1 参考链接
https://blog.csdn.net/sonysuqin/article/details/107297157
https://www.cnblogs.com/jiayayao/p/12649665.html
https://www.jianshu.com/p/3a4d24a71091?hmsr=toutiao.io&utm_medium=toutiao.io&utm_source=toutiao.io
2 相关代码
音视频同步的基本对象是AudioReceiveStream和VideoReceiveStream，两者都继承自Syncable；
2.1 同步前相关的初始化操作
namespace webrtc {
class Syncable {
public:
struct Info {
int64_t latest_receive_time_ms = 0;
uint32_t latest_received_capture_timestamp = 0;
uint32_t capture_time_ntp_secs = 0;
uint32_t capture_time_ntp_frac = 0;
uint32_t capture_time_source_clock = 0;
int current_delay_ms = 0;
};
virtual ~Syncable();
virtual int id() const = 0;
virtual absl::optional<Info> GetInfo() const = 0;
virtual uint32_t GetPlayoutTimestamp() const = 0;
virtual void SetMinimumPlayoutDelay(int delay_ms) = 0;
};
} // namespace webrtc
创建 VideoReceiveStream 并进行相应设置：
webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
webrtc::VideoReceiveStream::Config configuration) {
TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
receive_side_cc_.SetSendPeriodicFeedback(
SendPeriodicFeedback(configuration.rtp.extensions));
RegisterRateObserver();
VideoReceiveStream* receive_stream = new VideoReceiveStream(
task_queue_factory_, &video_receiver_controller_, num_cpu_cores_,
transport_send_ptr_->packet_router(), std::move(configuration),
module_process_thread_.get(), call_stats_.get(), clock_);
const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
{
WriteLockScoped write_lock(*receive_crit_);
if (config.rtp.rtx_ssrc) {
// We record identical config for the rtx stream as for the main
// stream. Since the transport_send_cc negotiation is per payload
// type, we may get an incorrect value for the rtx stream, but
// that is unlikely to matter in practice.
receive_rtp_config_.emplace(config.rtp.rtx_ssrc,
ReceiveRtpConfig(config));
}
receive_rtp_config_.emplace(config.rtp.remote_ssrc,
ReceiveRtpConfig(config));
video_receive_streams_.insert(receive_stream);
ConfigureSync(config.sync_group); // 音视频同步
}
receive_stream->SignalNetworkState(video_network_state_);
UpdateAggregateNetworkState();
event_log_->Log(std::make_unique<RtcEventVideoReceiveStreamConfig>(
CreateRtcLogStreamConfig(config)));
return receive_stream;
}
void Call::ConfigureSync(const std::string& sync_group) {
// Set sync only if there was no previous one.
if (sync_group.empty())
return;
AudioReceiveStream* sync_audio_stream = nullptr;
// Find existing audio stream.
const auto it = sync_stream_mapping_.find(sync_group);
if (it != sync_stream_mapping_.end()) {
sync_audio_stream = it->second;
} else {
// No configured audio stream, see if we can find one.
for (AudioReceiveStream* stream : audio_receive_streams_) {
if (stream->config().sync_group == sync_group) {
if (sync_audio_stream != nullptr) {
RTC_LOG(LS_WARNING)
<< "Attempting to sync more than one audio stream "
"within the same sync group. This is not "
"supported in the current implementation.";
break;
}
sync_audio_stream = stream;
}
}
}
if (sync_audio_stream)
sync_stream_mapping_[sync_group] = sync_audio_stream;
size_t num_synced_streams = 0;
for (VideoReceiveStream* video_stream : video_receive_streams_) {
if (video_stream->config().sync_group != sync_group)
continue;
++num_synced_streams;
if (num_synced_streams > 1) {
// TODO(pbos): Support synchronizing more than one A/V pair.
// https://code.google.com/p/webrtc/issues/detail?id=4762
RTC_LOG(LS_WARNING)
<< "Attempting to sync more than one audio/video pair "
"within the same sync group. This is not supported in "
"the current implementation.";
}
// Only sync the first A/V pair within this sync group.
if (num_synced_streams == 1) {
// sync_audio_stream may be null and that's ok.
video_stream->SetSync(sync_audio_stream); // 将 sync_audio_stream 和 video_stream 进行绑定以用来进行音视频的同步
} else {
video_stream->SetSync(nullptr);
}
}
}
void VideoReceiveStream::SetSync(Syncable* audio_syncable) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
rtp_stream_sync_.ConfigureSync(audio_syncable); // RtpStreamsSynchronizer::ConfigureSync
}
VideoReceiveStream::VideoReceiveStream(
TaskQueueFactory* task_queue_factory,
RtpStreamReceiverControllerInterface* receiver_controller,
int num_cpu_cores,
PacketRouter* packet_router,
VideoReceiveStream::Config config,
ProcessThread* process_thread, // process_thread 就是在 Call* Call::Create(const Call::Config& config) 中创建的 ProcessThread::Create("ModuleProcessThread")
CallStats* call_stats,
Clock* clock,
VCMTiming* timing)
: task_queue_factory_(task_queue_factory),
transport_adapter_(config.rtcp_send_transport),
config_(std::move(config)),
num_cpu_cores_(num_cpu_cores),
process_thread_(process_thread), //
clock_(clock),
call_stats_(call_stats),
source_tracker_(clock_),
stats_proxy_(&config_, clock_),
rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
timing_(timing),
video_receiver_(clock_, timing_.get()),
rtp_video_stream_receiver_(clock_,
&transport_adapter_,
call_stats,
packet_router,
&config_,
rtp_receive_statistics_.get(),
&stats_proxy_,
process_thread_,
this, // NackSender
nullptr, // Use default KeyFrameRequestSender
this, // OnCompleteFrameCallback
config_.frame_decryptor),
rtp_stream_sync_(this), // 将 webrtc::internal::VideoReceiveStream 的 this 指针传递进去，也就是 RtpStreamsSynchronizer 中的 syncable_video_ 实际指向的就是 VideoReceiveStream
process_thread_->RegisterModule(&rtp_stream_sync_, RTC_FROM_HERE); // 将 RtpStreamsSynchronizer rtp_stream_sync_ 的地址注册到处理线程中
void RtpStreamsSynchronizer::ConfigureSync(Syncable* syncable_audio) {
rtc::CritScope lock(&crit_);
if (syncable_audio == syncable_audio_) {
// This prevents expensive no-ops.
