
audio_frame->vad_activity_ = AudioFrame::kVadPassive;
audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
break;
}
case kOutputCNG: {
audio_frame->vad_activity_ = AudioFrame::kVadPassive;
audio_frame->speech_type_ = AudioFrame::kCNG;
break;
}
case kOutputPLC: {
// Don't change |audio_frame->vad_activity_|, it should be the same as
// |previous_audio_activity_|.
audio_frame->speech_type_ = AudioFrame::kPLC;
break;
}
case kOutputPLCtoCNG: {
audio_frame->vad_activity_ = AudioFrame::kVadPassive;
audio_frame->speech_type_ = AudioFrame::kPLCCNG;
break;
}
default:
assert(false);
}
} else {
// Always return kVadUnknown when receive VAD is inactive
audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
switch (type) {
case kOutputNormal: {
audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
break;
}
case kOutputCNG: {
audio_frame->speech_type_ = AudioFrame::kCNG;
break;
}
case kOutputPLC: {
audio_frame->speech_type_ = AudioFrame::kPLC;
break;
}
case kOutputPLCtoCNG: {
audio_frame->speech_type_ = AudioFrame::kPLCCNG;
break;
}
case kOutputVADPassive: {
// Normally, we should no get any VAD decision if post-decoding VAD is
// not active. However, if post-decoding VAD has been active then
// disabled, we might be here for couple of frames.
audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
LOG_F(LS_WARNING) << "Post-decoding VAD is disabled but output is "
<< "labeled VAD-passive";
break;
}
default:
assert(false);
}
}
}
// Is the given codec a CNG codec?
bool IsCng(int codec_id) {
return (codec_id == ACMCodecDB::kCNNB || codec_id == ACMCodecDB::kCNWB ||
codec_id == ACMCodecDB::kCNSWB || codec_id == ACMCodecDB::kCNFB);
}
} // namespace
AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
id_(config.id),
last_audio_decoder_(-1), // Invalid value.
previous_audio_activity_(AudioFrame::kVadPassive),
current_sample_rate_hz_(config.neteq_config.sample_rate_hz),
audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
nack_(),
nack_enabled_(false),
neteq_(NetEq::Create(config.neteq_config)),
vad_enabled_(true),
clock_(config.clock),
resampled_last_output_frame_(true),
av_sync_(false),
initial_delay_manager_(),
missing_packets_sync_stream_(),
late_packets_sync_stream_() {
assert(clock_);
for (int n = 0; n < ACMCodecDB::kMaxNumCodecs; ++n) {
decoders_[n].registered = false;
}
// Make sure we are on the same page as NetEq. Post-decode VAD is disabled by
// default in NetEq4, however, Audio Conference Mixer relies on VAD decision
// and fails if VAD decision is not provided.
if (vad_enabled_)
neteq_->EnableVad();
else
neteq_->DisableVad();
memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
}
AcmReceiver::~AcmReceiver() {
delete neteq_;
}
int AcmReceiver::SetMinimumDelay(int delay_ms) {
if (neteq_->SetMinimumDelay(delay_ms))
return 0;
LOG_FERR1(LS_ERROR, "AcmReceiver::SetExtraDelay", delay_ms);
return -1;
}
int AcmReceiver::SetInitialDelay(int delay_ms) {
if (delay_ms < 0 || delay_ms > 10000) {
return -1;
}
CriticalSectionScoped lock(crit_sect_.get());
if (delay_ms == 0) {
av_sync_ = false;
initial_delay_manager_.reset();
missing_packets_sync_stream_.reset();
late_packets_sync_stream_.reset();
neteq_->SetMinimumDelay(0);
return 0;
}
if (av_sync_ && initial_delay_manager_->PacketBuffered()) {
// Too late for this API. Only works before a call is started.
return -1;
}
// Most of places NetEq calls are not within AcmReceiver's critical section to
// improve performance. Here, this call has to be placed before the following
// block, therefore, we keep it inside critical section. Otherwise, we have to
// release |neteq_crit_sect_| and acquire it again, which seems an overkill.
