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Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ )

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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: 0f0763d86d
> > > >
> > > > 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: c08c191f7d
> > >
> > > 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: f20dd0014d
> >
> > 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: 04926b8264
>
> 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: 09d6ef00fc

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}
This commit is contained in:
philipel
2017-01-27 02:19:05 -08:00
committed by Commit bot
parent 54a05b2084
commit 27378f39ce
17 changed files with 242 additions and 309 deletions
Download Patch File Download Diff File Expand all files Collapse all files

52
webrtc/modules/video_coding/frame_buffer2.cc
View File

@ -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 @@ constexpr int kMaxFramesHistory = 50;
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 @@ FrameBuffer::FrameBuffer(Clock* clock,
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 @@ int FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
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 @@ bool FrameBuffer::UpdateFrameInfoWithIncomingFrame(const FrameObject& frame,
}
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, &current_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_);
}
}