return;
}
syncable_audio_ = syncable_audio; // syncable_audio_ 实际上指向的是 webrtc::internal::AudioReceiveStream
sync_.reset(nullptr);
if (syncable_audio_) {
sync_.reset(new StreamSynchronization(syncable_video_->id(), //
syncable_audio_->id()));
}
}
创建 AudioReceiveStream 并进行相应设置：
webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RegisterRateObserver();
event_log_->Log(std::make_unique<RtcEventAudioReceiveStreamConfig>(
CreateRtcLogStreamConfig(config)));
AudioReceiveStream* receive_stream = new AudioReceiveStream( //
clock_, &audio_receiver_controller_, transport_send_ptr_->packet_router(), // RtpStreamReceiverController audio_receiver_controller_;
module_process_thread_.get(), config, config_.audio_state, event_log_);// config_.audio_stat
{
WriteLockScoped write_lock(*receive_crit_);
receive_rtp_config_.emplace(config.rtp.remote_ssrc,
ReceiveRtpConfig(config));
audio_receive_streams_.insert(receive_stream);
ConfigureSync(config.sync_group); // 音视频同步
}
{
ReadLockScoped read_lock(*send_crit_);
auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc);
if (it != audio_send_ssrcs_.end()) {
receive_stream->AssociateSendStream(it->second);
}
}
UpdateAggregateNetworkState();
return receive_stream;
}
AudioReceiveStream::AudioReceiveStream(
Clock* clock,
RtpStreamReceiverControllerInterface* receiver_controller, // receiver_controller 就是 webrtc::internal::Call 中 audio_receiver_controller_的 地址
PacketRouter* packet_router,
ProcessThread* module_process_thread,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state, //
webrtc::RtcEventLog* event_log)
: AudioReceiveStream(clock,
receiver_controller,
packet_router,
config,
audio_state,
event_log,
CreateChannelReceive(clock, //
audio_state.get(),
module_process_thread,
config,
event_log)) {}
// 缩进表示内部的创建细节
std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
Clock* clock,
webrtc::AudioState* audio_state,
ProcessThread* module_process_thread,
const webrtc::AudioReceiveStream::Config& config,
RtcEventLog* event_log) {
RTC_DCHECK(audio_state);
internal::AudioState* internal_audio_state =
static_cast<internal::AudioState*>(audio_state);
return voe::CreateChannelReceive( //
clock, module_process_thread, internal_audio_state->audio_device_module(),
config.media_transport_config, config.rtcp_send_transport, event_log,// config.rtcp_send_transport 就是 WebRtcVoiceMediaChannel 的 this 指针，参考 WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream
config.rtp.local_ssrc, config.rtp.remote_ssrc,
config.jitter_buffer_max_packets, config.jitter_buffer_fast_accelerate,
config.jitter_buffer_min_delay_ms,
config.jitter_buffer_enable_rtx_handling, config.decoder_factory, // config.decoder_factory 就是 CreateBuiltinAudioDecoderFactory() 的返回值
config.codec_pair_id, config.frame_decryptor, config.crypto_options);
}
std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport, //
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory, // decoder_factory 就是 CreateBuiltinAudioDecoderFactory() 的返回值
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options) {
return std::make_unique<ChannelReceive>( //
clock, module_process_thread, audio_device_module, media_transport_config,
rtcp_send_transport, rtc_event_log, local_ssrc, remote_ssrc, //
jitter_buffer_max_packets, jitter_buffer_fast_playout,
jitter_buffer_min_delay_ms, jitter_buffer_enable_rtx_handling,
decoder_factory, codec_pair_id, frame_decryptor, crypto_options); // decoder_factory 就是 CreateBuiltinAudioDecoderFactory() 的返回值
}
// webrtc::voe::ChannelReceive
// class ChannelReceiveInterface : public RtpPacketSinkInterface
// class ChannelReceive : public ChannelReceiveInterface,
public MediaTransportAudioSinkInterface
===>
acm2::AcmReceiver acm_receiver_;//
ChannelReceive::ChannelReceive(
Clock* clock,
ProcessThread* module_process_thread,
AudioDeviceModule* audio_device_module,
const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport, //
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,// decoder_factory 就是 CreateBuiltinAudioDecoderFactory() 的返回值
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options)
: event_log_(rtc_event_log),
rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
remote_ssrc_(remote_ssrc),
acm_receiver_(AcmConfig(decoder_factory, // decoder_factory 就是 CreateBuiltinAudioDecoderFactory() 的返回值
codec_pair_id,
jitter_buffer_max_packets,
jitter_buffer_fast_playout)),
_outputAudioLevel(),
ntp_estimator_(clock),
playout_timestamp_rtp_(0),
playout_delay_ms_(0),
rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
capture_start_rtp_time_stamp_(-1),
capture_start_ntp_time_ms_(-1),
_moduleProcessThreadPtr(module_process_thread),
_audioDeviceModulePtr(audio_device_module),
_outputGain(1.0f),
associated_send_channel_(nullptr),
media_transport_config_(media_transport_config),
frame_decryptor_(frame_decryptor),
crypto_options_(crypto_options) {
// TODO(nisse): Use _moduleProcessThreadPtr instead?
module_process_thread_checker_.Detach();
RTC_DCHECK(module_process_thread);
RTC_DCHECK(audio_device_module);
acm_receiver_.ResetInitialDelay();
acm_receiver_.SetMinimumDelay(0);
acm_receiver_.SetMaximumDelay(0);
acm_receiver_.FlushBuffers();
_outputAudioLevel.ResetLevelFullRange();
rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc_, true);
RtpRtcp::Configuration configuration;
configuration.clock = clock;
configuration.audio = true;
configuration.receiver_only = true;
configuration.outgoing_transport = rtcp_send_transport;
configuration.receive_statistics = rtp_receive_statistics_.get();
configuration.event_log = event_log_;
configuration.local_media_ssrc = local_ssrc;
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);
_rtpRtcpModule->SetRemoteSSRC(remote_ssrc_);
_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE);
// Ensure that RTCP is enabled for the created channel.