if (!neteq_->SetMinimumDelay(delay_ms))
return -1;
const int kLatePacketThreshold = 5;
av_sync_ = true;
initial_delay_manager_.reset(new InitialDelayManager(delay_ms,
kLatePacketThreshold));
missing_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
late_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
return 0;
}
int AcmReceiver::SetMaximumDelay(int delay_ms) {
if (neteq_->SetMaximumDelay(delay_ms))
return 0;
LOG_FERR1(LS_ERROR, "AcmReceiver::SetExtraDelay", delay_ms);
return -1;
}
int AcmReceiver::LeastRequiredDelayMs() const {
return neteq_->LeastRequiredDelayMs();
}
int AcmReceiver::current_sample_rate_hz() const {
CriticalSectionScoped lock(crit_sect_.get());
return current_sample_rate_hz_;
}
// TODO(turajs): use one set of enumerators, e.g. the one defined in
// common_types.h
// TODO(henrik.lundin): This method is not used any longer. The call hierarchy
// stops in voe::Channel::SetNetEQPlayoutMode(). Remove it.
void AcmReceiver::SetPlayoutMode(AudioPlayoutMode mode) {
enum NetEqPlayoutMode playout_mode = kPlayoutOn;
switch (mode) {
case voice:
playout_mode = kPlayoutOn;
break;
case fax: // No change to background noise mode.
playout_mode = kPlayoutFax;
break;
case streaming:
playout_mode = kPlayoutStreaming;
break;
case off:
playout_mode = kPlayoutOff;
break;
}
neteq_->SetPlayoutMode(playout_mode);
}
AudioPlayoutMode AcmReceiver::PlayoutMode() const {
AudioPlayoutMode acm_mode = voice;
NetEqPlayoutMode mode = neteq_->PlayoutMode();
switch (mode) {
case kPlayoutOn:
acm_mode = voice;
break;
case kPlayoutOff:
acm_mode = off;
break;
case kPlayoutFax:
acm_mode = fax;
break;
case kPlayoutStreaming:
acm_mode = streaming;
break;
default:
assert(false);
}
return acm_mode;
}
int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
const uint8_t* incoming_payload,
size_t length_payload) {
uint32_t receive_timestamp = 0;
InitialDelayManager::PacketType packet_type =
InitialDelayManager::kUndefinedPacket;
bool new_codec = false;
const RTPHeader* header = &rtp_header.header; // Just a shorthand.
{
CriticalSectionScoped lock(crit_sect_.get());
int codec_id = RtpHeaderToCodecIndex(*header, incoming_payload);
if (codec_id < 0) {
LOG_F(LS_ERROR) << "Payload-type " << header->payloadType
<< " is not registered.";
return -1;
}
assert(codec_id < ACMCodecDB::kMaxNumCodecs);
const int sample_rate_hz = ACMCodecDB::CodecFreq(codec_id);
receive_timestamp = NowInTimestamp(sample_rate_hz);
if (IsCng(codec_id)) {
// If this is a CNG while the audio codec is not mono skip pushing in
// packets into NetEq.
if (last_audio_decoder_ >= 0 &&
decoders_[last_audio_decoder_].channels > 1)
return 0;
packet_type = InitialDelayManager::kCngPacket;
} else if (codec_id == ACMCodecDB::kAVT) {
packet_type = InitialDelayManager::kAvtPacket;
} else {
if (codec_id != last_audio_decoder_) {
// This is either the first audio packet or send codec is changed.
// Therefore, either NetEq buffer is empty or will be flushed when this
// packet inserted. Note that |last_audio_decoder_| is initialized to
// an invalid value (-1), hence, the above condition is true for the
// very first audio packet.
new_codec = true;
// Updating NACK'sampling rate is required, either first packet is
// received or codec is changed. Furthermore, reset is required if codec
// is changed (NetEq flushes its buffer so NACK should reset its list).
if (nack_enabled_) {
assert(nack_.get());
nack_->Reset();
nack_->UpdateSampleRate(sample_rate_hz);
}
last_audio_decoder_ = codec_id;
}
packet_type = InitialDelayManager::kAudioPacket;
}
if (nack_enabled_) {
assert(nack_.get());
nack_->UpdateLastReceivedPacket(header->sequenceNumber,
header->timestamp);
}
if (av_sync_) {
assert(initial_delay_manager_.get());
assert(missing_packets_sync_stream_.get());
// This updates |initial_delay_manager_| and specifies an stream of
// sync-packets, if required to be inserted. We insert the sync-packets
// when AcmReceiver lock is released and |decoder_lock_| is acquired.
initial_delay_manager_->UpdateLastReceivedPacket(
rtp_header, receive_timestamp, packet_type, new_codec, sample_rate_hz,
missing_packets_sync_stream_.get());
}
} // |crit_sect_| is released.
