Stop using VoEVideoSync in Call/VideoReceiveStream.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2452163004
Cr-Commit-Position: refs/heads/master@{#16375}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index aaf77a8..1f24b2c 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -21,6 +21,8 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
@@ -81,6 +83,8 @@
RTC_DCHECK(remote_bitrate_estimator);
RTC_DCHECK(rtp_header_parser_);
+ module_process_thread_checker_.DetachFromThread();
+
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
channel_proxy_->SetRtcEventLog(event_log);
@@ -125,7 +129,7 @@
}
AudioReceiveStream::~AudioReceiveStream() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
if (playing_) {
Stop();
@@ -138,7 +142,7 @@
}
void AudioReceiveStream::Start() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (playing_) {
return;
}
@@ -159,7 +163,7 @@
}
void AudioReceiveStream::Stop() {
- RTC_DCHECK_RUN_ON(&thread_checker_);
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (!playing_) {
return;
}
@@ -170,7 +174,7 @@
}
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
- RTC_DCHECK_RUN_ON(&thread_checker_);
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
webrtc::AudioReceiveStream::Stats stats;
stats.remote_ssrc = config_.rtp.remote_ssrc;
ScopedVoEInterface<VoECodec> codec(voice_engine());
@@ -220,22 +224,78 @@
}
void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_proxy_->SetSink(std::move(sink));
}
void AudioReceiveStream::SetGain(float gain) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
channel_proxy_->SetChannelOutputVolumeScaling(gain);
}
-const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- return config_;
+AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
+ int sample_rate_hz,
+ AudioFrame* audio_frame) {
+ return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+}
+
+int AudioReceiveStream::Ssrc() const {
+ return config_.rtp.remote_ssrc;
+}
+
+int AudioReceiveStream::PreferredSampleRate() const {
+ return channel_proxy_->NeededFrequency();
+}
+
+int AudioReceiveStream::id() const {
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+ return config_.rtp.remote_ssrc;
+}
+
+rtc::Optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
+ RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+ Syncable::Info info;
+
+ RtpRtcp* rtp_rtcp = nullptr;
+ RtpReceiver* rtp_receiver = nullptr;
+ channel_proxy_->GetRtpRtcp(&rtp_rtcp, &rtp_receiver);
+ RTC_DCHECK(rtp_rtcp);
+ RTC_DCHECK(rtp_receiver);
+
+ if (!rtp_receiver->Timestamp(&info.latest_received_capture_timestamp)) {
+ return rtc::Optional<Syncable::Info>();
+ }
+ if (!rtp_receiver->LastReceivedTimeMs(&info.latest_receive_time_ms)) {
+ return rtc::Optional<Syncable::Info>();
+ }
+ if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs,
+ &info.capture_time_ntp_frac,
+ nullptr,
+ nullptr,
+ &info.capture_time_source_clock) != 0) {
+ return rtc::Optional<Syncable::Info>();
+ }
+
+ int jitter_buffer_delay_ms = 0;
+ int playout_buffer_delay_ms = 0;
+ channel_proxy_->GetDelayEstimate(&jitter_buffer_delay_ms,
+ &playout_buffer_delay_ms);
+ info.current_delay_ms = jitter_buffer_delay_ms + playout_buffer_delay_ms;
+ return rtc::Optional<Syncable::Info>(info);
+}
+
+uint32_t AudioReceiveStream::GetPlayoutTimestamp() const {
+ // Called on video capture thread.
+ return channel_proxy_->GetPlayoutTimestamp();
+}
+
+void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
+ RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+ return channel_proxy_->SetMinimumPlayoutDelay(delay_ms);
}
void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (send_stream) {
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
std::unique_ptr<voe::ChannelProxy> send_channel_proxy =
@@ -247,7 +307,7 @@
}
void AudioReceiveStream::SignalNetworkState(NetworkState state) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
}
bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
@@ -286,24 +346,9 @@
return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
}
-AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
- int sample_rate_hz,
- AudioFrame* audio_frame) {
- return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
-}
-
-int AudioReceiveStream::PreferredSampleRate() const {
- return channel_proxy_->NeededFrequency();
-}
-
-int AudioReceiveStream::Ssrc() const {
- return config_.rtp.remote_ssrc;
-}
-
-internal::AudioState* AudioReceiveStream::audio_state() const {
- auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
- RTC_DCHECK(audio_state);
- return audio_state;
+const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+ return config_;
}
VoiceEngine* AudioReceiveStream::voice_engine() const {
@@ -312,6 +357,12 @@
return voice_engine;
}
+internal::AudioState* AudioReceiveStream::audio_state() const {
+ auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
+ RTC_DCHECK(audio_state);
+ return audio_state;
+}
+
int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
ScopedVoEInterface<VoEBase> base(voice_engine());
if (playout) {
@@ -320,6 +371,5 @@
return base->StopPlayout(config_.voe_channel_id);
}
}
-
} // namespace internal
} // namespace webrtc
