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/*
* Copyright (c) 2015 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 "audio/audio_send_stream.h"
#include <string>
#include <utility>
#include <vector>
#include "audio/audio_state.h"
#include "audio/conversion.h"
#include "audio/scoped_voe_interface.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/bitrate_controller/include/bitrate_controller.h"
#include "modules/congestion_controller/include/send_side_congestion_controller.h"
#include "modules/pacing/paced_sender.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/function_view.h"
#include "rtc_base/logging.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/timeutils.h"
#include "voice_engine/channel_proxy.h"
#include "voice_engine/include/voe_base.h"
#include "voice_engine/transmit_mixer.h"
#include "voice_engine/voice_engine_impl.h"
namespace webrtc {
namespace internal {
// TODO(eladalon): Subsequent CL will make these values experiment-dependent.
constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000;
constexpr size_t kPacketLossRateMinNumAckedPackets = 50;
constexpr size_t kRecoverablePacketLossRateMinNumAckedPairs = 40;
namespace {
void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy,
rtc::FunctionView<void(AudioEncoder*)> lambda) {
channel_proxy->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
RTC_DCHECK(encoder_ptr);
lambda(encoder_ptr->get());
});
}
} // namespace
// TODO(saza): Move this declaration further down when we can use
// std::make_unique.
class AudioSendStream::TimedTransport : public Transport {
public:
TimedTransport(Transport* transport, TimeInterval* time_interval)
: transport_(transport), lifetime_(time_interval) {}
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) {
if (lifetime_) {
lifetime_->Extend();
}
return transport_->SendRtp(packet, length, options);
}
bool SendRtcp(const uint8_t* packet, size_t length) {
return transport_->SendRtcp(packet, length);
}
~TimedTransport() {}
private:
Transport* transport_;
TimeInterval* lifetime_;
};
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
RtpTransportControllerSendInterface* transport,
BitrateAllocator* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const rtc::Optional<RtpState>& suspended_rtp_state)
: worker_queue_(worker_queue),
config_(Config(nullptr)),
audio_state_(audio_state),
event_log_(event_log),
bitrate_allocator_(bitrate_allocator),
transport_(transport),
packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
kPacketLossRateMinNumAckedPackets,
kRecoverablePacketLossRateMinNumAckedPairs),
rtp_rtcp_module_(nullptr),
suspended_rtp_state_(suspended_rtp_state) {
LOG(LS_INFO) << "AudioSendStream: " << config.ToString();
RTC_DCHECK_NE(config.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
RTC_DCHECK(transport);
RTC_DCHECK(transport->send_side_cc());
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config.voe_channel_id);
channel_proxy_->SetRtcEventLog(event_log_);
channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
channel_proxy_->SetRTCPStatus(true);
transport_->send_side_cc()->RegisterPacketFeedbackObserver(this);
RtpReceiver* rtpReceiver = nullptr; // Unused, but required for call.
channel_proxy_->GetRtpRtcp(&rtp_rtcp_module_, &rtpReceiver);
RTC_DCHECK(rtp_rtcp_module_);
ConfigureStream(this, config, true);
pacer_thread_checker_.DetachFromThread();
}
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
transport_->send_side_cc()->DeRegisterPacketFeedbackObserver(this);
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->ResetSenderCongestionControlObjects();
channel_proxy_->SetRtcEventLog(nullptr);
channel_proxy_->SetRtcpRttStats(nullptr);
}
const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return config_;
}
void AudioSendStream::Reconfigure(
const webrtc::AudioSendStream::Config& new_config) {
ConfigureStream(this, new_config, false);
}
void AudioSendStream::ConfigureStream(
webrtc::internal::AudioSendStream* stream,
const webrtc::AudioSendStream::Config& new_config,
bool first_time) {
LOG(LS_INFO) << "AudioSendStream::Configuring: " << new_config.ToString();
const auto& channel_proxy = stream->channel_proxy_;
const auto& old_config = stream->config_;
if (first_time || old_config.rtp.ssrc != new_config.rtp.ssrc) {
channel_proxy->SetLocalSSRC(new_config.rtp.ssrc);
if (stream->suspended_rtp_state_) {
stream->rtp_rtcp_module_->SetRtpState(*stream->suspended_rtp_state_);
}
}
if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {
channel_proxy->SetRTCP_CNAME(new_config.rtp.c_name);
}
// TODO(solenberg): Config NACK history window (which is a packet count),
// using the actual packet size for the configured codec.
