Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ )
Reason for revert:
Incoming fix: https://codereview.chromium.org/2675693002/
Original issue's description:
> Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ )
>
> Reason for revert:
> Breaks downstream bots
>
> Original issue's description:
> > Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ )
> >
> > Reason for revert:
> > Bugfixes related to the new jitter buffer has landed.
> >
> > Original issue's description:
> > > Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
> > >
> > > Reason for revert:
> > > Breaks tests downstream.
> > >
> > > Original issue's description:
> > > > Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
> > > >
> > > > Reason for revert:
> > > > Fix in this CL: https://codereview.chromium.org/2640793003/
> > > >
> > > > Original issue's description:
> > > > > Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
> > > > >
> > > > > Reason for revert:
> > > > > Breaks android bots.
> > > > >
> > > > > Original issue's description:
> > > > > > Make the new jitter buffer the default jitter buffer.
> > > > > >
> > > > > > This CL contains only the changes necessary to make the switch to the new jitter
> > > > > > buffer, clean up will be done in follow up CLs.
> > > > > >
> > > > > > In this CL:
> > > > > > - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > > > > > new video jitter buffer the default one.
> > > > > > - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > > > > > WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> > > > > >
> > > > > > BUG=webrtc:5514
> > > > > >
> > > > > > Review-Url: https://codereview.webrtc.org/2627463004
> > > > > > Cr-Commit-Position: refs/heads/master@{#16114}
> > > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/0f0763d86d5d4e7f27e8dece02560e39c6da97d6
> > > > >
> > > > > TBR=stefan@webrtc.org,terelius@webrtc.org
> > > > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > > > NOPRESUBMIT=true
> > > > > NOTREECHECKS=true
> > > > > NOTRY=true
> > > > > BUG=webrtc:5514
> > > > >
> > > > > Review-Url: https://codereview.webrtc.org/2632123005
> > > > > Cr-Commit-Position: refs/heads/master@{#16117}
> > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/c08c191f7d206dc0de945185370d18f29d556931
> > > >
> > > > TBR=stefan@webrtc.org,terelius@webrtc.org
> > > > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > > > BUG=webrtc:5514
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2642753002
> > > > Cr-Commit-Position: refs/heads/master@{#16149}
> > > > Committed: https://chromium.googlesource.com/external/webrtc/+/f20dd0014d1cfc8a2e859a9e177e7fe2b21274ca
> > >
> > > TBR=stefan@webrtc.org,terelius@webrtc.org,philipel@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:5514
> > >
> > > Review-Url: https://codereview.webrtc.org/2638423003
> > > Cr-Commit-Position: refs/heads/master@{#16159}
> > > Committed: https://chromium.googlesource.com/external/webrtc/+/04926b82641c426d764aa6e013e133db519129db
> >
> > TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2652043005
> > Cr-Commit-Position: refs/heads/master@{#16293}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/09d6ef00fc21b9f2c2c27e50e5e2952329ac4b4b
>
> TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2656983002
> Cr-Commit-Position: refs/heads/master@{#16316}
> Committed: https://chromium.googlesource.com/external/webrtc/+/27378f39ced81acb1c2a61808e5e42fcf65d4b8d
TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2670183002
Cr-Commit-Position: refs/heads/master@{#16420}
diff --git a/webrtc/modules/video_coding/frame_buffer2.cc b/webrtc/modules/video_coding/frame_buffer2.cc
index 027b943..dcbcb1f 100644
--- a/webrtc/modules/video_coding/frame_buffer2.cc
