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Calculate JitterBufferDelayInMs in the new jitter buffer.

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JitterBufferDelayInMs is used for the WebRTC-NewVideoJitterBuffer finch
experiment, and therefore needs to be calculated.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2534093003
Cr-Commit-Position: refs/heads/master@{#15313}
This commit is contained in:
philipel
2016-11-30 01:31:40 -08:00
committed by Commit bot
parent e69b46863a
commit be74270ebe
3 changed files with 37 additions and 9 deletions
Download Patch File Download Diff File Expand all files Collapse all files

21
webrtc/modules/video_coding/frame_buffer2.cc
View File

@ -45,9 +45,7 @@ FrameBuffer::FrameBuffer(Clock* clock,
num_frames_history_(0),
num_frames_buffered_(0),
stopped_(false),
protection_mode_(kProtectionNack),
num_total_frames_(0),
num_key_frames_(0) {}
protection_mode_(kProtectionNack) {}
FrameBuffer::~FrameBuffer() {
UpdateHistograms();
@ -133,6 +131,8 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
timing_->UpdateCurrentDelay(frame->RenderTime(),
clock_->TimeInMilliseconds());
UpdateJitterDelay();
PropagateDecodability(next_frame_it->second);
AdvanceLastDecodedFrame(next_frame_it);
*frame_out = std::move(frame);
@ -364,6 +364,16 @@ bool FrameBuffer::UpdateFrameInfoWithIncomingFrame(const FrameObject& frame,
return true;
}
void FrameBuffer::UpdateJitterDelay() {
int unused;
int delay;
timing_->GetTimings(&unused, &unused, &unused, &unused, &delay, &unused,
&unused);
accumulated_delay_ += delay;
++accumulated_delay_samples_;
}
void FrameBuffer::UpdateHistograms() const {
rtc::CritScope lock(&crit_);
if (num_total_frames_ > 0) {
@ -373,6 +383,11 @@ void FrameBuffer::UpdateHistograms() const {
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_);
}
}
} // namespace video_coding

12
webrtc/modules/video_coding/frame_buffer2.h
View File

@ -139,6 +139,8 @@ class FrameBuffer {
FrameMap::iterator info)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
void UpdateJitterDelay() EXCLUSIVE_LOCKS_REQUIRED(crit_);
void UpdateHistograms() const;
FrameMap frames_ GUARDED_BY(crit_);
@ -155,10 +157,16 @@ class FrameBuffer {
int num_frames_buffered_ GUARDED_BY(crit_);
bool stopped_ GUARDED_BY(crit_);
VCMVideoProtection protection_mode_ GUARDED_BY(crit_);
int num_total_frames_ GUARDED_BY(crit_);
int num_key_frames_ GUARDED_BY(crit_);
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

13
webrtc/video/receive_statistics_proxy.cc
View File

@ -15,6 +15,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/system_wrappers/include/metrics.h"
namespace webrtc {
@ -92,10 +93,14 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
if (decode_ms != -1)
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
if (jb_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
jb_delay_ms);
if (field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") !=
"Enabled") {
int jb_delay_ms =
jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
if (jb_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
jb_delay_ms);
}
}
int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
if (target_delay_ms != -1) {