if (first_time || old_config.rtp.nack.rtp_history_ms !=
new_config.rtp.nack.rtp_history_ms) {
channel_proxy->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
new_config.rtp.nack.rtp_history_ms / 20);
}
if (first_time ||
new_config.send_transport != old_config.send_transport) {
if (old_config.send_transport) {
channel_proxy->DeRegisterExternalTransport();
}
if (new_config.send_transport) {
stream->timed_send_transport_adapter_.reset(new TimedTransport(
new_config.send_transport, &stream->active_lifetime_));
} else {
stream->timed_send_transport_adapter_.reset(nullptr);
}
channel_proxy->RegisterExternalTransport(
stream->timed_send_transport_adapter_.get());
}
// RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
// reserved for padding and MUST NOT be used as a local identifier.
// So it should be safe to use 0 here to indicate "not configured".
struct ExtensionIds {
int audio_level = 0;
int transport_sequence_number = 0;
};
auto find_extension_ids = [](const std::vector<RtpExtension>& extensions) {
ExtensionIds ids;
for (const auto& extension : extensions) {
if (extension.uri == RtpExtension::kAudioLevelUri) {
ids.audio_level = extension.id;
} else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
ids.transport_sequence_number = extension.id;
}
}
return ids;
};
const ExtensionIds old_ids = find_extension_ids(old_config.rtp.extensions);
const ExtensionIds new_ids = find_extension_ids(new_config.rtp.extensions);
// Audio level indication
if (first_time || new_ids.audio_level != old_ids.audio_level) {
channel_proxy->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0,
new_ids.audio_level);
}
// Transport sequence number
if (first_time ||
new_ids.transport_sequence_number != old_ids.transport_sequence_number) {
if (!first_time) {
channel_proxy->ResetSenderCongestionControlObjects();
stream->bandwidth_observer_.reset();
}
if (new_ids.transport_sequence_number != 0) {
channel_proxy->EnableSendTransportSequenceNumber(
new_ids.transport_sequence_number);
stream->transport_->send_side_cc()->EnablePeriodicAlrProbing(true);
stream->bandwidth_observer_.reset(stream->transport_->send_side_cc()
->GetBitrateController()
->CreateRtcpBandwidthObserver());
}
channel_proxy->RegisterSenderCongestionControlObjects(
stream->transport_, stream->bandwidth_observer_.get());
}
if (!ReconfigureSendCodec(stream, new_config)) {
LOG(LS_ERROR) << "Failed to set up send codec state.";
}
ReconfigureBitrateObserver(stream, new_config);
stream->config_ = new_config;
}
void AudioSendStream::Start() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) {
ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps);
}
ScopedVoEInterface<VoEBase> base(voice_engine());
int error = base->StartSend(config_.voe_channel_id);
if (error != 0) {
LOG(LS_ERROR) << "AudioSendStream::Start failed with error: " << error;
}
}
void AudioSendStream::Stop() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RemoveBitrateObserver();
ScopedVoEInterface<VoEBase> base(voice_engine());
int error = base->StopSend(config_.voe_channel_id);
if (error != 0) {
LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error;
}
}
bool AudioSendStream::SendTelephoneEvent(int payload_type,
int payload_frequency, int event,
int duration_ms) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type,
payload_frequency) &&
channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
}
void AudioSendStream::SetMuted(bool muted) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
channel_proxy_->SetInputMute(muted);
}
webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
webrtc::AudioSendStream::Stats stats;
stats.local_ssrc = config_.rtp.ssrc;
webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
stats.bytes_sent = call_stats.bytesSent;
stats.packets_sent = call_stats.packetsSent;
// RTT isn't known until a RTCP report is received. Until then, VoiceEngine
// returns 0 to indicate an error value.
if (call_stats.rttMs > 0) {
stats.rtt_ms = call_stats.rttMs;
}
// TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable
// implementation.