+++ b/webrtc/modules/video_coding/frame_buffer2.cc
@@ -16,6 +16,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
#include "webrtc/modules/video_coding/jitter_estimator.h"
#include "webrtc/modules/video_coding/timing.h"
#include "webrtc/system_wrappers/include/clock.h"
@@ -34,7 +35,8 @@
FrameBuffer::FrameBuffer(Clock* clock,
VCMJitterEstimator* jitter_estimator,
- VCMTiming* timing)
+ VCMTiming* timing,
+ VCMReceiveStatisticsCallback* stats_callback)
: clock_(clock),
new_countinuous_frame_event_(false, false),
jitter_estimator_(jitter_estimator),
@@ -45,11 +47,10 @@
num_frames_history_(0),
num_frames_buffered_(0),
stopped_(false),
- protection_mode_(kProtectionNack) {}
+ protection_mode_(kProtectionNack),
+ stats_callback_(stats_callback) {}
-FrameBuffer::~FrameBuffer() {
- UpdateHistograms();
-}
+FrameBuffer::~FrameBuffer() {}
FrameBuffer::ReturnReason FrameBuffer::NextFrame(
int64_t max_wait_time_ms,
@@ -172,9 +173,8 @@
rtc::CritScope lock(&crit_);
RTC_DCHECK(frame);
- ++num_total_frames_;
- if (frame->num_references == 0)
- ++num_key_frames_;
+ if (stats_callback_)
+ stats_callback_->OnCompleteFrame(frame->num_references == 0, frame->size());
FrameKey key(frame->picture_id, frame->spatial_layer);
int last_continuous_picture_id =
@@ -388,28 +388,22 @@
}
void FrameBuffer::UpdateJitterDelay() {
- int unused;
- int delay;
- timing_->GetTimings(&unused, &unused, &unused, &unused, &delay, &unused,
- &unused);
+ if (!stats_callback_)
+ return;
- accumulated_delay_ += delay;
- ++accumulated_delay_samples_;
-}
-
-void FrameBuffer::UpdateHistograms() const {
- rtc::CritScope lock(&crit_);
- if (num_total_frames_ > 0) {
- int key_frames_permille = (static_cast<float>(num_key_frames_) * 1000.0f /
- static_cast<float>(num_total_frames_) +
- 0.5f);
- RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
- key_frames_permille);
- }
-
- if (accumulated_delay_samples_ > 0) {
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
- accumulated_delay_ / accumulated_delay_samples_);
+ int decode_ms;
+ int max_decode_ms;
+ int current_delay_ms;
+ int target_delay_ms;
+ int jitter_buffer_ms;
+ int min_playout_delay_ms;
+ int render_delay_ms;
+ if (timing_->GetTimings(&decode_ms, &max_decode_ms, ¤t_delay_ms,
+ &target_delay_ms, &jitter_buffer_ms,
+ &min_playout_delay_ms, &render_delay_ms)) {
+ stats_callback_->OnFrameBufferTimingsUpdated(
+ decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
+ jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
}
}
diff --git a/webrtc/modules/video_coding/frame_buffer2.h b/webrtc/modules/video_coding/frame_buffer2.h
index 529428d..b554f5b 100644
--- a/webrtc/modules/video_coding/frame_buffer2.h
+++ b/webrtc/modules/video_coding/frame_buffer2.h
@@ -28,6 +28,7 @@
namespace webrtc {
class Clock;
+class VCMReceiveStatisticsCallback;
class VCMJitterEstimator;
class VCMTiming;
@@ -39,7 +40,8 @@
FrameBuffer(Clock* clock,
VCMJitterEstimator* jitter_estimator,
- VCMTiming* timing);
+ VCMTiming* timing,
+ VCMReceiveStatisticsCallback* stats_proxy);
virtual ~FrameBuffer();
@@ -141,8 +143,6 @@
void UpdateJitterDelay() EXCLUSIVE_LOCKS_REQUIRED(crit_);
- void UpdateHistograms() const;
-
void ClearFramesAndHistory() EXCLUSIVE_LOCKS_REQUIRED(crit_);
FrameMap frames_ GUARDED_BY(crit_);
@@ -161,16 +161,9 @@
int num_frames_buffered_ GUARDED_BY(crit_);
bool stopped_ GUARDED_BY(crit_);
VCMVideoProtection protection_mode_ GUARDED_BY(crit_);
+ VCMReceiveStatisticsCallback* const stats_callback_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameBuffer);
-
- // For WebRTC.Video.JitterBufferDelayInMs metric.
- int64_t accumulated_delay_ = 0;
- int64_t accumulated_delay_samples_ = 0;
-
- // For WebRTC.Video.KeyFramesReceivedInPermille metric.
