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56ef305b6767fb762e1d49d976bdcc3cadb1d90a
platform-external-webrtc/audio/audio_send_stream.cc
Oskar Sundbom 56ef305b67 Move event logging of config into AudioSendStream. 
It was previously logged in Call, but streams are not always created
with the full configuration, which caused header extensions to be
missing from the log.

Bug: webrtc:9885
Change-Id: I86c0424004c4629ebab0f6b155b83fb90e15b131
Reviewed-on: https://webrtc-review.googlesource.com/c/108601
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25483}
2018-11-02 12:04:24 +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 "absl/memory/memory.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/audio_format.h"
#include "api/call/transport.h"
#include "api/crypto/frameencryptorinterface.h"
#include "audio/audio_state.h"
#include "audio/channel_send.h"
#include "audio/channel_send_proxy.h"
#include "audio/conversion.h"
#include "call/rtp_config.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "common_audio/vad/include/vad.h"
#include "common_types.h" // NOLINT(build/include)
#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_processing/include/audio_processing.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/strings/audio_format_to_string.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace internal {
namespace {
// 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;
void CallEncoder(const std::unique_ptr<voe::ChannelSendProxy>& channel_proxy,
rtc::FunctionView<void(AudioEncoder*)> lambda) {
channel_proxy->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
RTC_DCHECK(encoder_ptr);
lambda(encoder_ptr->get());
});
}
std::unique_ptr<voe::ChannelSendProxy> CreateChannelAndProxy(
rtc::TaskQueue* worker_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* event_log,
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed) {
return absl::make_unique<voe::ChannelSendProxy>(
absl::make_unique<voe::ChannelSend>(
worker_queue, module_process_thread, media_transport, rtcp_rtt_stats,
event_log, frame_encryptor, crypto_options, extmap_allow_mixed));
}
void UpdateEventLogStreamConfig(RtcEventLog* event_log,
const AudioSendStream::Config& config,
const AudioSendStream::Config* old_config) {
using SendCodecSpec = AudioSendStream::Config::SendCodecSpec;
// Only update if any of the things we log have changed.
auto payload_types_equal = [](const absl::optional<SendCodecSpec>& a,
const absl::optional<SendCodecSpec>& b) {
if (a.has_value() && b.has_value()) {
return a->format.name == b->format.name &&
a->payload_type == b->payload_type;
}
return !a.has_value() && !b.has_value();
};
if (old_config && config.rtp.ssrc == old_config->rtp.ssrc &&
config.rtp.extensions == old_config->rtp.extensions &&
payload_types_equal(config.send_codec_spec,
old_config->send_codec_spec)) {
return;
}
auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
rtclog_config->local_ssrc = config.rtp.ssrc;
rtclog_config->rtp_extensions = config.rtp.extensions;
if (config.send_codec_spec) {
rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
config.send_codec_spec->payload_type, 0);
}
event_log->Log(absl::make_unique<RtcEventAudioSendStreamConfig>(
std::move(rtclog_config)));
}
} // namespace
// Helper class to track the actively sending lifetime of this stream.
