mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-12 21:30:45 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions

Fix rate statistic when time window running out of samples

Browse source 
Current rate statistic tracker has assumption, the tracking window will
always be full after first filled up. This assumption looks not always
true. One example is the input_framerate_ tracker inside
video_stream_sender.cc which is used for setup frame droper and encoder.
Whenever there is a gap in video stream, like mute/unmute,
pacer pause/unpause etc. The fps detected from the rate_statistics
becomes samples_filled_partial_window / full_window_size, which could
be extremely low for a while. This creates a misalignment between the
fps we told encoder/frame dropper, and the real fps we fed into them,
which causes short-term serious overshot and very bad experience on
delay, avsync, congestion etc. This may also depends on how fast
encoder could react to the gap between set fps and real fps, but
libvpx and openh264 at least cannot handle this well.

So propose a fix to update first timestamp after tracker window
drained. This will give more accurate fps estimate similar based on
active window after sample gets drained

Bug: webrtc:13403
Change-Id: I96792c11091fe8bfa63e669f4360a3b3e95593e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35447}
This commit is contained in:
Yun Zhang 2021-11-23 01:11:20 -08:00 committed by WebRTC LUCI CQ
parent f0290eb6fd
commit 4774a9fcb8
2 changed files with 12 additions and 4 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
rtc_base/rate_statistics.cc
View file

@ -58,7 +58,7 @@ void RateStatistics::Update(int64_t count, int64_t now_ms) {
RTC_DCHECK_GE(count, 0);
EraseOld(now_ms);
if (first_timestamp_ == -1) {
if (first_timestamp_ == -1 || num_samples_ == 0) {
first_timestamp_ = now_ms;
}

14
rtc_base/rate_statistics_unittest.cc
View file

@ -148,14 +148,15 @@ TEST_F(RateStatisticsTest, ResetAfterSilence) {
now_ms += kWindowMs + 1;
EXPECT_FALSE(static_cast<bool>(stats_.Rate(now_ms)));
// Silence over window size should trigger auto reset for coming sample.
stats_.Update(1000, now_ms);
++now_ms;
stats_.Update(1000, now_ms);
// We expect two samples of 1000 bytes, and that the bitrate is measured over
// 500 ms, i.e. 2 * 8 * 1000 / 0.500 = 32000.
EXPECT_EQ(32000u, *stats_.Rate(now_ms));
// active window instead of full window, which is now_ms - first_timestamp + 1
EXPECT_EQ(kExpectedBitrate, *stats_.Rate(now_ms));
// Reset, add the same samples again.
// Manual reset, add the same samples again.
stats_.Reset();
EXPECT_FALSE(static_cast<bool>(stats_.Rate(now_ms)));
stats_.Update(1000, now_ms);
@ -272,9 +273,16 @@ TEST_F(RateStatisticsTest, HandlesQuietPeriods) {
EXPECT_FALSE(static_cast<bool>(stats_.Rate(now_ms)));
// Move window a long way out.
// This will cause an automatic reset of the window
// First data point won't give a valid result
now_ms += 2 * kWindowMs;
stats_.Update(0, now_ms);
bitrate = stats_.Rate(now_ms);
EXPECT_FALSE(static_cast<bool>(stats_.Rate(now_ms)));
// Second data point gives valid result
++now_ms;
stats_.Update(0, now_ms);
bitrate = stats_.Rate(now_ms);
EXPECT_TRUE(static_cast<bool>(bitrate));
EXPECT_EQ(0u, *bitrate);
}