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Fix issue with zero rtt reports when using FlexFEC and add perf test.

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BUG=webrtc:7938

Review-Url: https://codereview.webrtc.org/2966153002
Cr-Commit-Position: refs/heads/master@{#18898}
This commit is contained in:
stefan 2017-07-05 03:03:02 -07:00 committed by Commit Bot
parent 070efc088e
commit 889d9654f7
3 changed files with 57 additions and 2 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
View File

@ -209,8 +209,10 @@ void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
if (first_report_time_ms_ == -1)
first_report_time_ms_ = now_ms;
// Update RTT.
last_round_trip_time_ms_ = rtt;
// Update RTT if we were able to compute an RTT based on this RTCP.
// FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
if (rtt > 0)
last_round_trip_time_ms_ = rtt;
// Check sequence number diff and weight loss report
if (number_of_packets > 0) {

28
webrtc/video/full_stack_tests.cc
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@ -136,6 +136,34 @@ TEST_F(FullStackTest, ForemanCifPlr5Flexfec) {
RunTest(foreman_cif);
}
TEST_F(FullStackTest, ForemanCif500kbpsPlr3Flexfec) {
VideoQualityTest::Params foreman_cif;
foreman_cif.call.send_side_bwe = true;
foreman_cif.video = {true, 352, 288, 30, 30000, 500000, 2000000,
false, "VP8", 1, 0, 0, false, true,
"", "foreman_cif"};
foreman_cif.analyzer = {"foreman_cif_500kbps_delay_50_0_plr_3_flexfec", 0.0,
0.0, kFullStackTestDurationSecs};
foreman_cif.pipe.loss_percent = 3;
foreman_cif.pipe.link_capacity_kbps = 500;
foreman_cif.pipe.queue_delay_ms = 50;
RunTest(foreman_cif);
}
TEST_F(FullStackTest, ForemanCif500kbpsPlr3Ulpfec) {
VideoQualityTest::Params foreman_cif;
foreman_cif.call.send_side_bwe = true;
foreman_cif.video = {true, 352, 288, 30, 30000, 500000, 2000000,
false, "VP8", 1, 0, 0, true, false,
"", "foreman_cif"};
foreman_cif.analyzer = {"foreman_cif_500kbps_delay_50_0_plr_3_ulpfec", 0.0,
0.0, kFullStackTestDurationSecs};
foreman_cif.pipe.loss_percent = 3;
foreman_cif.pipe.link_capacity_kbps = 500;
foreman_cif.pipe.queue_delay_ms = 50;
RunTest(foreman_cif);
}
#if defined(WEBRTC_USE_H264)
TEST_F(FullStackTest, ForemanCifWithoutPacketlossH264) {
// TODO(pbos): Decide on psnr/ssim thresholds for foreman_cif.

25
webrtc/video/video_quality_test.cc
View File

@ -155,6 +155,7 @@ class VideoAnalyzer : public PacketReceiver,
Clock* clock)
: transport_(transport),
receiver_(nullptr),
call_(nullptr),
send_stream_(nullptr),
receive_stream_(nullptr),
captured_frame_forwarder_(this, clock),
@ -168,6 +169,7 @@ class VideoAnalyzer : public PacketReceiver,
selected_tl_(selected_tl),
pre_encode_proxy_(this),
encode_timing_proxy_(this),
last_fec_bytes_(0),
frames_to_process_(duration_frames),
frames_recorded_(0),
frames_processed_(0),
@ -230,6 +232,12 @@ class VideoAnalyzer : public PacketReceiver,
video_capturer->AddOrUpdateSink(InputInterface(), wants);
}
void SetCall(Call* call) {
rtc::CritScope lock(&crit_);
RTC_DCHECK(!call_);
call_ = call;
}
void SetSendStream(VideoSendStream* stream) {
rtc::CritScope lock(&crit_);
RTC_DCHECK(!send_stream_);
@ -650,6 +658,9 @@ class VideoAnalyzer : public PacketReceiver,
while (!done_.Wait(kSendStatsPollingIntervalMs)) {
rtc::CritScope crit(&comparison_lock_);
Call::Stats call_stats = call_->GetStats();
send_bandwidth_bps_.AddSample(call_stats.send_bandwidth_bps);
VideoSendStream::Stats send_stats = send_stream_->GetStats();
// It's not certain that we yet have estimates for any of these stats.
// Check that they are positive before mixing them in.
@ -661,6 +672,13 @@ class VideoAnalyzer : public PacketReceiver,
encode_usage_percent_.AddSample(send_stats.encode_usage_percent);
if (send_stats.media_bitrate_bps > 0)
media_bitrate_bps_.AddSample(send_stats.media_bitrate_bps);
size_t fec_bytes = 0;
for (auto kv : send_stats.substreams) {
fec_bytes += kv.second.rtp_stats.fec.payload_bytes +
kv.second.rtp_stats.fec.padding_bytes;
}
fec_bitrate_bps_.AddSample((fec_bytes - last_fec_bytes_) * 8);
last_fec_bytes_ = fec_bytes;
if (receive_stream_ != nullptr) {
VideoReceiveStream::Stats receive_stats = receive_stream_->GetStats();
@ -764,6 +782,8 @@ class VideoAnalyzer : public PacketReceiver,
PrintResult("encode_frame_rate", encode_frame_rate_, " fps");
PrintResult("encode_time", encode_time_ms_, " ms");
PrintResult("media_bitrate", media_bitrate_bps_, " bps");
PrintResult("fec_bitrate", fec_bitrate_bps_, " bps");
PrintResult("send_bandwidth", send_bandwidth_bps_, " bps");
if (receive_stream_ != nullptr) {
PrintResult("decode_time", decode_time_ms_, " ms");
@ -969,6 +989,7 @@ class VideoAnalyzer : public PacketReceiver,
frames_.push_back(video_frame);
}
Call* call_;
VideoSendStream* send_stream_;
VideoReceiveStream* receive_stream_;
CapturedFrameForwarder captured_frame_forwarder_;
@ -997,8 +1018,11 @@ class VideoAnalyzer : public PacketReceiver,
test::Statistics decode_time_ms_ GUARDED_BY(comparison_lock_);
test::Statistics decode_time_max_ms_ GUARDED_BY(comparison_lock_);
test::Statistics media_bitrate_bps_ GUARDED_BY(comparison_lock_);
test::Statistics fec_bitrate_bps_ GUARDED_BY(comparison_lock_);
test::Statistics send_bandwidth_bps_ GUARDED_BY(comparison_lock_);
test::Statistics memory_usage_ GUARDED_BY(comparison_lock_);
size_t last_fec_bytes_;
const int frames_to_process_;
int frames_recorded_;
@ -1717,6 +1741,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
kSendRtxSsrcs[params_.ss.selected_stream],
static_cast<size_t>(params_.ss.selected_stream), params.ss.selected_sl,
params_.video.selected_tl, is_quick_test_enabled, clock_);
analyzer.SetCall(sender_call_.get());
analyzer.SetReceiver(receiver_call_->Receiver());
send_transport.SetReceiver(&analyzer);
recv_transport.SetReceiver(sender_call_->Receiver());