Reland "Refactor and remove media_optimization::MediaOptimization."

This reverts commit 6613f8e98ab3654ade7e8f5352d8d6711b157499.

Reason for revert: This change seemed innocent after all, so undoing speculative revert.

Original change's description:
> Revert "Refactor and remove media_optimization::MediaOptimization."
> 
> This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b.
> 
> Reason for revert: Speculative revert due to downstream crashes.
> 
> Original change's description:
> > Refactor and remove media_optimization::MediaOptimization.
> > 
> > This CL removes MediaOptmization and folds some of its functionality
> > into VideoStreamEncoder.
> > 
> > The FPS tracking is now handled by a RateStatistics instance. Frame
> > dropping is still handled by FrameDropper. Both of these now live
> > directly in VideoStreamEncoder.
> > There is no intended change in behavior from this CL, but due to a new
> > way of measuring frame rate, some minor perf changes can be expected.
> > 
> > A small change in behavior is that OnBitrateUpdated is now called
> > directly rather than on the next frame. Since both encoding frame and
> > setting rate allocations happen on the encoder worker thread, there's
> > really no reason to cache bitrates and wait until the next frame.
> > An edge case though is that if a new bitrate is set before the first
> > frame, we must remember that bitrate and then apply it after the video
> > bitrate allocator has been first created.
> > 
> > In addition to existing unit tests, manual tests have been used to
> > confirm that frame dropping works as expected with misbehaving encoders.
> > 
> > Bug: webrtc:10164
> > Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> > Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26147}
> 
> TBR=nisse@webrtc.org,sprang@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:10164
> Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
> Reviewed-on: https://webrtc-review.googlesource.com/c/116780
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26191}

TBR=nisse@webrtc.org,sprang@webrtc.org

Change-Id: Ieda1fad301de002460bb0bf5a75267ea065176a8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10164
Reviewed-on: https://webrtc-review.googlesource.com/c/116960
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26213}
This commit is contained in:
Niels Möller
2019-01-11 11:11:10 +01:00
committed by Commit Bot
parent 9c843906ca
commit 6bb5ab9740
13 changed files with 370 additions and 517 deletions

