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}
diff --git a/webrtc/modules/video_coding/frame_buffer2.cc b/webrtc/modules/video_coding/frame_buffer2.cc
index 2a7e85e..0831c0c 100644
--- a/webrtc/modules/video_coding/frame_buffer2.cc
+++ b/webrtc/modules/video_coding/frame_buffer2.cc
@@ -16,7 +16,6 @@
#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"
@@ -35,8 +34,7 @@
FrameBuffer::FrameBuffer(Clock* clock,
VCMJitterEstimator* jitter_estimator,
- VCMTiming* timing,
- VCMReceiveStatisticsCallback* stats_callback)
+ VCMTiming* timing)
: clock_(clock),
new_countinuous_frame_event_(false, false),
jitter_estimator_(jitter_estimator),
@@ -47,10 +45,11 @@
num_frames_history_(0),
num_frames_buffered_(0),
stopped_(false),
- protection_mode_(kProtectionNack),
- stats_callback_(stats_callback) {}
+ protection_mode_(kProtectionNack) {}
-FrameBuffer::~FrameBuffer() {}
+FrameBuffer::~FrameBuffer() {
+ UpdateHistograms();
+}
FrameBuffer::ReturnReason FrameBuffer::NextFrame(
int64_t max_wait_time_ms,
@@ -166,8 +165,9 @@
rtc::CritScope lock(&crit_);
RTC_DCHECK(frame);
- if (stats_callback_)
- stats_callback_->OnCompleteFrame(frame->num_references == 0, frame->size());
+ ++num_total_frames_;
+ if (frame->num_references == 0)
+ ++num_key_frames_;
FrameKey key(frame->picture_id, frame->spatial_layer);
int last_continuous_picture_id =
@@ -381,22 +381,28 @@
}
void FrameBuffer::UpdateJitterDelay() {
- if (!stats_callback_)
- return;
+ int unused;
+ int delay;
+ timing_->GetTimings(&unused, &unused, &unused, &unused, &delay, &unused,
+ &unused);
- 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);
+ 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_);
}
}