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Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ )

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Reason for revert:
Breaks android buildbots.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with appropriate changes to API to not break depending projects.
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2812913002
> Cr-Commit-Position: refs/heads/master@{#17651}
> Committed: 774f6b4b96

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2809653004
Cr-Commit-Position: refs/heads/master@{#17653}
This commit is contained in:
ilnik
2017-04-11 06:20:05 -07:00
committed by Commit bot
parent 4d027576a6
commit 27c46e2872
48 changed files with 91 additions and 473 deletions
Download Patch File Download Diff File Expand all files Collapse all files

115
webrtc/video/end_to_end_tests.cc
View File

@ -2652,8 +2652,7 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx,
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.CurrentDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.OnewayDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EncodeTimeInMs"));
@ -2693,118 +2692,6 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx,
metrics::NumSamples("WebRTC.Video.ReceivedFecPacketsInPercent"));
}
TEST_F(EndToEndTest, ContentTypeSwitches) {
class StatsObserver : public test::BaseTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
StatsObserver() : BaseTest(kLongTimeoutMs), num_frames_received_(0) {}
bool ShouldCreateReceivers() const override { return true; }
void OnFrame(const VideoFrame& video_frame) override {
// The RTT is needed to estimate |ntp_time_ms| which is used by
// end-to-end delay stats. Therefore, start counting received frames once
// |ntp_time_ms| is valid.
if (video_frame.ntp_time_ms() > 0 &&
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >=
video_frame.ntp_time_ms()) {
rtc::CritScope lock(&crit_);
++num_frames_received_;
}
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (MinNumberOfFramesReceived())
observation_complete_.Set();
return SEND_PACKET;
}
bool MinNumberOfFramesReceived() const {
const int kMinRequiredHistogramSamples = 200;
rtc::CritScope lock(&crit_);
return num_frames_received_ > kMinRequiredHistogramSamples;
}
// May be called several times.
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out waiting for enough packets.";
// Reset frame counter so next PerformTest() call will do something.
{
rtc::CritScope lock(&crit_);
num_frames_received_ = 0;
}
}
rtc::CriticalSection crit_;
int num_frames_received_ GUARDED_BY(&crit_);
} test;
metrics::Reset();
Call::Config send_config(test.GetSenderCallConfig());
CreateSenderCall(send_config);
Call::Config recv_config(test.GetReceiverCallConfig());
CreateReceiverCall(recv_config);
receive_transport_.reset(test.CreateReceiveTransport());
send_transport_.reset(test.CreateSendTransport(sender_call_.get()));
send_transport_->SetReceiver(receiver_call_->Receiver());
receive_transport_->SetReceiver(sender_call_->Receiver());
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
CreateSendConfig(1, 0, 0, send_transport_.get());
CreateMatchingReceiveConfigs(receive_transport_.get());
// Modify send and receive configs.
video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_receive_configs_[0].renderer = &test;
// RTT needed for RemoteNtpTimeEstimator for the receive stream.
video_receive_configs_[0].rtp.rtcp_xr.receiver_reference_time_report = true;
// Start with realtime video.
video_encoder_config_.content_type =
VideoEncoderConfig::ContentType::kRealtimeVideo;
// Second encoder config for the second part of the test uses screenshare
VideoEncoderConfig encoder_config_with_screenshare_ =
video_encoder_config_.Copy();
encoder_config_with_screenshare_.content_type =
VideoEncoderConfig::ContentType::kScreen;
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
test.PerformTest();
// Replace old send stream.
sender_call_->DestroyVideoSendStream(video_send_stream_);
video_send_stream_ = sender_call_->CreateVideoSendStream(
video_send_config_.Copy(), encoder_config_with_screenshare_.Copy());
video_send_stream_->SetSource(
frame_generator_capturer_.get(),
VideoSendStream::DegradationPreference::kBalanced);
video_send_stream_->Start();
// Continue to run test but now with screenshare.
test.PerformTest();
send_transport_->StopSending();
receive_transport_->StopSending();
Stop();
DestroyStreams();
DestroyCalls();
// Delete the call for Call stats to be reported.
sender_call_.reset();
receiver_call_.reset();
// Verify that stats have been updated for both screenshare and video.
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayInMs"));
EXPECT_EQ(1,
metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayMaxInMs"));
EXPECT_EQ(
1, metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayMaxInMs"));
}
TEST_F(EndToEndTest, VerifyHistogramStatsWithRtx) {
const bool kEnabledRtx = true;
const bool kEnabledRed = false;