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Unified Diff: webrtc/video/video_receive_stream.cc

Issue 2452163004: Stop using VoEVideoSync in Call/VideoReceiveStream. (Closed)
Patch Set: comment Created 3 years, 11 months ago
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Index: webrtc/video/video_receive_stream.cc
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 5fd436d6a4caea4672327772d1350fc8c7c01d2e..25418c4d8fe40df1f1d1e07b8d41a4e55646ae2a 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -22,6 +22,8 @@
#include "webrtc/common_video/h264/profile_level_id.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/modules/video_coding/frame_object.h"
#include "webrtc/modules/video_coding/include/video_coding.h"
@@ -33,7 +35,6 @@
#include "webrtc/video/call_stats.h"
#include "webrtc/video/receive_statistics_proxy.h"
#include "webrtc/video_receive_stream.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
namespace webrtc {
@@ -190,7 +191,6 @@ VideoReceiveStream::VideoReceiveStream(
CongestionController* congestion_controller,
PacketRouter* packet_router,
VideoReceiveStream::Config config,
- webrtc::VoiceEngine* voice_engine,
ProcessThread* process_thread,
CallStats* call_stats,
VieRemb* remb)
@@ -220,7 +220,7 @@ VideoReceiveStream::VideoReceiveStream(
this, // KeyFrameRequestSender
this, // OnCompleteFrameCallback
timing_.get()),
- rtp_stream_sync_(&video_receiver_, &rtp_stream_receiver_),
+ rtp_stream_sync_(this),
jitter_buffer_experiment_(
field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") ==
"Enabled") {
@@ -230,6 +230,8 @@ VideoReceiveStream::VideoReceiveStream(
RTC_DCHECK(congestion_controller_);
RTC_DCHECK(call_stats_);
+ module_process_thread_checker_.DetachFromThread();
+
RTC_DCHECK(!config_.decoders.empty());
std::set<int> decoder_payload_types;
for (const Decoder& decoder : config_.decoders) {
@@ -254,6 +256,7 @@ VideoReceiveStream::VideoReceiveStream(
}
VideoReceiveStream::~VideoReceiveStream() {
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
Stop();
@@ -265,6 +268,7 @@ VideoReceiveStream::~VideoReceiveStream() {
}
void VideoReceiveStream::SignalNetworkState(NetworkState state) {
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
rtp_stream_receiver_.SignalNetworkState(state);
}
@@ -281,21 +285,17 @@ bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
bool VideoReceiveStream::OnRecoveredPacket(const uint8_t* packet,
size_t length) {
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return rtp_stream_receiver_.OnRecoveredPacket(packet, length);
}
-void VideoReceiveStream::SetSyncChannel(VoiceEngine* voice_engine,
- int audio_channel_id) {
- if (voice_engine && audio_channel_id != -1) {
- VoEVideoSync* voe_sync_interface = VoEVideoSync::GetInterface(voice_engine);
- rtp_stream_sync_.ConfigureSync(audio_channel_id, voe_sync_interface);
- voe_sync_interface->Release();
- } else {
- rtp_stream_sync_.ConfigureSync(-1, nullptr);
- }
+void VideoReceiveStream::SetSync(Syncable* audio_syncable) {
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+ rtp_stream_sync_.ConfigureSync(audio_syncable);
}
void VideoReceiveStream::Start() {
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
if (decode_thread_.IsRunning())
return;
@@ -346,6 +346,7 @@ void VideoReceiveStream::Start() {
}
void VideoReceiveStream::Stop() {
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
rtp_stream_receiver_.StopReceive();
// TriggerDecoderShutdown will release any waiting decoder thread and make it
// stop immediately, instead of waiting for a timeout. Needs to be called
@@ -407,7 +408,9 @@ void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) {
// function itself, another in GetChannel() and a third in
// GetPlayoutTimestamp. Seems excessive. Anyhow, I'm assuming the function
// succeeds most of the time, which leads to grabbing a fourth lock.
- if (rtp_stream_sync_.GetStreamSyncOffsetInMs(video_frame, &sync_offset_ms,
+ if (rtp_stream_sync_.GetStreamSyncOffsetInMs(video_frame.timestamp(),
+ video_frame.render_time_ms(),
+ &sync_offset_ms,
&estimated_freq_khz)) {
// TODO(tommi): OnSyncOffsetUpdated grabs a lock.
stats_proxy_.OnSyncOffsetUpdated(sync_offset_ms, estimated_freq_khz);
@@ -461,6 +464,46 @@ void VideoReceiveStream::OnCompleteFrame(
rtp_stream_receiver_.FrameContinuous(last_continuous_pid);
}
+int VideoReceiveStream::id() const {
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+ return config_.rtp.remote_ssrc;
+}
+
+rtc::Optional<Syncable::Info> VideoReceiveStream::GetInfo() const {
+ RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+ Syncable::Info info;
+
+ RtpReceiver* rtp_receiver = rtp_stream_receiver_.GetRtpReceiver();
+ 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>();
+
+ RtpRtcp* rtp_rtcp = rtp_stream_receiver_.rtp_rtcp();
+ RTC_DCHECK(rtp_rtcp);
+ 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>();
+ }
+
+ info.current_delay_ms = video_receiver_.Delay();
+ return rtc::Optional<Syncable::Info>(info);
+}
+
+uint32_t VideoReceiveStream::GetPlayoutTimestamp() const {
+ RTC_NOTREACHED();
+ return 0;
+}
+
+void VideoReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
+ RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+ video_receiver_.SetMinimumPlayoutDelay(delay_ms);
+}
+
bool VideoReceiveStream::DecodeThreadFunction(void* ptr) {
static_cast<VideoReceiveStream*>(ptr)->Decode();
return true;
@@ -489,6 +532,5 @@ void VideoReceiveStream::Decode() {
video_receiver_.Decode(kMaxDecodeWaitTimeMs);
}
}
-
} // namespace internal
} // namespace webrtc
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