Fix issue with zero rtt reports when using FlexFEC and add perf test.
BUG=webrtc:7938
Review-Url: https://codereview.webrtc.org/2966153002
Cr-Commit-Position: refs/heads/master@{#18898}
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index dc14905..416afbe 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -209,8 +209,10 @@
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) {
diff --git a/webrtc/video/full_stack_tests.cc b/webrtc/video/full_stack_tests.cc
index 4a7334f..e15b41a 100644
--- a/webrtc/video/full_stack_tests.cc
+++ b/webrtc/video/full_stack_tests.cc
@@ -136,6 +136,34 @@
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.
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 5e99b50..1ae2e16 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -155,6 +155,7 @@
Clock* clock)
: transport_(transport),
receiver_(nullptr),
+ call_(nullptr),
send_stream_(nullptr),
receive_stream_(nullptr),
captured_frame_forwarder_(this, clock),
@@ -168,6 +169,7 @@
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 @@
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 @@
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 @@
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 @@
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 @@
frames_.push_back(video_frame);
}
+ Call* call_;
VideoSendStream* send_stream_;
VideoReceiveStream* receive_stream_;
CapturedFrameForwarder captured_frame_forwarder_;
@@ -997,8 +1018,11 @@
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 @@
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());
