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webrtc_m130/audio/audio_send_stream.cc
Alex Narest b3944f021d Media track ID visibility at BWE level 
Track IDs are assigned by application during track creation. 
Track IDs are used by custom bitrate allocation strategies to identify tracks. 
Track ID can be empty, in that case bitrate allocation strategies will not be able to handle
these tracks specifically and will handle them as a default.

Bug: webrtc:8243
Change-Id: I89987e33328320bfd0539ad532342df6da144c98
Reviewed-on: https://webrtc-review.googlesource.com/4820
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20285}
2017-10-13 13:47:07 +00:00

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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);
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();
// Signal congestion controller this object is ready for OnPacket* callbacks.
transport_->send_side_cc()->RegisterPacketFeedbackObserver(this);
}
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
transport_->send_side_cc()->DeRegisterPacketFeedbackObserver(this);
channel_proxy_->RegisterTransport(nullptr);
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->RegisterTransport(nullptr);
}
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->RegisterTransport(
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, config_.track_id);
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
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