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
if (media_transport()) {
media_transport()->SetReceiveAudioSink(this);
}
}
AudioCodingModule::Config AcmConfig(
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory, // decoder_factory 就是 CreateBuiltinAudioDecoderFactory() 的返回值
absl::optional<AudioCodecPairId> codec_pair_id,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout) {
AudioCodingModule::Config acm_config;
acm_config.decoder_factory = decoder_factory; // decoder_factory 就是 CreateBuiltinAudioDecoderFactory() 的返回值
acm_config.neteq_config.codec_pair_id = codec_pair_id;
acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets;
acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout;
acm_config.neteq_config.enable_muted_state = true;
return acm_config;
}
AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
: last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
neteq_(NetEq::Create(config.neteq_config,
config.clock,
config.decoder_factory)), // config.decoder_factory 就是 CreateBuiltinAudioDecoderFactory() 的返回值
clock_(config.clock),
resampled_last_output_frame_(true) {
RTC_DCHECK(clock_);
memset(last_audio_buffer_.get(), 0,
sizeof(int16_t) * AudioFrame::kMaxDataSizeSamples);
}
NetEq* NetEq::Create(
const NetEq::Config& config,
Clock* clock,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) { // decoder_factory 就是 CreateBuiltinAudioDecoderFactory() 的返回值
return new NetEqImpl(config, //
NetEqImpl::Dependencies(config, clock, decoder_factory));
}
NetEqImpl::Dependencies::Dependencies(
const NetEq::Config& config,
Clock* clock,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
: clock(clock),
tick_timer(new TickTimer),
stats(new StatisticsCalculator),
buffer_level_filter(new BufferLevelFilter),
decoder_database(
new DecoderDatabase(decoder_factory, config.codec_pair_id)), //
delay_peak_detector(
new DelayPeakDetector(tick_timer.get(), config.enable_rtx_handling)),
delay_manager(DelayManager::Create(config.max_packets_in_buffer,
config.min_delay_ms,
config.enable_rtx_handling,
delay_peak_detector.get(),
tick_timer.get(),
stats.get())),
dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),
dtmf_tone_generator(new DtmfToneGenerator),
packet_buffer(
new PacketBuffer(config.max_packets_in_buffer, tick_timer.get())),
red_payload_splitter(new RedPayloadSplitter),
timestamp_scaler(new TimestampScaler(*decoder_database)),
accelerate_factory(new AccelerateFactory),
expand_factory(new ExpandFactory),
preemptive_expand_factory(new PreemptiveExpandFactory) {}
AudioReceiveStream::AudioReceiveStream(
Clock* clock,
RtpStreamReceiverControllerInterface* receiver_controller,//
PacketRouter* packet_router,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state, //
webrtc::RtcEventLog* event_log,
std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)// channel_receive 实际指向的是 webrtc::voe::ChannelReceive
: audio_state_(audio_state),
channel_receive_(std::move(channel_receive)), // move 后 channel_receive_ 实际上指向的也是 webrtc::voe::ChannelReceive
source_tracker_(clock) {
RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc;
RTC_DCHECK(config.decoder_factory);
RTC_DCHECK(config.rtcp_send_transport);
RTC_DCHECK(audio_state_);
RTC_DCHECK(channel_receive_);
module_process_thread_checker_.Detach();
if (!config.media_transport_config.media_transport) {
RTC_DCHECK(receiver_controller);
RTC_DCHECK(packet_router);
// Configure bandwidth estimation.
channel_receive_->RegisterReceiverCongestionControlObjects(packet_router); //
// Register with transport.
rtp_stream_receiver_ = receiver_controller->CreateReceiver( //RtpStreamReceiverController::CreateReceiver
config.rtp.remote_ssrc, channel_receive_.get()); // channel_receive_ 实际上指向的也是 webrtc::voe::ChannelReceive
} // rtp_stream_receiver_ 指向的是 webrtc::RtpStreamReceiverController::Receiver
ConfigureStream(this, config, true); //
}
// webrtc::voe::ChannelReceive
// class ChannelReceiveInterface : public RtpPacketSinkInterface
// class ChannelReceive : public ChannelReceiveInterface,
public MediaTransportAudioSinkInterface
// class webrtc::RtpStreamReceiverController::Receiver : public RtpStreamReceiverInterface
std::unique_ptr<RtpStreamReceiverInterface>
RtpStreamReceiverController::CreateReceiver(uint32_t ssrc,
RtpPacketSinkInterface* sink) {
return std::make_unique<Receiver>(this, ssrc, sink); // webrtc::RtpStreamReceiverController::Receiver
}
在名称为 ModuleProcessThread 的线程中进行音视频的同步:
std::unique_ptr<ProcessThread> ProcessThread::Create(const char* thread_name) {
return std::unique_ptr<ProcessThread>(new ProcessThreadImpl(thread_name));
}
ProcessThreadImpl::ProcessThreadImpl(const char* thread_name)
: stop_(false), thread_name_(thread_name) {}
class ProcessThreadImpl : public ProcessThread {
public:
explicit ProcessThreadImpl(const char* thread_name);
~ProcessThreadImpl() override;
void Start() override;
void Stop() override;
void WakeUp(Module* module) override;
void PostTask(std::unique_ptr<QueuedTask> task) override;
void RegisterModule(Module* module, const rtc::Location& from) override;
void DeRegisterModule(Module* module) override;
protected:
static void Run(void* obj);
bool Process();
private:
struct ModuleCallback {
ModuleCallback() = delete;
ModuleCallback(ModuleCallback&& cb) = default;
ModuleCallback(const ModuleCallback& cb) = default;
ModuleCallback(Module* module, const rtc::Location& location)
: module(module), location(location) {}
bool operator==(const ModuleCallback& cb) const {
return cb.module == module;
}
Module* const module;
int64_t next_callback = 0; // Absolute timestamp.
