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}
diff --git a/webrtc/modules/video_coding/frame_buffer2.h b/webrtc/modules/video_coding/frame_buffer2.h
index f667fd5..7af48d3 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_);
@@ -160,16 +160,9 @@
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