// If |missing_packets_sync_stream_| is allocated then we are in AV-sync and
// we may need to insert sync-packets. We don't check |av_sync_| as we are
// outside AcmReceiver's critical section.
if (missing_packets_sync_stream_.get()) {
InsertStreamOfSyncPackets(missing_packets_sync_stream_.get());
}
if (neteq_->InsertPacket(rtp_header, incoming_payload, length_payload,
receive_timestamp) < 0) {
LOG_FERR1(LS_ERROR, "AcmReceiver::InsertPacket", header->payloadType) <<
" Failed to insert packet";
return -1;
}
return 0;
}
int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
enum NetEqOutputType type;
int samples_per_channel;
int num_channels;
bool return_silence = false;
{
// Accessing members, take the lock.
CriticalSectionScoped lock(crit_sect_.get());
if (av_sync_) {
assert(initial_delay_manager_.get());
assert(late_packets_sync_stream_.get());
return_silence = GetSilence(desired_freq_hz, audio_frame);
uint32_t timestamp_now = NowInTimestamp(current_sample_rate_hz_);
initial_delay_manager_->LatePackets(timestamp_now,
late_packets_sync_stream_.get());
}
}
// If |late_packets_sync_stream_| is allocated then we have been in AV-sync
// mode and we might have to insert sync-packets.
if (late_packets_sync_stream_.get()) {
InsertStreamOfSyncPackets(late_packets_sync_stream_.get());
if (return_silence) // Silence generated, don't pull from NetEq.
return 0;
}
// Accessing members, take the lock.
CriticalSectionScoped lock(crit_sect_.get());
// Always write the output to |audio_buffer_| first.
if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
audio_buffer_.get(),
&samples_per_channel,
&num_channels,
&type) != NetEq::kOK) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "NetEq Failed.";
return -1;
}
// Update NACK.
int decoded_sequence_num = 0;
uint32_t decoded_timestamp = 0;
bool update_nack = nack_enabled_ && // Update NACK only if it is enabled.
neteq_->DecodedRtpInfo(&decoded_sequence_num, &decoded_timestamp);
if (update_nack) {
assert(nack_.get());
nack_->UpdateLastDecodedPacket(decoded_sequence_num, decoded_timestamp);
}
// NetEq always returns 10 ms of audio.
current_sample_rate_hz_ = samples_per_channel * 100;
// Update if resampling is required.
bool need_resampling = (desired_freq_hz != -1) &&
(current_sample_rate_hz_ != desired_freq_hz);
if (need_resampling && !resampled_last_output_frame_) {
// Prime the resampler with the last frame.
int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
samples_per_channel =
resampler_.Resample10Msec(last_audio_buffer_.get(),
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
temp_output);
if (samples_per_channel < 0) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio")
<< "Resampling last_audio_buffer_ failed.";
return -1;
}
}
// The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
// through resampling, or through straight memcpy.
// TODO(henrik.lundin) Glitches in the output may appear if the output rate
// from NetEq changes. See WebRTC issue 3923.
if (need_resampling) {
samples_per_channel =
resampler_.Resample10Msec(audio_buffer_.get(),
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
audio_frame->data_);
if (samples_per_channel < 0) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio")
<< "Resampling audio_buffer_ failed.";
return -1;
}
resampled_last_output_frame_ = true;
} else {
resampled_last_output_frame_ = false;
// We might end up here ONLY if codec is changed.
memcpy(audio_frame->data_,
audio_buffer_.get(),
samples_per_channel * num_channels * sizeof(int16_t));
}
// Swap buffers, so that the current audio is stored in |last_audio_buffer_|
// for next time.
audio_buffer_.swap(last_audio_buffer_);
audio_frame->num_channels_ = num_channels;
audio_frame->samples_per_channel_ = samples_per_channel;
audio_frame->sample_rate_hz_ = samples_per_channel * 100;
// Should set |vad_activity| before calling SetAudioFrameActivityAndType().
audio_frame->vad_activity_ = previous_audio_activity_;
SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
previous_audio_activity_ = audio_frame->vad_activity_;
call_stats_.DecodedByNetEq(audio_frame->speech_type_);
// Computes the RTP timestamp of the first sample in |audio_frame| from
// |GetPlayoutTimestamp|, which is the timestamp of the last sample of
// |audio_frame|.
uint32_t playout_timestamp = 0;
if (GetPlayoutTimestamp(&playout_timestamp)) {
audio_frame->timestamp_ =
playout_timestamp - audio_frame->samples_per_channel_;
} else {
// Remain 0 until we have a valid |playout_timestamp|.