const rtc::Location location;
private:
ModuleCallback& operator=(ModuleCallback&);
};
typedef std::list<ModuleCallback> ModuleList;
// Warning: For some reason, if |lock_| comes immediately before |modules_|
// with the current class layout, we will start to have mysterious crashes
// on Mac 10.9 debug. I (Tommi) suspect we're hitting some obscure alignemnt
// issues, but I haven't figured out what they are, if there are alignment
// requirements for mutexes on Mac or if there's something else to it.
// So be careful with changing the layout.
rtc::CriticalSection lock_; // Used to guard modules_, tasks_ and stop_.
rtc::ThreadChecker thread_checker_;
rtc::Event wake_up_;
// TODO(pbos): Remove unique_ptr and stop recreating the thread.
std::unique_ptr<rtc::PlatformThread> thread_; // 平台相关的线程
ModuleList modules_;
std::queue<QueuedTask*> queue_;
bool stop_;
const char* thread_name_;
};
void ProcessThreadImpl::Start() {
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!thread_.get());
if (thread_.get())
return;
RTC_DCHECK(!stop_);
for (ModuleCallback& m : modules_)
m.module->ProcessThreadAttached(this);
thread_.reset(
new rtc::PlatformThread(&ProcessThreadImpl::Run, this, thread_name_));
thread_->Start();
}
void ProcessThreadImpl::Run(void* obj) {
ProcessThreadImpl* impl = static_cast<ProcessThreadImpl*>(obj);
while (impl->Process()) {
}
}
bool ProcessThreadImpl::Process() {
TRACE_EVENT1("webrtc", "ProcessThreadImpl", "name", thread_name_);
int64_t now = rtc::TimeMillis();
int64_t next_checkpoint = now + (1000 * 60);
{
rtc::CritScope lock(&lock_);
if (stop_)
return false;
for (ModuleCallback& m : modules_) {
// TODO(tommi): Would be good to measure the time TimeUntilNextProcess
// takes and dcheck if it takes too long (e.g. >=10ms). Ideally this
// operation should not require taking a lock, so querying all modules
// should run in a matter of nanoseconds.
if (m.next_callback == 0)
m.next_callback = GetNextCallbackTime(m.module, now);
if (m.next_callback <= now ||
m.next_callback == kCallProcessImmediately) {
{
TRACE_EVENT2("webrtc", "ModuleProcess", "function",
m.location.function_name(), "file",
m.location.file_and_line());
m.module->Process(); // 上面的设置：process_thread_->RegisterModule(&rtp_stream_sync_, RTC_FROM_HERE)，rtp_stream_sync_ 是 RtpStreamsSynchronizer 类型
} // 所以这里的会执行 RtpStreamsSynchronizer::Process
// Use a new 'now' reference to calculate when the next callback
// should occur. We'll continue to use 'now' above for the baseline
// of calculating how long we should wait, to reduce variance.
int64_t new_now = rtc::TimeMillis();
m.next_callback = GetNextCallbackTime(m.module, new_now);
}
if (m.next_callback < next_checkpoint)
next_checkpoint = m.next_callback;
}
while (!queue_.empty()) {
QueuedTask* task = queue_.front();
queue_.pop();
lock_.Leave();
task->Run();
delete task;
lock_.Enter();
}
}
int64_t time_to_wait = next_checkpoint - rtc::TimeMillis();
if (time_to_wait > 0)
wake_up_.Wait(static_cast<int>(time_to_wait));
return true;
}
void RtpStreamsSynchronizer::Process() {
RTC_DCHECK_RUN_ON(&process_thread_checker_);
last_sync_time_ = rtc::TimeNanos();
rtc::CritScope lock(&crit_);
if (!syncable_audio_) {
return;
}
RTC_DCHECK(sync_.get());
absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo(); // AudioReceiveStream::GetInfo
if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
return;
}
int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
absl::optional<Syncable::Info> video_info = syncable_video_->GetInfo(); // VideoReceiveStream::GetInfo
if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
return;
}
if (last_video_receive_ms == video_measurement_.latest_receive_time_ms) {
// No new video packet has been received since last update.
return;
}
int relative_delay_ms;
// Calculate how much later or earlier the audio stream is compared to video.
if (!sync_->ComputeRelativeDelay(audio_measurement_, video_measurement_, // StreamSynchronization::ComputeRelativeDelay
&relative_delay_ms)) {
return;
}
TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay",
video_info->current_delay_ms);
TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay",
audio_info->current_delay_ms);
TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
int target_audio_delay_ms = 0;
int target_video_delay_ms = video_info->current_delay_ms;
// Calculate the necessary extra audio delay and desired total video
// delay to get the streams in sync.
if (!sync_->ComputeDelays(relative_delay_ms, audio_info->current_delay_ms, // StreamSynchronization::ComputeDelays
&target_audio_delay_ms, &target_video_delay_ms)) {
return;
}
syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms); // AudioReceiveStream::SetMinimumPlayoutDelay
syncable_video_->SetMinimumPlayoutDelay(target_video_delay_ms); // VideoReceiveStream::SetMinimumPlayoutDelay
}
// 计算最近一对音视频包的相对延迟
bool StreamSynchronization::ComputeRelativeDelay(
const Measurements& audio_measurement,
const Measurements& video_measurement,
int* relative_delay_ms) {
assert(relative_delay_ms);
int64_t audio_last_capture_time_ms;
if (!audio_measurement.rtp_to_ntp.Estimate(audio_measurement.latest_timestamp,
&audio_last_capture_time_ms)) {
return false;
}
int64_t video_last_capture_time_ms;
if (!video_measurement.rtp_to_ntp.Estimate(video_measurement.latest_timestamp,
&video_last_capture_time_ms)) {
return false;
}
if (video_last_capture_time_ms < 0) {
return false;
}
// Positive diff means that video_measurement is behind audio_measurement.
