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Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )

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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}
This commit is contained in:
philipel
2017-01-17 04:03:53 -08:00
committed by Commit bot
parent 6c0fd4341c
commit c08c191f7d
16 changed files with 235 additions and 307 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,
@ -163,8 +162,9 @@ 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 =
@ -365,22 +365,28 @@ 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_);
}
}