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Revert "Preparation for ReceiveStatisticsProxy lock reduction."

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This reverts commit 24eed2735b2135227bcfefbabf34a89f9a5fec99.

Reason for revert: Speculative revert: breaks downstream project

Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
> 
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
> 
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
> 
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
> 
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
> 
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
> 
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}

TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
This commit is contained in:
Artem Titov
2020-04-07 18:02:39 +00:00
committed by Mirko Bonadei
parent 7e60483915
commit 16cc9efd54
19 changed files with 100 additions and 655 deletions
Download Patch File Download Diff File Expand all files Collapse all files

53
call/call_perf_tests.cc
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@ -96,24 +96,21 @@ class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
static const int kMinRunTimeMs = 30000;
public:
explicit VideoRtcpAndSyncObserver(TaskQueueBase* task_queue,
Clock* clock,
const std::string& test_label)
explicit VideoRtcpAndSyncObserver(Clock* clock, const std::string& test_label)
: test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs),
clock_(clock),
test_label_(test_label),
creation_time_ms_(clock_->TimeInMilliseconds()),
task_queue_(task_queue) {}
first_time_in_sync_(-1),
receive_stream_(nullptr) {}
void OnFrame(const VideoFrame& video_frame) override {
task_queue_->PostTask(ToQueuedTask([this]() { CheckStats(); }));
}
void CheckStats() {
if (!receive_stream_)
return;
VideoReceiveStream::Stats stats = receive_stream_->GetStats();
VideoReceiveStream::Stats stats;
{
rtc::CritScope lock(&crit_);
if (receive_stream_)
stats = receive_stream_->GetStats();
}
if (stats.sync_offset_ms == std::numeric_limits<int>::max())
return;
@ -138,8 +135,7 @@ class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
}
void set_receive_stream(VideoReceiveStream* receive_stream) {
RTC_DCHECK_EQ(task_queue_, TaskQueueBase::Current());
// Note that receive_stream may be nullptr.
rtc::CritScope lock(&crit_);
receive_stream_ = receive_stream;
}
@ -152,10 +148,10 @@ class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
Clock* const clock_;
std::string test_label_;
const int64_t creation_time_ms_;
int64_t first_time_in_sync_ = -1;
VideoReceiveStream* receive_stream_ = nullptr;
int64_t first_time_in_sync_;
rtc::CriticalSection crit_;
VideoReceiveStream* receive_stream_ RTC_GUARDED_BY(crit_);
std::vector<double> sync_offset_ms_list_;
TaskQueueBase* const task_queue_;
};
void CallPerfTest::TestAudioVideoSync(FecMode fec,
@ -172,8 +168,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
audio_net_config.queue_delay_ms = 500;
audio_net_config.loss_percent = 5;
auto observer = std::make_unique<VideoRtcpAndSyncObserver>(
task_queue(), Clock::GetRealTimeClock(), test_label);
VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), test_label);
std::map<uint8_t, MediaType> audio_pt_map;
std::map<uint8_t, MediaType> video_pt_map;
@ -223,7 +218,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
});
audio_send_transport = std::make_unique<test::PacketTransport>(
task_queue(), sender_call_.get(), observer.get(),
task_queue(), sender_call_.get(), &observer,
test::PacketTransport::kSender, audio_pt_map,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
@ -231,7 +226,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
audio_send_transport->SetReceiver(receiver_call_->Receiver());
video_send_transport = std::make_unique<test::PacketTransport>(
task_queue(), sender_call_.get(), observer.get(),
task_queue(), sender_call_.get(), &observer,
test::PacketTransport::kSender, video_pt_map,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(
@ -239,7 +234,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
video_send_transport->SetReceiver(receiver_call_->Receiver());
receive_transport = std::make_unique<test::PacketTransport>(
task_queue(), receiver_call_.get(), observer.get(),
task_queue(), receiver_call_.get(), &observer,
test::PacketTransport::kReceiver, payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(
@ -264,7 +259,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
video_receive_configs_[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
}
video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
video_receive_configs_[0].renderer = observer.get();
video_receive_configs_[0].renderer = &observer;
video_receive_configs_[0].sync_group = kSyncGroup;
AudioReceiveStream::Config audio_recv_config;
@ -286,7 +281,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
}
EXPECT_EQ(1u, video_receive_streams_.size());
observer->set_receive_stream(video_receive_streams_[0]);
observer.set_receive_stream(video_receive_streams_[0]);
drifting_clock = std::make_unique<DriftingClock>(clock_, video_ntp_speed);
CreateFrameGeneratorCapturerWithDrift(drifting_clock.get(), video_rtp_speed,
kDefaultFramerate, kDefaultWidth,
@ -298,13 +293,10 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
audio_receive_stream->Start();
});
EXPECT_TRUE(observer->Wait())
EXPECT_TRUE(observer.Wait())
<< "Timed out while waiting for audio and video to be synchronized.";
SendTask(RTC_FROM_HERE, task_queue(), [&]() {
// Clear the pointer to the receive stream since it will now be deleted.
observer->set_receive_stream(nullptr);
audio_send_stream->Stop();
audio_receive_stream->Stop();
@ -322,7 +314,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
DestroyCalls();
});
observer->PrintResults();
observer.PrintResults();
// In quick test synchronization may not be achieved in time.
if (!field_trial::IsEnabled("WebRTC-QuickPerfTest")) {
@ -331,9 +323,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
#endif
}
task_queue()->PostTask(
ToQueuedTask([to_delete = observer.release()]() { delete to_delete; }));
}
TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithoutClockDrift) {