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,
ProcessThread* module_process_thread,
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
TimeInterval* overall_call_lifetime)
: AudioSendStream(config,
audio_state,
worker_queue,
rtp_transport,
bitrate_allocator,
event_log,
rtcp_rtt_stats,
suspended_rtp_state,
overall_call_lifetime,
CreateChannelAndProxy(worker_queue,
module_process_thread,
config.media_transport,
rtcp_rtt_stats,
event_log,
config.frame_encryptor,
config.crypto_options,
config.rtp.extmap_allow_mixed)) {}
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
RtpTransportControllerSendInterface* rtp_transport,
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
TimeInterval* overall_call_lifetime,
std::unique_ptr<voe::ChannelSendProxy> channel_proxy)
: worker_queue_(worker_queue),
config_(Config(/*send_transport=*/nullptr,
/*media_transport=*/nullptr)),
audio_state_(audio_state),
channel_proxy_(std::move(channel_proxy)),
event_log_(event_log),
bitrate_allocator_(bitrate_allocator),
rtp_transport_(rtp_transport),
packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
kPacketLossRateMinNumAckedPackets,
kRecoverablePacketLossRateMinNumAckedPairs),
rtp_rtcp_module_(nullptr),
suspended_rtp_state_(suspended_rtp_state),
overall_call_lifetime_(overall_call_lifetime) {
RTC_LOG(LS_INFO) << "AudioSendStream: " << config.rtp.ssrc;
RTC_DCHECK(worker_queue_);
RTC_DCHECK(audio_state_);
RTC_DCHECK(channel_proxy_);
RTC_DCHECK(bitrate_allocator_);
// TODO(nisse): Eventually, we should have only media_transport. But for the
// time being, we can have either. When media transport is injected, there
// should be no rtp_transport, and below check should be strengthened to XOR
// (either rtp_transport or media_transport but not both).
RTC_DCHECK(rtp_transport || config.media_transport);
RTC_DCHECK(overall_call_lifetime_);
channel_proxy_->SetRTCPStatus(true);
rtp_rtcp_module_ = channel_proxy_->GetRtpRtcp();
RTC_DCHECK(rtp_rtcp_module_);
ConfigureStream(this, config, true);
pacer_thread_checker_.DetachFromThread();
if (rtp_transport_) {
// Signal congestion controller this object is ready for OnPacket*
// callbacks.
rtp_transport_->RegisterPacketFeedbackObserver(this);
}
}
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc;
RTC_DCHECK(!sending_);
if (rtp_transport_) {
rtp_transport_->DeRegisterPacketFeedbackObserver(this);
channel_proxy_->RegisterTransport(nullptr);
channel_proxy_->ResetSenderCongestionControlObjects();
}
// Lifetime can only be updated after deregistering
// |timed_send_transport_adapter_| in the underlying channel object to avoid
// data races in |active_lifetime_|.
overall_call_lifetime_->Extend(active_lifetime_);
}
const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return config_;
}
void AudioSendStream::Reconfigure(
const webrtc::AudioSendStream::Config& new_config) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
ConfigureStream(this, new_config, false);
}
AudioSendStream::ExtensionIds AudioSendStream::FindExtensionIds(
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;
} else if (extension.uri == RtpExtension::kMidUri) {
ids.mid = extension.id;
}
}
return ids;
}
void AudioSendStream::ConfigureStream(
webrtc::internal::AudioSendStream* stream,
const webrtc::AudioSendStream::Config& new_config,
bool first_time) {
RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: "
<< new_config.ToString();
UpdateEventLogStreamConfig(stream->event_log_, new_config,
first_time ? nullptr : &stream->config_);
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());
}
// Enable the frame encryptor if a new frame encryptor has been provided.
if (first_time || new_config.frame_encryptor != old_config.frame_encryptor) {
channel_proxy->SetFrameEncryptor(new_config.frame_encryptor);
}
if (first_time ||
new_config.rtp.extmap_allow_mixed != old_config.rtp.extmap_allow_mixed) {
channel_proxy->SetExtmapAllowMixed(new_config.rtp.extmap_allow_mixed);
}
const ExtensionIds old_ids = FindExtensionIds(old_config.rtp.extensions);
const ExtensionIds new_ids = FindExtensionIds(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);
}
bool transport_seq_num_id_changed =
new_ids.transport_sequence_number != old_ids.transport_sequence_number;
if (first_time ||
(transport_seq_num_id_changed &&
!webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC"))) {
if (!first_time) {
channel_proxy->ResetSenderCongestionControlObjects();
}
RtcpBandwidthObserver* bandwidth_observer = nullptr;
bool has_transport_sequence_number =
new_ids.transport_sequence_number != 0 &&
!webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC");
if (has_transport_sequence_number) {
channel_proxy->EnableSendTransportSequenceNumber(
new_ids.transport_sequence_number);
// Probing in application limited region is only used in combination with
// send side congestion control, wich depends on feedback packets which
// requires transport sequence numbers to be enabled.