- int64_t num_total_frames_ = 0;
- int64_t num_key_frames_ = 0;
};
} // namespace video_coding
diff --git a/webrtc/modules/video_coding/frame_buffer2_unittest.cc b/webrtc/modules/video_coding/frame_buffer2_unittest.cc
index 1761f58..c3d8ce7 100644
--- a/webrtc/modules/video_coding/frame_buffer2_unittest.cc
+++ b/webrtc/modules/video_coding/frame_buffer2_unittest.cc
@@ -25,6 +25,9 @@
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
+using testing::_;
+using testing::Return;
+
namespace webrtc {
namespace video_coding {
@@ -54,6 +57,16 @@
return std::max<int>(0, render_time_ms - now_ms - kDecodeTime);
}
+ bool GetTimings(int* decode_ms,
+ int* max_decode_ms,
+ int* current_delay_ms,
+ int* target_delay_ms,
+ int* jitter_buffer_ms,
+ int* min_playout_delay_ms,
+ int* render_delay_ms) const override {
+ return true;
+ }
+
private:
static constexpr int kDelayMs = 50;
static constexpr int kDecodeTime = kDelayMs / 2;
@@ -82,6 +95,27 @@
int64_t ReceivedTime() const override { return 0; }
int64_t RenderTime() const override { return _renderTimeMs; }
+
+ // In EncodedImage |_length| is used to descibe its size and |_size| to
+ // describe its capacity.
+ void SetSize(int size) { _length = size; }
+};
+
+class VCMReceiveStatisticsCallbackMock : public VCMReceiveStatisticsCallback {
+ public:
+ MOCK_METHOD2(OnReceiveRatesUpdated,
+ void(uint32_t bitRate, uint32_t frameRate));
+ MOCK_METHOD2(OnCompleteFrame, void(bool is_keyframe, size_t size_bytes));
+ MOCK_METHOD1(OnDiscardedPacketsUpdated, void(int discarded_packets));
+ MOCK_METHOD1(OnFrameCountsUpdated, void(const FrameCounts& frame_counts));
+ MOCK_METHOD7(OnFrameBufferTimingsUpdated,
+ void(int decode_ms,
+ int max_decode_ms,
+ int current_delay_ms,
+ int target_delay_ms,
+ int jitter_buffer_ms,
+ int min_playout_delay_ms,
+ int render_delay_ms));
};
class TestFrameBuffer2 : public ::testing::Test {
@@ -95,7 +129,7 @@
: clock_(0),
timing_(&clock_),
jitter_estimator_(&clock_),
- buffer_(&clock_, &jitter_estimator_, &timing_),
+ buffer_(&clock_, &jitter_estimator_, &timing_, &stats_callback_),
rand_(0x34678213),
tear_down_(false),
extract_thread_(&ExtractLoop, this, "Extract Thread"),
@@ -190,6 +224,7 @@
FrameBuffer buffer_;
std::vector<std::unique_ptr<FrameObject>> frames_;
Random rand_;
+ ::testing::NiceMock<VCMReceiveStatisticsCallbackMock> stats_callback_;
int64_t max_wait_time_;
bool tear_down_;
@@ -437,5 +472,30 @@
CheckNoFrame(2);
}
+TEST_F(TestFrameBuffer2, StatsCallback) {
+ uint16_t pid = Rand();
+ uint32_t ts = Rand();
+ const int kFrameSize = 5000;
+
+ EXPECT_CALL(stats_callback_, OnCompleteFrame(true, kFrameSize));
+ EXPECT_CALL(stats_callback_,
+ OnFrameBufferTimingsUpdated(_, _, _, _, _, _, _));
+
+ {
+ std::unique_ptr<FrameObjectFake> frame(new FrameObjectFake());
+ frame->SetSize(kFrameSize);
+ frame->picture_id = pid;
+ frame->spatial_layer = 0;
+ frame->timestamp = ts;
+ frame->num_references = 0;
+ frame->inter_layer_predicted = false;
+
+ EXPECT_EQ(buffer_.InsertFrame(std::move(frame)), pid);
+ }
+
+ ExtractFrame();
+ CheckFrame(0, pid, 0);
+}
+
} // namespace video_coding
} // namespace webrtc
diff --git a/webrtc/modules/video_coding/include/video_coding_defines.h b/webrtc/modules/video_coding/include/video_coding_defines.h