audio_frame->timestamp_ = 0;
}
return 0;
}
int32_t AcmReceiver::AddCodec(int acm_codec_id,
uint8_t payload_type,
int channels,
AudioDecoder* audio_decoder) {
assert(acm_codec_id >= 0 && acm_codec_id < ACMCodecDB::kMaxNumCodecs);
NetEqDecoder neteq_decoder = ACMCodecDB::neteq_decoders_[acm_codec_id];
// Make sure the right decoder is registered for Opus.
if (neteq_decoder == kDecoderOpus && channels == 2) {
neteq_decoder = kDecoderOpus_2ch;
}
CriticalSectionScoped lock(crit_sect_.get());
// The corresponding NetEq decoder ID.
// If this coder has been registered before.
if (decoders_[acm_codec_id].registered) {
if (decoders_[acm_codec_id].payload_type == payload_type &&
decoders_[acm_codec_id].channels == channels) {
// Re-registering the same codec with the same payload-type. Do nothing
// and return.
return 0;
}
// Changing the payload-type or number of channels for this codec.
// First unregister. Then register with new payload-type/channels.
if (neteq_->RemovePayloadType(decoders_[acm_codec_id].payload_type) !=
NetEq::kOK) {
LOG_F(LS_ERROR) << "Cannot remover payload "
<< decoders_[acm_codec_id].payload_type;
return -1;
}
}
int ret_val;
if (!audio_decoder) {
ret_val = neteq_->RegisterPayloadType(neteq_decoder, payload_type);
} else {
ret_val = neteq_->RegisterExternalDecoder(
audio_decoder, neteq_decoder, payload_type);
}
if (ret_val != NetEq::kOK) {
LOG_FERR3(LS_ERROR, "AcmReceiver::AddCodec", acm_codec_id, payload_type,
channels);
// Registration failed, delete the allocated space and set the pointer to
// NULL, for the record.
decoders_[acm_codec_id].registered = false;
return -1;
}
decoders_[acm_codec_id].registered = true;
decoders_[acm_codec_id].payload_type = payload_type;
decoders_[acm_codec_id].channels = channels;
return 0;
}
void AcmReceiver::EnableVad() {
neteq_->EnableVad();
CriticalSectionScoped lock(crit_sect_.get());
vad_enabled_ = true;
}
void AcmReceiver::DisableVad() {
neteq_->DisableVad();
CriticalSectionScoped lock(crit_sect_.get());
vad_enabled_ = false;
}
void AcmReceiver::FlushBuffers() {
neteq_->FlushBuffers();
}
// If failed in removing one of the codecs, this method continues to remove as
// many as it can.
int AcmReceiver::RemoveAllCodecs() {
int ret_val = 0;
CriticalSectionScoped lock(crit_sect_.get());
for (int n = 0; n < ACMCodecDB::kMaxNumCodecs; ++n) {
if (decoders_[n].registered) {
if (neteq_->RemovePayloadType(decoders_[n].payload_type) == 0) {
decoders_[n].registered = false;
} else {
LOG_F(LS_ERROR) << "Cannot remove payload "
<< decoders_[n].payload_type;
ret_val = -1;
}
}
}
// No codec is registered, invalidate last audio decoder.
last_audio_decoder_ = -1;
return ret_val;
}
int AcmReceiver::RemoveCodec(uint8_t payload_type) {
int codec_index = PayloadType2CodecIndex(payload_type);
if (codec_index < 0) { // Such a payload-type is not registered.
return 0;
}
if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
LOG_FERR1(LS_ERROR, "AcmReceiver::RemoveCodec", payload_type);
return -1;
}
CriticalSectionScoped lock(crit_sect_.get());
decoders_[codec_index].registered = false;
if (last_audio_decoder_ == codec_index)
last_audio_decoder_ = -1; // Codec is removed, invalidate last decoder.