*relative_delay_ms =
video_measurement.latest_receive_time_ms -
audio_measurement.latest_receive_time_ms -
(video_last_capture_time_ms - audio_last_capture_time_ms);
if (*relative_delay_ms > kMaxDeltaDelayMs ||
*relative_delay_ms < -kMaxDeltaDelayMs) {
return false;
}
return true;
}
// 音视频同步的具体逻辑
// 参考链接： https://blog.csdn.net/sonysuqin/article/details/107297157
bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
int current_audio_delay_ms,
int* total_audio_delay_target_ms,
int* total_video_delay_target_ms) {
assert(total_audio_delay_target_ms && total_video_delay_target_ms);
int current_video_delay_ms = *total_video_delay_target_ms;
RTC_LOG(LS_VERBOSE) << "Audio delay: " << current_audio_delay_ms
<< " current diff: " << relative_delay_ms
<< " for stream " << audio_stream_id_;
// Calculate the difference between the lowest possible video delay and
// the current audio delay.
int current_diff_ms =
current_video_delay_ms - current_audio_delay_ms + relative_delay_ms;
avg_diff_ms_ =
((kFilterLength - 1) * avg_diff_ms_ + current_diff_ms) / kFilterLength;
if (abs(avg_diff_ms_) < kMinDeltaMs) {
// Don't adjust if the diff is within our margin.
return false;
}
// Make sure we don't move too fast.
int diff_ms = avg_diff_ms_ / 2;
diff_ms = std::min(diff_ms, kMaxChangeMs);
diff_ms = std::max(diff_ms, -kMaxChangeMs);
// Reset the average after a move to prevent overshooting reaction.
avg_diff_ms_ = 0;
if (diff_ms > 0) {
// The minimum video delay is longer than the current audio delay.
// We need to decrease extra video delay, or add extra audio delay.
if (channel_delay_.extra_video_delay_ms > base_target_delay_ms_) {
// We have extra delay added to ViE. Reduce this delay before adding
// extra delay to VoE.
channel_delay_.extra_video_delay_ms -= diff_ms;
channel_delay_.extra_audio_delay_ms = base_target_delay_ms_;
} else { // channel_delay_.extra_video_delay_ms > 0
// We have no extra video delay to remove, increase the audio delay.
channel_delay_.extra_audio_delay_ms += diff_ms;
channel_delay_.extra_video_delay_ms = base_target_delay_ms_;
}
} else { // if (diff_ms > 0)
// The video delay is lower than the current audio delay.
// We need to decrease extra audio delay, or add extra video delay.
if (channel_delay_.extra_audio_delay_ms > base_target_delay_ms_) {
// We have extra delay in VoiceEngine.
// Start with decreasing the voice delay.
// Note: diff_ms is negative; add the negative difference.
channel_delay_.extra_audio_delay_ms += diff_ms;
channel_delay_.extra_video_delay_ms = base_target_delay_ms_;
} else { // channel_delay_.extra_audio_delay_ms > base_target_delay_ms_
// We have no extra delay in VoiceEngine, increase the video delay.
// Note: diff_ms is negative; subtract the negative difference.
channel_delay_.extra_video_delay_ms -= diff_ms; // X - (-Y) = X + Y.
channel_delay_.extra_audio_delay_ms = base_target_delay_ms_;
}
}
// Make sure that video is never below our target.
channel_delay_.extra_video_delay_ms =
std::max(channel_delay_.extra_video_delay_ms, base_target_delay_ms_);
int new_video_delay_ms;
if (channel_delay_.extra_video_delay_ms > base_target_delay_ms_) {
new_video_delay_ms = channel_delay_.extra_video_delay_ms;
} else {
// No change to the extra video delay. We are changing audio and we only
// allow to change one at the time.
new_video_delay_ms = channel_delay_.last_video_delay_ms;
}
// Make sure that we don't go below the extra video delay.
new_video_delay_ms =
std::max(new_video_delay_ms, channel_delay_.extra_video_delay_ms);
// Verify we don't go above the maximum allowed video delay.
new_video_delay_ms =
std::min(new_video_delay_ms, base_target_delay_ms_ + kMaxDeltaDelayMs);
int new_audio_delay_ms;
if (channel_delay_.extra_audio_delay_ms > base_target_delay_ms_) {
new_audio_delay_ms = channel_delay_.extra_audio_delay_ms;
} else {
// No change to the audio delay. We are changing video and we only
// allow to change one at the time.
new_audio_delay_ms = channel_delay_.last_audio_delay_ms;
}
// Make sure that we don't go below the extra audio delay.
new_audio_delay_ms =
std::max(new_audio_delay_ms, channel_delay_.extra_audio_delay_ms);
// Verify we don't go above the maximum allowed audio delay.
new_audio_delay_ms =
std::min(new_audio_delay_ms, base_target_delay_ms_ + kMaxDeltaDelayMs);
// Remember our last audio and video delays.
channel_delay_.last_video_delay_ms = new_video_delay_ms;
channel_delay_.last_audio_delay_ms = new_audio_delay_ms;
RTC_LOG(LS_VERBOSE) << "Sync video delay " << new_video_delay_ms
<< " for video stream " << video_stream_id_
<< " and audio delay "
<< channel_delay_.extra_audio_delay_ms
<< " for audio stream " << audio_stream_id_;
// Return values.