index 122ddc6..dede5b6 100644
--- a/webrtc/modules/video_coding/include/video_coding_defines.h
+++ b/webrtc/modules/video_coding/include/video_coding_defines.h
@@ -90,8 +90,16 @@
class VCMReceiveStatisticsCallback {
public:
virtual void OnReceiveRatesUpdated(uint32_t bitRate, uint32_t frameRate) = 0;
+ virtual void OnCompleteFrame(bool is_keyframe, size_t size_bytes) = 0;
virtual void OnDiscardedPacketsUpdated(int discarded_packets) = 0;
virtual void OnFrameCountsUpdated(const FrameCounts& frame_counts) = 0;
+ virtual void OnFrameBufferTimingsUpdated(int decode_ms,
+ int max_decode_ms,
+ int current_delay_ms,
+ int target_delay_ms,
+ int jitter_buffer_ms,
+ int min_playout_delay_ms,
+ int render_delay_ms) = 0;
protected:
virtual ~VCMReceiveStatisticsCallback() {}
diff --git a/webrtc/modules/video_coding/timing.h b/webrtc/modules/video_coding/timing.h
index e7d2b1f..429c282 100644
--- a/webrtc/modules/video_coding/timing.h
+++ b/webrtc/modules/video_coding/timing.h
@@ -94,13 +94,13 @@
// Return current timing information. Returns true if the first frame has been
// decoded, false otherwise.
- bool GetTimings(int* decode_ms,
- int* max_decode_ms,
- int* current_delay_ms,
- int* target_delay_ms,
- int* jitter_buffer_ms,
- int* min_playout_delay_ms,
- int* render_delay_ms) const;
+ virtual bool GetTimings(int* decode_ms,
+ int* max_decode_ms,
+ int* current_delay_ms,
+ int* target_delay_ms,
+ int* jitter_buffer_ms,
+ int* min_playout_delay_ms,
+ int* render_delay_ms) const;
enum { kDefaultRenderDelayMs = 10 };
enum { kDelayMaxChangeMsPerS = 100 };
diff --git a/webrtc/modules/video_coding/video_receiver.cc b/webrtc/modules/video_coding/video_receiver.cc
index 129a1b5..14f1265 100644
--- a/webrtc/modules/video_coding/video_receiver.cc
+++ b/webrtc/modules/video_coding/video_receiver.cc
@@ -56,31 +56,14 @@
void VideoReceiver::Process() {
// Receive-side statistics
+
+ // TODO(philipel): Remove this if block when we know what to do with
+ // ReceiveStatisticsProxy::QualitySample.
if (_receiveStatsTimer.TimeUntilProcess() == 0) {
_receiveStatsTimer.Processed();
rtc::CritScope cs(&process_crit_);
if (_receiveStatsCallback != nullptr) {
- uint32_t bitRate;
- uint32_t frameRate;
- _receiver.ReceiveStatistics(&bitRate, &frameRate);
- _receiveStatsCallback->OnReceiveRatesUpdated(bitRate, frameRate);
- }
-
- if (_decoderTimingCallback != nullptr) {
- int decode_ms;
- int max_decode_ms;
- int current_delay_ms;
- int target_delay_ms;
- int jitter_buffer_ms;
- int min_playout_delay_ms;
- int render_delay_ms;
- if (_timing->GetTimings(&decode_ms, &max_decode_ms, ¤t_delay_ms,
- &target_delay_ms, &jitter_buffer_ms,
- &min_playout_delay_ms, &render_delay_ms)) {
- _decoderTimingCallback->OnDecoderTiming(
- decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
- jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
- }
+ _receiveStatsCallback->OnReceiveRatesUpdated(0, 0);
}
}
@@ -292,7 +275,7 @@
return ret;
}
-// Used for the WebRTC-NewVideoJitterBuffer experiment.
+// Used for the new jitter buffer.
// TODO(philipel): Clean up among the Decode functions as we replace
// VCMEncodedFrame with FrameObject.
int32_t VideoReceiver::Decode(const webrtc::VCMEncodedFrame* frame) {