return 0;
}
void AcmReceiver::set_id(int id) {
CriticalSectionScoped lock(crit_sect_.get());
id_ = id;
}
bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
if (av_sync_) {
assert(initial_delay_manager_.get());
if (initial_delay_manager_->buffering()) {
return initial_delay_manager_->GetPlayoutTimestamp(timestamp);
}
}
return neteq_->GetPlayoutTimestamp(timestamp);
}
int AcmReceiver::last_audio_codec_id() const {
CriticalSectionScoped lock(crit_sect_.get());
return last_audio_decoder_;
}
int AcmReceiver::last_audio_payload_type() const {
CriticalSectionScoped lock(crit_sect_.get());
if (last_audio_decoder_ < 0)
return -1;
assert(decoders_[last_audio_decoder_].registered);
return decoders_[last_audio_decoder_].payload_type;
}
int AcmReceiver::RedPayloadType() const {
CriticalSectionScoped lock(crit_sect_.get());
if (ACMCodecDB::kRED < 0 ||
!decoders_[ACMCodecDB::kRED].registered) {
LOG_F(LS_WARNING) << "RED is not registered.";
return -1;
}
return decoders_[ACMCodecDB::kRED].payload_type;
}
int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
CriticalSectionScoped lock(crit_sect_.get());
if (last_audio_decoder_ < 0) {
return -1;
}
assert(decoders_[last_audio_decoder_].registered);
memcpy(codec, &ACMCodecDB::database_[last_audio_decoder_], sizeof(CodecInst));
codec->pltype = decoders_[last_audio_decoder_].payload_type;
codec->channels = decoders_[last_audio_decoder_].channels;
return 0;
}
void AcmReceiver::NetworkStatistics(ACMNetworkStatistics* acm_stat) {
NetEqNetworkStatistics neteq_stat;
// NetEq function always returns zero, so we don't check the return value.
neteq_->NetworkStatistics(&neteq_stat);
acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
acm_stat->currentExpandRate = neteq_stat.expand_rate;
acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
acm_stat->clockDriftPPM = neteq_stat.clockdrift_ppm;
acm_stat->addedSamples = neteq_stat.added_zero_samples;
std::vector<int> waiting_times;
neteq_->WaitingTimes(&waiting_times);
size_t size = waiting_times.size();
if (size == 0) {
acm_stat->meanWaitingTimeMs = -1;
acm_stat->medianWaitingTimeMs = -1;
acm_stat->minWaitingTimeMs = -1;
acm_stat->maxWaitingTimeMs = -1;
} else {
std::sort(waiting_times.begin(), waiting_times.end());
if ((size & 0x1) == 0) {
acm_stat->medianWaitingTimeMs = (waiting_times[size / 2 - 1] +
waiting_times[size / 2]) / 2;
} else {
acm_stat->medianWaitingTimeMs = waiting_times[size / 2];
}
acm_stat->minWaitingTimeMs = waiting_times.front();
acm_stat->maxWaitingTimeMs = waiting_times.back();
double sum = 0;
for (size_t i = 0; i < size; ++i) {
sum += waiting_times[i];
}
acm_stat->meanWaitingTimeMs = static_cast<int>(sum / size);
}
}
int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
CodecInst* codec) const {
CriticalSectionScoped lock(crit_sect_.get());
int codec_index = PayloadType2CodecIndex(payload_type);
if (codec_index < 0) {
LOG_FERR1(LS_ERROR, "AcmReceiver::DecoderByPayloadType", payload_type);
return -1;
}
memcpy(codec, &ACMCodecDB::database_[codec_index], sizeof(CodecInst));
codec->pltype = decoders_[codec_index].payload_type;
codec->channels = decoders_[codec_index].channels;
return 0;
}
int AcmReceiver::PayloadType2CodecIndex(uint8_t payload_type) const {
for (int n = 0; n < ACMCodecDB::kMaxNumCodecs; ++n) {
if (decoders_[n].registered && decoders_[n].payload_type == payload_type) {
return n;
}
}
return -1;
}
int AcmReceiver::EnableNack(size_t max_nack_list_size) {
// Don't do anything if |max_nack_list_size| is out of range.
if (max_nack_list_size == 0 || max_nack_list_size > Nack::kNackListSizeLimit)
return -1;
CriticalSectionScoped lock(crit_sect_.get());
if (!nack_enabled_) {
nack_.reset(Nack::Create(kNackThresholdPackets));
nack_enabled_ = true;
// Sampling rate might need to be updated if we change from disable to
// enable. Do it if the receive codec is valid.
if (last_audio_decoder_ >= 0) {
nack_->UpdateSampleRate(
ACMCodecDB::database_[last_audio_decoder_].plfreq);
}
}
return nack_->SetMaxNackListSize(max_nack_list_size);
}
void AcmReceiver::DisableNack() {
CriticalSectionScoped lock(crit_sect_.get());
nack_.reset(); // Memory is released.
nack_enabled_ = false;
}
std::vector<uint16_t> AcmReceiver::GetNackList(
int round_trip_time_ms) const {
CriticalSectionScoped lock(crit_sect_.get());
if (round_trip_time_ms < 0) {
WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
"GetNackList: round trip time cannot be negative."