*total_video_delay_target_ms = new_video_delay_ms;
*total_audio_delay_target_ms = new_audio_delay_ms;
return true;
}3 AudioReceiveStream::GetInfo
absl::optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
absl::optional<Syncable::Info> info = channel_receive_->GetSyncInfo(); // ChannelReceive::GetSyncInfo
if (!info)
return absl::nullopt;
info->current_delay_ms = channel_receive_->GetDelayEstimate(); // ChannelReceive::GetDelayEstimate
return info;
}
ChannelReceive::ChannelReceive(
Clock* clock,
ProcessThread* module_process_thread, // 就是在 Call::Create 中通过 ProcessThread::Create("ModuleProcessThread") 创建的 thread
AudioDeviceModule* audio_device_module,
const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
uint32_t remote_ssrc,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout,
int jitter_buffer_min_delay_ms,
bool jitter_buffer_enable_rtx_handling,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options)
: event_log_(rtc_event_log),
rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
remote_ssrc_(remote_ssrc),
acm_receiver_(AcmConfig(decoder_factory,
codec_pair_id,
jitter_buffer_max_packets,
jitter_buffer_fast_playout)),
_outputAudioLevel(),
ntp_estimator_(clock),
playout_timestamp_rtp_(0),
playout_delay_ms_(0),
rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
capture_start_rtp_time_stamp_(-1),
capture_start_ntp_time_ms_(-1),
_moduleProcessThreadPtr(module_process_thread), //module_process_thread 就是在 Call::Create 中通过 ProcessThread::Create("ModuleProcessThread") 创建的 thread
_audioDeviceModulePtr(audio_device_module),
_outputGain(1.0f),
associated_send_channel_(nullptr),
media_transport_config_(media_transport_config),
frame_decryptor_(frame_decryptor),
crypto_options_(crypto_options) {
// TODO(nisse): Use _moduleProcessThreadPtr instead?
module_process_thread_checker_.Detach();
RTC_DCHECK(module_process_thread);
RTC_DCHECK(audio_device_module);
acm_receiver_.ResetInitialDelay();
acm_receiver_.SetMinimumDelay(0);
acm_receiver_.SetMaximumDelay(0);
acm_receiver_.FlushBuffers();
_outputAudioLevel.ResetLevelFullRange();
rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc_, true);
RtpRtcp::Configuration configuration; //
configuration.clock = clock;
configuration.audio = true;
configuration.receiver_only = true;
configuration.outgoing_transport = rtcp_send_transport;
configuration.receive_statistics = rtp_receive_statistics_.get();
configuration.event_log = event_log_;
configuration.local_media_ssrc = local_ssrc;
_rtpRtcpModule = RtpRtcp::Create(configuration); // RtpRtcp::Create 返回的是 webrtc::ModuleRtpRtcpImpl
_rtpRtcpModule->SetSendingMediaStatus(false);
_rtpRtcpModule->SetRemoteSSRC(remote_ssrc_);
// _moduleProcessThreadPtr 就是在 Call::Create 中通过 ProcessThread::Create("ModuleProcessThread") 创建的 thread
_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE);// 注册 module ，这样就会在 ModuleProcessThread 线程中运行 ModuleRtpRtcpImpl::Process
// Ensure that RTCP is enabled for the created channel.
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
if (media_transport()) {
media_transport()->SetReceiveAudioSink(this);
}
}
std::unique_ptr<RtpRtcp> RtpRtcp::Create(const Configuration& configuration) {
RTC_DCHECK(configuration.clock);
return std::make_unique<ModuleRtpRtcpImpl>(configuration);
}
absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
RTC_DCHECK(module_process_thread_checker_.IsCurrent());
Syncable::Info info;
if (_rtpRtcpModule->RemoteNTP(&info.capture_time_ntp_secs, // webrtc::ModuleRtpRtcpImpl::RemoteNTP
&info.capture_time_ntp_frac, nullptr, nullptr,
&info.capture_time_source_clock) != 0) {
return absl::nullopt;
}
{
// last_received_rtp_timestamp_ 最近一次收到的RTP包中的时间戳，在 ChannelReceive::OnRtpPacket 中更新
// last_received_rtp_system_time_ms_ 最近一次收到RTP包时对应的本地时间，单位为ms，在 ChannelReceive::OnRtpPacket 中更新
rtc::CritScope cs(&sync_info_lock_);
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
return absl::nullopt;
}
info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
}
return info;//
}
int32_t ModuleRtpRtcpImpl::RemoteNTP(uint32_t* received_ntpsecs,
uint32_t* received_ntpfrac,
uint32_t* rtcp_arrival_time_secs,
uint32_t* rtcp_arrival_time_frac,
uint32_t* rtcp_timestamp) const {
return rtcp_receiver_.NTP(received_ntpsecs, received_ntpfrac, // RTCPReceiver::NTP
rtcp_arrival_time_secs, rtcp_arrival_time_frac,
rtcp_timestamp)
? 0
: -1;
}
// received_ntp_secs 用来保存最近一次接收到的SR包中的NTP时间中的单位为秒的部分
// received_ntp_frac 用来保存最近一次接收到的SR包中的NTP时间中的单位为1/2^32秒的部分
// rtcp_arrival_time_secs 用来保存接收到这个SR包的时候，本地时间的NTP表示中的单位为秒的部分
// rtcp_arrival_time_frac 用来保存接收到这个SR包的时候，本地时间的NTP表示中的单位为1/2^32秒的部分
// rtcp_timestamp 用来保存最近一次接收到的SR包中的 RTP 时间
bool RTCPReceiver::NTP(uint32_t* received_ntp_secs,
uint32_t* received_ntp_frac,
uint32_t* rtcp_arrival_time_secs,
uint32_t* rtcp_arrival_time_frac,
uint32_t* rtcp_timestamp) const {
rtc::CritScope lock(&rtcp_receiver_lock_);
if (!last_received_sr_ntp_.Valid())
return false;
// NTP from incoming SenderReport.