stats.aec_quality_min = -1;
if (config_.send_codec_spec) {
const auto& spec = *config_.send_codec_spec;
stats.codec_name = spec.format.name;
stats.codec_payload_type = rtc::Optional<int>(spec.payload_type);
// Get data from the last remote RTCP report.
for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
// Lookup report for send ssrc only.
if (block.source_SSRC == stats.local_ssrc) {
stats.packets_lost = block.cumulative_num_packets_lost;
stats.fraction_lost = Q8ToFloat(block.fraction_lost);
stats.ext_seqnum = block.extended_highest_sequence_number;
// Convert timestamps to milliseconds.
if (spec.format.clockrate_hz / 1000 > 0) {
stats.jitter_ms =
block.interarrival_jitter / (spec.format.clockrate_hz / 1000);
}
break;
}
}
}
ScopedVoEInterface<VoEBase> base(voice_engine());
RTC_DCHECK(base->transmit_mixer());
stats.audio_level = base->transmit_mixer()->AudioLevelFullRange();
RTC_DCHECK_LE(0, stats.audio_level);
stats.total_input_energy = base->transmit_mixer()->GetTotalInputEnergy();
stats.total_input_duration = base->transmit_mixer()->GetTotalInputDuration();
RTC_DCHECK(audio_state_->audio_processing());
auto audio_processing_stats =
audio_state_->audio_processing()->GetStatistics();
stats.echo_delay_median_ms = audio_processing_stats.delay_median;
stats.echo_delay_std_ms = audio_processing_stats.delay_standard_deviation;
stats.echo_return_loss = audio_processing_stats.echo_return_loss.instant();
stats.echo_return_loss_enhancement =
audio_processing_stats.echo_return_loss_enhancement.instant();
stats.residual_echo_likelihood =
audio_processing_stats.residual_echo_likelihood;
stats.residual_echo_likelihood_recent_max =
audio_processing_stats.residual_echo_likelihood_recent_max;
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
stats.typing_noise_detected = audio_state->typing_noise_detected();
stats.ana_statistics = channel_proxy_->GetANAStatistics();
return stats;
}
void AudioSendStream::SignalNetworkState(NetworkState state) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
}
bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
return channel_proxy_->ReceivedRTCPPacket(packet, length);
}
uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt,
int64_t bwe_period_ms) {
// A send stream may be allocated a bitrate of zero if the allocator decides
// to disable it. For now we ignore this decision and keep sending on min
// bitrate.
if (bitrate_bps == 0) {
bitrate_bps = config_.min_bitrate_bps;
}
RTC_DCHECK_GE(bitrate_bps,
static_cast<uint32_t>(config_.min_bitrate_bps));
// The bitrate allocator might allocate an higher than max configured bitrate
// if there is room, to allow for, as example, extra FEC. Ignore that for now.
const uint32_t max_bitrate_bps = config_.max_bitrate_bps;
if (bitrate_bps > max_bitrate_bps)
bitrate_bps = max_bitrate_bps;
channel_proxy_->SetBitrate(bitrate_bps, bwe_period_ms);
// The amount of audio protection is not exposed by the encoder, hence
// always returning 0.
return 0;
}
void AudioSendStream::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
// Only packets that belong to this stream are of interest.
if (ssrc == config_.rtp.ssrc) {
rtc::CritScope lock(&packet_loss_tracker_cs_);
// TODO(eladalon): This function call could potentially reset the window,
// setting both PLR and RPLR to unknown. Consider (during upcoming
// refactoring) passing an indication of such an event.
packet_loss_tracker_.OnPacketAdded(seq_num, rtc::TimeMillis());
}
}
void AudioSendStream::OnPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
rtc::Optional<float> plr;
rtc::Optional<float> rplr;
{
rtc::CritScope lock(&packet_loss_tracker_cs_);
packet_loss_tracker_.OnPacketFeedbackVector(packet_feedback_vector);
plr = packet_loss_tracker_.GetPacketLossRate();
rplr = packet_loss_tracker_.GetRecoverablePacketLossRate();
}
// TODO(eladalon): If R/PLR go back to unknown, no indication is given that
// the previously sent value is no longer relevant. This will be taken care
// of with some refactoring which is now being done.