" round_trip_time_ms=%d", round_trip_time_ms);
}
if (nack_enabled_ && round_trip_time_ms >= 0) {
assert(nack_.get());
return nack_->GetNackList(round_trip_time_ms);
}
std::vector<uint16_t> empty_list;
return empty_list;
}
void AcmReceiver::ResetInitialDelay() {
{
CriticalSectionScoped lock(crit_sect_.get());
av_sync_ = false;
initial_delay_manager_.reset(NULL);
missing_packets_sync_stream_.reset(NULL);
late_packets_sync_stream_.reset(NULL);
}
neteq_->SetMinimumDelay(0);
// TODO(turajs): Should NetEq Buffer be flushed?
}
// This function is called within critical section, no need to acquire a lock.
bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) {
assert(av_sync_);
assert(initial_delay_manager_.get());
if (!initial_delay_manager_->buffering()) {
return false;
}
// We stop accumulating packets, if the number of packets or the total size
// exceeds a threshold.
int num_packets;
int max_num_packets;
const float kBufferingThresholdScale = 0.9f;
neteq_->PacketBufferStatistics(&num_packets, &max_num_packets);
if (num_packets > max_num_packets * kBufferingThresholdScale) {
initial_delay_manager_->DisableBuffering();
return false;
}
// Update statistics.
call_stats_.DecodedBySilenceGenerator();
// Set the values if already got a packet, otherwise set to default values.
if (last_audio_decoder_ >= 0) {
current_sample_rate_hz_ = ACMCodecDB::database_[last_audio_decoder_].plfreq;
frame->num_channels_ = decoders_[last_audio_decoder_].channels;
} else {
frame->num_channels_ = 1;
}
// Set the audio frame's sampling frequency.
if (desired_sample_rate_hz > 0) {
frame->sample_rate_hz_ = desired_sample_rate_hz;
} else {
frame->sample_rate_hz_ = current_sample_rate_hz_;
}
frame->samples_per_channel_ = frame->sample_rate_hz_ / 100; // Always 10 ms.
frame->speech_type_ = AudioFrame::kCNG;
frame->vad_activity_ = AudioFrame::kVadPassive;
int samples = frame->samples_per_channel_ * frame->num_channels_;
memset(frame->data_, 0, samples * sizeof(int16_t));
return true;
}
int AcmReceiver::RtpHeaderToCodecIndex(
const RTPHeader &rtp_header, const uint8_t* payload) const {
uint8_t payload_type = rtp_header.payloadType;
if (ACMCodecDB::kRED >= 0 && // This ensures that RED is defined in WebRTC.
decoders_[ACMCodecDB::kRED].registered &&
payload_type == decoders_[ACMCodecDB::kRED].payload_type) {
// This is a RED packet, get the payload of the audio codec.
payload_type = payload[0] & 0x7F;
}
// Check if the payload is registered.
return PayloadType2CodecIndex(payload_type);
}
uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
// Down-cast the time to (32-6)-bit since we only care about
// the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
// We masked 6 most significant bits of 32-bit so there is no overflow in
// the conversion from milliseconds to timestamp.
const uint32_t now_in_ms = static_cast<uint32_t>(
clock_->TimeInMilliseconds() & 0x03ffffff);
return static_cast<uint32_t>(
(decoder_sampling_rate / 1000) * now_in_ms);
}
// This function only interacts with |neteq_|, therefore, it does not have to
// be within critical section of AcmReceiver. It is inserting packets
// into NetEq, so we call it when |decode_lock_| is acquired. However, this is
// not essential as sync-packets do not interact with codecs (especially BWE).
void AcmReceiver::InsertStreamOfSyncPackets(
InitialDelayManager::SyncStream* sync_stream) {
assert(sync_stream);
assert(av_sync_);
for (int n = 0; n < sync_stream->num_sync_packets; ++n) {
neteq_->InsertSyncPacket(sync_stream->rtp_info,
sync_stream->receive_timestamp);
++sync_stream->rtp_info.header.sequenceNumber;
sync_stream->rtp_info.header.timestamp += sync_stream->timestamp_step;
sync_stream->receive_timestamp += sync_stream->timestamp_step;
}
}
void AcmReceiver::GetDecodingCallStatistics(
AudioDecodingCallStats* stats) const {
CriticalSectionScoped lock(crit_sect_.get());
*stats = call_stats_.GetDecodingStatistics();
}
} // namespace acm2
} // namespace webrtc