View File

@ -553,6 +553,16 @@ class VideoStreamEncoderTest : public ::testing::Test {
force_init_encode_failed_ = force_failure;
}
void SimulateOvershoot(double rate_factor) {
rtc::CritScope lock(&local_crit_sect_);
rate_factor_ = rate_factor;
}
uint32_t GetLastFramerate() {
rtc::CritScope lock(&local_crit_sect_);
return last_framerate_;
}
private:
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
@ -599,6 +609,25 @@ class VideoStreamEncoderTest : public ::testing::Test {
return res;
}
int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
uint32_t framerate) {
rtc::CritScope lock(&local_crit_sect_);
VideoBitrateAllocation adjusted_rate_allocation;
for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
if (rate_allocation.HasBitrate(si, ti)) {
adjusted_rate_allocation.SetBitrate(
si, ti,
static_cast<uint32_t>(rate_allocation.GetBitrate(si, ti) *
rate_factor_));
}
}
}
last_framerate_ = framerate;
return FakeEncoder::SetRateAllocation(adjusted_rate_allocation,
framerate);
}
rtc::CriticalSection local_crit_sect_;
bool block_next_encode_ RTC_GUARDED_BY(local_crit_sect_) = false;
rtc::Event continue_encode_event_;
@ -610,6 +639,8 @@ class VideoStreamEncoderTest : public ::testing::Test {
std::vector<std::unique_ptr<Vp8TemporalLayers>> allocated_temporal_layers_
RTC_GUARDED_BY(local_crit_sect_);
bool force_init_encode_failed_ RTC_GUARDED_BY(local_crit_sect_) = false;
double rate_factor_ RTC_GUARDED_BY(local_crit_sect_) = 1.0;
uint32_t last_framerate_ RTC_GUARDED_BY(local_crit_sect_) = 0;
};
class TestSink : public VideoStreamEncoder::EncoderSink {
@ -2093,9 +2124,8 @@ TEST_F(VideoStreamEncoderTest, CallsBitrateObserver) {
DefaultVideoBitrateAllocator(fake_encoder_.codec_config())
.GetAllocation(kLowTargetBitrateBps, kDefaultFps);
// First called on bitrate updated, then again on first frame.
EXPECT_CALL(bitrate_observer, OnBitrateAllocationUpdated(expected_bitrate))
.Times(2);
.Times(1);
video_stream_encoder_->OnBitrateUpdated(kLowTargetBitrateBps, 0, 0);
const int64_t kStartTimeMs = 1;
@ -3156,9 +3186,6 @@ TEST_F(VideoStreamEncoderTest, DoesNotUpdateBitrateAllocationWhenSuspended) {
MockBitrateObserver bitrate_observer;
video_stream_encoder_->SetBitrateAllocationObserver(&bitrate_observer);
EXPECT_CALL(bitrate_observer, OnBitrateAllocationUpdated(_)).Times(1);
// Initial bitrate update.
video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
video_stream_encoder_->WaitUntilTaskQueueIsIdle();
@ -3227,4 +3254,80 @@ TEST_F(VideoStreamEncoderTest,
video_stream_encoder_->Stop();
}
TEST_F(VideoStreamEncoderTest, DropsFramesWhenEncoderOvershoots) {
const int kFrameWidth = 320;
const int kFrameHeight = 240;
const int kFps = 30;
const int kTargetBitrateBps = 120000;
const int kNumFramesInRun = kFps * 5; // Runs of five seconds.
video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
int64_t timestamp_ms = fake_clock_.TimeNanos() / rtc::kNumNanosecsPerMillisec;
max_framerate_ = kFps;
// Insert 3 seconds of video, verify number of drops with normal bitrate.
fake_encoder_.SimulateOvershoot(1.0);
int num_dropped = 0;
for (int i = 0; i < kNumFramesInRun; ++i) {
video_source_.IncomingCapturedFrame(
CreateFrame(timestamp_ms, kFrameWidth, kFrameHeight));
// Wait up to two frame durations for a frame to arrive.
if (!TimedWaitForEncodedFrame(timestamp_ms, 2 * 1000 / kFps)) {
++num_dropped;
}
timestamp_ms += 1000 / kFps;
}
// Frame drops should be less than 5%
EXPECT_LT(num_dropped, 5 * kNumFramesInRun / 100);
// Make encoder produce frames at double the expected bitrate during 3 seconds
// of video, verify number of drops. Rate needs to be slightly changed in
// order to force the rate to be reconfigured.
fake_encoder_.SimulateOvershoot(2.0);
video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps + 1000, 0, 0);
num_dropped = 0;
for (int i = 0; i < kNumFramesInRun; ++i) {
video_source_.IncomingCapturedFrame(
CreateFrame(timestamp_ms, kFrameWidth, kFrameHeight));
// Wait up to two frame durations for a frame to arrive.
if (!TimedWaitForEncodedFrame(timestamp_ms, 2 * 1000 / kFps)) {
++num_dropped;
}
timestamp_ms += 1000 / kFps;
}
// Frame drops should be more than 40%.
EXPECT_GT(num_dropped, 40 * kNumFramesInRun / 100);
video_stream_encoder_->Stop();
}
TEST_F(VideoStreamEncoderTest, ConfiguresCorrectFrameRate) {
const int kFrameWidth = 320;
const int kFrameHeight = 240;
const int kActualInputFps = 24;
const int kTargetBitrateBps = 120000;
ASSERT_GT(max_framerate_, kActualInputFps);
int64_t timestamp_ms = fake_clock_.TimeNanos() / rtc::kNumNanosecsPerMillisec;
max_framerate_ = kActualInputFps;
video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
// Insert 3 seconds of video, with an input fps lower than configured max.
for (int i = 0; i < kActualInputFps * 3; ++i) {
video_source_.IncomingCapturedFrame(
CreateFrame(timestamp_ms, kFrameWidth, kFrameHeight));
// Wait up to two frame durations for a frame to arrive.
WaitForEncodedFrame(timestamp_ms);
timestamp_ms += 1000 / kActualInputFps;
}
EXPECT_NEAR(kActualInputFps, fake_encoder_.GetLastFramerate(), 1);
video_stream_encoder_->Stop();
}
} // namespace webrtc