if (received_ntp_secs)
*received_ntp_secs = remote_sender_ntp_time_.seconds();
if (received_ntp_frac)
*received_ntp_frac = remote_sender_ntp_time_.fractions();
// Rtp time from incoming SenderReport.
if (rtcp_timestamp)
*rtcp_timestamp = remote_sender_rtp_time_;
// Local NTP time when we received a RTCP packet with a send block.
if (rtcp_arrival_time_secs)
*rtcp_arrival_time_secs = last_received_sr_ntp_.seconds();
if (rtcp_arrival_time_frac)
*rtcp_arrival_time_frac = last_received_sr_ntp_.fractions();
return true;
}
uint32_t ChannelReceive::GetDelayEstimate() const {
RTC_DCHECK(worker_thread_checker_.IsCurrent() ||
module_process_thread_checker_.IsCurrent());
rtc::CritScope lock(&video_sync_lock_);
return acm_receiver_.FilteredCurrentDelayMs() + playout_delay_ms_; // AcmReceiver::FilteredCurrentDelayMs
}
int AcmReceiver::FilteredCurrentDelayMs() const {
return neteq_->FilteredCurrentDelayMs(); // NetEqImpl::FilteredCurrentDelayMs
}
int NetEqImpl::FilteredCurrentDelayMs() const {
rtc::CritScope lock(&crit_sect_);
// Sum up the filtered packet buffer level with the future length of the sync
// buffer.
const int delay_samples = buffer_level_filter_->filtered_current_level() +
sync_buffer_->FutureLength();
// The division below will truncate. The return value is in ms.
return delay_samples / rtc::CheckedDivExact(fs_hz_, 1000);
}
//更新度量信息
RtpStreamsSynchronizer::Process
===>
UpdateMeasurements(&audio_measurement_, *audio_info)
bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
const Syncable::Info& info) {
RTC_DCHECK(stream);
stream->latest_timestamp = info.latest_received_capture_timestamp;
stream->latest_receive_time_ms = info.latest_receive_time_ms;
bool new_rtcp_sr = false;
if (!stream->rtp_to_ntp.UpdateMeasurements( // RtpToNtpEstimator::UpdateMeasurements
info.capture_time_ntp_secs, info.capture_time_ntp_frac,
info.capture_time_source_clock, &new_rtcp_sr)) {
return false;
}
return true;
}
// 由上面的调用可知：
// ntp_secs: 从接收端观测到的，最近一次接收到的SR包中的NTP时间中的单位为秒的部分
// ntp_frac: 从接收端观测到的，最近一次接收到的SR包中的NTP时间中的单位为1/2^32秒的部分
// rtp_timestamp: 从接收端观测到的，最近一次接收到的SR包中的 RTP 时间戳
bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
uint32_t ntp_frac,
uint32_t rtp_timestamp,
bool* new_rtcp_sr) {
*new_rtcp_sr = false;
int64_t unwrapped_rtp_timestamp = unwrapper_.Unwrap(rtp_timestamp);
RtcpMeasurement new_measurement(ntp_secs, ntp_frac, unwrapped_rtp_timestamp);
if (Contains(measurements_, new_measurement)) {
// RTCP SR report already added.
return true;
}
if (!new_measurement.ntp_time.Valid())
return false;
int64_t ntp_ms_new = new_measurement.ntp_time.ToMs();
bool invalid_sample = false;
if (!measurements_.empty()) {
int64_t old_rtp_timestamp = measurements_.front().unwrapped_rtp_timestamp;
int64_t old_ntp_ms = measurements_.front().ntp_time.ToMs();
if (ntp_ms_new <= old_ntp_ms ||
ntp_ms_new > old_ntp_ms + kMaxAllowedRtcpNtpIntervalMs) {
invalid_sample = true;
} else if (unwrapped_rtp_timestamp <= old_rtp_timestamp) {
RTC_LOG(LS_WARNING)
<< "Newer RTCP SR report with older RTP timestamp, dropping";
invalid_sample = true;
} else if (unwrapped_rtp_timestamp - old_rtp_timestamp > (1 << 25)) {
// Sanity check. No jumps too far into the future in rtp.
invalid_sample = true;
}
}
if (invalid_sample) {
++consecutive_invalid_samples_;
if (consecutive_invalid_samples_ < kMaxInvalidSamples) {
return false;
}
RTC_LOG(LS_WARNING) << "Multiple consecutively invalid RTCP SR reports, "
"clearing measurements.";
measurements_.clear();
params_ = absl::nullopt;
}
consecutive_invalid_samples_ = 0;
// Insert new RTCP SR report.
if (measurements_.size() == kNumRtcpReportsToUse)
measurements_.pop_back();
measurements_.push_front(new_measurement);
*new_rtcp_sr = true;
// List updated, calculate new parameters.