if (plr) {
channel_proxy_->OnTwccBasedUplinkPacketLossRate(*plr);
}
if (rplr) {
channel_proxy_->OnRecoverableUplinkPacketLossRate(*rplr);
}
}
void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
transport_->send_side_cc()->SetTransportOverhead(
transport_overhead_per_packet);
channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
}
RtpState AudioSendStream::GetRtpState() const {
return rtp_rtcp_module_->GetRtpState();
}
const TimeInterval& AudioSendStream::GetActiveLifetime() const {
return active_lifetime_;
}
VoiceEngine* AudioSendStream::voice_engine() const {
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
VoiceEngine* voice_engine = audio_state->voice_engine();
RTC_DCHECK(voice_engine);
return voice_engine;
}
// Apply current codec settings to a single voe::Channel used for sending.
bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
const Config& new_config) {
RTC_DCHECK(new_config.send_codec_spec);
const auto& spec = *new_config.send_codec_spec;
RTC_DCHECK(new_config.encoder_factory);
std::unique_ptr<AudioEncoder> encoder =
new_config.encoder_factory->MakeAudioEncoder(spec.payload_type,
spec.format);
if (!encoder) {
LOG(LS_ERROR) << "Unable to create encoder for " << spec.format;
return false;
}
// If a bitrate has been specified for the codec, use it over the
// codec's default.
if (spec.target_bitrate_bps) {
encoder->OnReceivedTargetAudioBitrate(*spec.target_bitrate_bps);
}
// Enable ANA if configured (currently only used by Opus).
if (new_config.audio_network_adaptor_config) {
if (encoder->EnableAudioNetworkAdaptor(
*new_config.audio_network_adaptor_config, stream->event_log_)) {
LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
<< new_config.rtp.ssrc;
} else {
RTC_NOTREACHED();
}
}
// Wrap the encoder in a an AudioEncoderCNG, if VAD is enabled.
if (spec.cng_payload_type) {
AudioEncoderCng::Config cng_config;
cng_config.num_channels = encoder->NumChannels();
cng_config.payload_type = *spec.cng_payload_type;
cng_config.speech_encoder = std::move(encoder);
cng_config.vad_mode = Vad::kVadNormal;
encoder.reset(new AudioEncoderCng(std::move(cng_config)));
stream->RegisterCngPayloadType(
*spec.cng_payload_type,
new_config.send_codec_spec->format.clockrate_hz);
}
stream->channel_proxy_->SetEncoder(new_config.send_codec_spec->payload_type,
std::move(encoder));
return true;
}
bool AudioSendStream::ReconfigureSendCodec(AudioSendStream* stream,
const Config& new_config) {
const auto& old_config = stream->config_;
if (!new_config.send_codec_spec) {
// We cannot de-configure a send codec. So we will do nothing.
// By design, the send codec should have not been configured.
RTC_DCHECK(!old_config.send_codec_spec);
return true;
}
if (new_config.send_codec_spec == old_config.send_codec_spec &&
new_config.audio_network_adaptor_config ==
old_config.audio_network_adaptor_config) {
return true;
}
// If we have no encoder, or the format or payload type's changed, create a
// new encoder.
if (!old_config.send_codec_spec ||
new_config.send_codec_spec->format !=
old_config.send_codec_spec->format ||
new_config.send_codec_spec->payload_type !=
old_config.send_codec_spec->payload_type) {
return SetupSendCodec(stream, new_config);
}
const rtc::Optional<int>& new_target_bitrate_bps =
new_config.send_codec_spec->target_bitrate_bps;
// If a bitrate has been specified for the codec, use it over the
// codec's default.