if (stream->rtp_transport_) {
stream->rtp_transport_->EnablePeriodicAlrProbing(true);
bandwidth_observer = stream->rtp_transport_->GetBandwidthObserver();
}
}
if (stream->rtp_transport_) {
channel_proxy->RegisterSenderCongestionControlObjects(
stream->rtp_transport_, bandwidth_observer);
}
}
// MID RTP header extension.
if ((first_time || new_ids.mid != old_ids.mid ||
new_config.rtp.mid != old_config.rtp.mid) &&
new_ids.mid != 0 && !new_config.rtp.mid.empty()) {
channel_proxy->SetMid(new_config.rtp.mid, new_ids.mid);
}
if (!ReconfigureSendCodec(stream, new_config)) {
RTC_LOG(LS_ERROR) << "Failed to set up send codec state.";
}
if (stream->sending_) {
ReconfigureBitrateObserver(stream, new_config);
}
stream->config_ = new_config;
}
void AudioSendStream::Start() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (sending_) {
return;
}
bool has_transport_sequence_number =
FindExtensionIds(config_.rtp.extensions).transport_sequence_number != 0 &&
!webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC");
if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1 &&
!config_.has_dscp &&
(has_transport_sequence_number ||
!webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") ||
webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC"))) {
// Audio BWE is enabled.
rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps,
config_.bitrate_priority,
has_transport_sequence_number);
} else {
rtp_rtcp_module_->SetAsPartOfAllocation(false);
}
channel_proxy_->StartSend();
sending_ = true;
audio_state()->AddSendingStream(this, encoder_sample_rate_hz_,
encoder_num_channels_);
}
void AudioSendStream::Stop() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!sending_) {
return;
}
RemoveBitrateObserver();
channel_proxy_->StopSend();
sending_ = false;
audio_state()->RemoveSendingStream(this);
}
void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
channel_proxy_->ProcessAndEncodeAudio(std::move(audio_frame));
}
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 {
return GetStats(true);
}
webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
bool has_remote_tracks) const {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
webrtc::AudioSendStream::Stats stats;
stats.local_ssrc = config_.rtp.ssrc;
stats.target_bitrate_bps = channel_proxy_->GetBitrate();
webrtc::CallSendStatistics 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;
}
if (config_.send_codec_spec) {
const auto& spec = *config_.send_codec_spec;
stats.codec_name = spec.format.name;
stats.codec_payload_type = 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;
}
}
}
AudioState::Stats input_stats = audio_state()->GetAudioInputStats();
stats.audio_level = input_stats.audio_level;
stats.total_input_energy = input_stats.total_energy;
stats.total_input_duration = input_stats.total_duration;
stats.typing_noise_detected = audio_state()->typing_noise_detected();
stats.ana_statistics = channel_proxy_->GetANAStatistics();
RTC_DCHECK(audio_state_->audio_processing());
stats.apm_statistics =
audio_state_->audio_processing()->GetStatistics(has_remote_tracks);
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(BitrateAllocationUpdate update) {
// 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 (update.bitrate_bps == 0) {
update.bitrate_bps = config_.min_bitrate_bps;
}
RTC_DCHECK_GE(update.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 (update.bitrate_bps > max_bitrate_bps)
update.bitrate_bps = max_bitrate_bps;
channel_proxy_->SetBitrate(update.bitrate_bps, update.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());
absl::optional<float> plr;
absl::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());
channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
}
RtpState AudioSendStream::GetRtpState() const {
return rtp_rtcp_module_->GetRtpState();
}
const voe::ChannelSendProxy& AudioSendStream::GetChannelProxy() const {
RTC_DCHECK(channel_proxy_.get());
return *channel_proxy_.get();
}
internal::AudioState* AudioSendStream::audio_state() {
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
RTC_DCHECK(audio_state);
return audio_state;
}
const internal::AudioState* AudioSendStream::audio_state() const {
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
RTC_DCHECK(audio_state);
return audio_state;
}
void AudioSendStream::StoreEncoderProperties(int sample_rate_hz,
size_t num_channels) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
encoder_sample_rate_hz_ = sample_rate_hz;
encoder_num_channels_ = num_channels;
if (sending_) {
// Update AudioState's information about the stream.