UpdateParameters();
return true;
}
4 VideoReceiveStream::GetInfo
absl::optional<Syncable::Info> VideoReceiveStream::GetInfo() const {
RTC_DCHECK_RUN_ON(&module_process_sequence_checker_);
absl::optional<Syncable::Info> info =
rtp_video_stream_receiver_.GetSyncInfo(); // RtpVideoStreamReceiver::GetSyncInfo
if (!info)
return absl::nullopt;
info->current_delay_ms = timing_->TargetVideoDelay(); // VCMTiming::TargetVideoDelay
return info;
}
absl::optional<Syncable::Info> RtpVideoStreamReceiver::GetSyncInfo() const {
Syncable::Info info;
if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs, // ModuleRtpRtcpImpl::RemoteNTP
&info.capture_time_ntp_frac, nullptr, nullptr,
&info.capture_time_source_clock) != 0) {
return absl::nullopt;
}
{
// last_received_rtp_timestamp_ 最近一次收到的RTP包中的时间戳，在 RtpVideoStreamReceiver::OnRtpPacket 中更新
// last_received_rtp_system_time_ms_ 最近一次收到RTP包时对应的本地时间，单位为ms，在 RtpVideoStreamReceiver::OnRtpPacket
rtc::CritScope lock(&sync_info_lock_);
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
return absl::nullopt;
}
info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
}
// Leaves info.current_delay_ms uninitialized.
return info;
}
int32_t ModuleRtpRtcpImpl::RemoteNTP(uint32_t* received_ntpsecs,
uint32_t* received_ntpfrac,
uint32_t* rtcp_arrival_time_secs,
uint32_t* rtcp_arrival_time_frac,
uint32_t* rtcp_timestamp) const {
return rtcp_receiver_.NTP(received_ntpsecs, received_ntpfrac, // RTCPReceiver::NTP
rtcp_arrival_time_secs, rtcp_arrival_time_frac,
rtcp_timestamp)
? 0
: -1;
}
// received_ntp_secs 用来保存最近一次接收到的SR包中的NTP时间中的单位为秒的部分
// received_ntp_frac 用来保存最近一次接收到的SR包中的NTP时间中的单位为1/2^32秒的部分
// rtcp_arrival_time_secs 用来保存接收到这个SR包的时候，本地时间的NTP表示中的单位为秒的部分
// rtcp_arrival_time_frac 用来保存接收到这个SR包的时候，本地时间的NTP表示中的单位为1/2^32秒的部分
// rtcp_timestamp 用来保存最近一次接收到的SR包中的 RTP 时间
bool RTCPReceiver::NTP(uint32_t* received_ntp_secs,
uint32_t* received_ntp_frac,
uint32_t* rtcp_arrival_time_secs,
uint32_t* rtcp_arrival_time_frac,
uint32_t* rtcp_timestamp) const {
rtc::CritScope lock(&rtcp_receiver_lock_);
if (!last_received_sr_ntp_.Valid())
return false;
// NTP from incoming SenderReport.
if (received_ntp_secs)
*received_ntp_secs = remote_sender_ntp_time_.seconds();
if (received_ntp_frac)
*received_ntp_frac = remote_sender_ntp_time_.fractions();
// Rtp time from incoming SenderReport.
if (rtcp_timestamp)
*rtcp_timestamp = remote_sender_rtp_time_;
// Local NTP time when we received a RTCP packet with a send block.
if (rtcp_arrival_time_secs)
*rtcp_arrival_time_secs = last_received_sr_ntp_.seconds();
if (rtcp_arrival_time_frac)
*rtcp_arrival_time_frac = last_received_sr_ntp_.fractions();
return true;
}
int VCMTiming::TargetVideoDelay() const {
rtc::CritScope cs(&crit_sect_);
return TargetDelayInternal();
}
int VCMTiming::TargetDelayInternal() const {
return std::max(min_playout_delay_ms_,
jitter_delay_ms_ + RequiredDecodeTimeMs() + render_delay_ms_);
}
//更新度量信息
RtpStreamsSynchronizer::Process
===>
UpdateMeasurements(&video_measurement_, *video_info)
bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
const Syncable::Info& info) {
RTC_DCHECK(stream);
stream->latest_timestamp = info.latest_received_capture_timestamp;
stream->latest_receive_time_ms = info.latest_receive_time_ms;
bool new_rtcp_sr = false;
if (!stream->rtp_to_ntp.UpdateMeasurements( // RtpToNtpEstimator::UpdateMeasurements
info.capture_time_ntp_secs, info.capture_time_ntp_frac,
info.capture_time_source_clock, &new_rtcp_sr)) {
return false;
}
return true;
}
bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
uint32_t ntp_frac,
uint32_t rtp_timestamp,
bool* new_rtcp_sr) {
*new_rtcp_sr = false;
int64_t unwrapped_rtp_timestamp = unwrapper_.Unwrap(rtp_timestamp);
RtcpMeasurement new_measurement(ntp_secs, ntp_frac, unwrapped_rtp_timestamp);
if (Contains(measurements_, new_measurement)) {
// RTCP SR report already added.
return true;
}
if (!new_measurement.ntp_time.Valid())
return false;
int64_t ntp_ms_new = new_measurement.ntp_time.ToMs();
bool invalid_sample = false;
if (!measurements_.empty()) {
int64_t old_rtp_timestamp = measurements_.front().unwrapped_rtp_timestamp;
int64_t old_ntp_ms = measurements_.front().ntp_time.ToMs();
if (ntp_ms_new <= old_ntp_ms ||
ntp_ms_new > old_ntp_ms + kMaxAllowedRtcpNtpIntervalMs) {
invalid_sample = true;
} else if (unwrapped_rtp_timestamp <= old_rtp_timestamp) {
RTC_LOG(LS_WARNING)
<< "Newer RTCP SR report with older RTP timestamp, dropping";
invalid_sample = true;
} else if (unwrapped_rtp_timestamp - old_rtp_timestamp > (1 << 25)) {
// Sanity check. No jumps too far into the future in rtp.
invalid_sample = true;
}
}
if (invalid_sample) {
++consecutive_invalid_samples_;
if (consecutive_invalid_samples_ < kMaxInvalidSamples) {
return false;
}
RTC_LOG(LS_WARNING) << "Multiple consecutively invalid RTCP SR reports, "
"clearing measurements.";
measurements_.clear();
params_ = absl::nullopt;
}
consecutive_invalid_samples_ = 0;
// Insert new RTCP SR report.
if (measurements_.size() == kNumRtcpReportsToUse)
measurements_.pop_back();
measurements_.push_front(new_measurement);
*new_rtcp_sr = true;
// List updated, calculate new parameters.
UpdateParameters();
return true;
}
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