if (new_target_bitrate_bps &&
new_target_bitrate_bps !=
old_config.send_codec_spec->target_bitrate_bps) {
CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
encoder->OnReceivedTargetAudioBitrate(*new_target_bitrate_bps);
});
}
ReconfigureANA(stream, new_config);
ReconfigureCNG(stream, new_config);
return true;
}
void AudioSendStream::ReconfigureANA(AudioSendStream* stream,
const Config& new_config) {
if (new_config.audio_network_adaptor_config ==
stream->config_.audio_network_adaptor_config) {
return;
}
if (new_config.audio_network_adaptor_config) {
CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
if (encoder->EnableAudioNetworkAdaptor(
*new_config.audio_network_adaptor_config, stream->event_log_)) {
LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
<< new_config.rtp.ssrc;
} else {
RTC_NOTREACHED();
}
});
} else {
CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
encoder->DisableAudioNetworkAdaptor();
});
LOG(LS_INFO) << "Audio network adaptor disabled on SSRC "
<< new_config.rtp.ssrc;
}
}
void AudioSendStream::ReconfigureCNG(AudioSendStream* stream,
const Config& new_config) {
if (new_config.send_codec_spec->cng_payload_type ==
stream->config_.send_codec_spec->cng_payload_type) {
return;
}
// Register the CNG payload type if it's been added, don't do anything if CNG
// is removed. Payload types must not be redefined.
if (new_config.send_codec_spec->cng_payload_type) {
stream->RegisterCngPayloadType(
*new_config.send_codec_spec->cng_payload_type,
new_config.send_codec_spec->format.clockrate_hz);
}
// Wrap or unwrap the encoder in an AudioEncoderCNG.
stream->channel_proxy_->ModifyEncoder(
[&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
std::unique_ptr<AudioEncoder> old_encoder(std::move(*encoder_ptr));
auto sub_encoders = old_encoder->ReclaimContainedEncoders();
if (!sub_encoders.empty()) {
// Replace enc with its sub encoder. We need to put the sub
// encoder in a temporary first, since otherwise the old value
// of enc would be destroyed before the new value got assigned,
// which would be bad since the new value is a part of the old
// value.
auto tmp = std::move(sub_encoders[0]);
old_encoder = std::move(tmp);
}
if (new_config.send_codec_spec->cng_payload_type) {
AudioEncoderCng::Config config;
config.speech_encoder = std::move(old_encoder);
config.num_channels = config.speech_encoder->NumChannels();
config.payload_type = *new_config.send_codec_spec->cng_payload_type;
config.vad_mode = Vad::kVadNormal;
encoder_ptr->reset(new AudioEncoderCng(std::move(config)));
} else {
*encoder_ptr = std::move(old_encoder);
}
});
}
void AudioSendStream::ReconfigureBitrateObserver(
AudioSendStream* stream,
const webrtc::AudioSendStream::Config& new_config) {
// Since the Config's default is for both of these to be -1, this test will
// allow us to configure the bitrate observer if the new config has bitrate
// limits set, but would only have us call RemoveBitrateObserver if we were
// previously configured with bitrate limits.
if (stream->config_.min_bitrate_bps == new_config.min_bitrate_bps &&
stream->config_.max_bitrate_bps == new_config.max_bitrate_bps) {
return;
}
if (new_config.min_bitrate_bps != -1 && new_config.max_bitrate_bps != -1) {
stream->ConfigureBitrateObserver(new_config.min_bitrate_bps,
new_config.max_bitrate_bps);
} else {
stream->RemoveBitrateObserver();
}
}
void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps,
int max_bitrate_bps) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(max_bitrate_bps, min_bitrate_bps);
rtc::Event thread_sync_event(false /* manual_reset */, false);
worker_queue_->PostTask([&] {
// We may get a callback immediately as the observer is registered, so make
// sure the bitrate limits in config_ are up-to-date.
config_.min_bitrate_bps = min_bitrate_bps;
config_.max_bitrate_bps = max_bitrate_bps;
bitrate_allocator_->AddObserver(this, min_bitrate_bps, max_bitrate_bps, 0,
true);
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
}
void AudioSendStream::RemoveBitrateObserver() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
rtc::Event thread_sync_event(false /* manual_reset */, false);
worker_queue_->PostTask([this, &thread_sync_event] {
bitrate_allocator_->RemoveObserver(this);
thread_sync_event.Set();
});
thread_sync_event.Wait(rtc::Event::kForever);
}
void AudioSendStream::RegisterCngPayloadType(int payload_type,
int clockrate_hz) {
const CodecInst codec = {payload_type, "CN", clockrate_hz, 0, 1, 0};
if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
rtp_rtcp_module_->DeRegisterSendPayload(codec.pltype);
if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
"RTP/RTCP module";
}
}
}
} // namespace internal
} // namespace webrtc