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}
diff --git a/webrtc/modules/video_coding/frame_buffer2.cc b/webrtc/modules/video_coding/frame_buffer2.cc
index db4928c..279c613 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,
@@ -163,8 +162,9 @@
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 =
@@ -365,22 +365,28 @@
}
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_);
}
}