audio_state()->AddSendingStream(this, sample_rate_hz, num_channels);
}
}
// 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, new_config.codec_pair_id);
if (!encoder) {
RTC_DLOG(LS_ERROR) << "Unable to create encoder for "
<< rtc::ToString(spec.format);
return false;
}
// If other side does not support audio TWCC and WebRTC-Audio-ABWENoTWCC is
// not enabled, do not update target audio bitrate if we are in
// WebRTC-Audio-SendSideBwe-For-Video experiment
const bool do_not_update_target_bitrate =
!webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC") &&
webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe-For-Video") &&
!FindExtensionIds(new_config.rtp.extensions).transport_sequence_number;
// If a bitrate has been specified for the codec, use it over the
// codec's default.
if (!do_not_update_target_bitrate && 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_)) {
RTC_DLOG(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->StoreEncoderProperties(encoder->SampleRateHz(),
encoder->NumChannels());
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);
}
// If other side does not support audio TWCC and WebRTC-Audio-ABWENoTWCC is
// not enabled, do not update target audio bitrate if we are in
// WebRTC-Audio-SendSideBwe-For-Video experiment
const bool do_not_update_target_bitrate =
!webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC") &&
webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe-For-Video") &&
!FindExtensionIds(new_config.rtp.extensions).transport_sequence_number;
const absl::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 (!do_not_update_target_bitrate && 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_)) {
RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
<< new_config.rtp.ssrc;
} else {
RTC_NOTREACHED();
}
});
} else {
CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
encoder->DisableAudioNetworkAdaptor();
});
RTC_DLOG(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.
int new_transport_seq_num_id =
FindExtensionIds(new_config.rtp.extensions).transport_sequence_number;
if (stream->config_.min_bitrate_bps == new_config.min_bitrate_bps &&
stream->config_.max_bitrate_bps == new_config.max_bitrate_bps &&
stream->config_.bitrate_priority == new_config.bitrate_priority &&
(FindExtensionIds(stream->config_.rtp.extensions)
.transport_sequence_number == new_transport_seq_num_id ||
!webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
return;
}
bool has_transport_sequence_number = new_transport_seq_num_id != 0;
if (new_config.min_bitrate_bps != -1 && new_config.max_bitrate_bps != -1 &&
!new_config.has_dscp &&
(has_transport_sequence_number ||
!webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
stream->ConfigureBitrateObserver(
new_config.min_bitrate_bps, new_config.max_bitrate_bps,
new_config.bitrate_priority, has_transport_sequence_number);
stream->rtp_rtcp_module_->SetAsPartOfAllocation(true);
} else {
stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(false);
stream->RemoveBitrateObserver();
stream->rtp_rtcp_module_->SetAsPartOfAllocation(false);
}
}
void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps,
int max_bitrate_bps,
double bitrate_priority,
bool has_packet_feedback) {
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;
config_.bitrate_priority = bitrate_priority;
// This either updates the current observer or adds a new observer.
bitrate_allocator_->AddObserver(
this, MediaStreamAllocationConfig{
static_cast<uint32_t>(min_bitrate_bps),
static_cast<uint32_t>(max_bitrate_bps), 0, true,
config_.track_id, bitrate_priority, has_packet_feedback});
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) {
RTC_DLOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
"RTP/RTCP module";
}
}
}
} // namespace internal
} // namespace webrtc
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