Make the new jitter buffer the default jitter buffer.

webrtc/video/receive_statistics_proxy.cc

Index: webrtc/video/receive_statistics_proxy.cc
diff --git a/webrtc/video/receive_statistics_proxy.cc b/webrtc/video/receive_statistics_proxy.cc
index 3f23c0047a51bf80f81c2aa01e658e9e487a9167..46c97a3cf417b96751710f1cf747af4575a0830f 100644
--- a/webrtc/video/receive_statistics_proxy.cc
+++ b/webrtc/video/receive_statistics_proxy.cc
@@ -10,7 +10,9 @@
 
 #include "webrtc/video/receive_statistics_proxy.h"
 
+#include <algorithm>
 #include <cmath>
+#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
@@ -39,6 +41,9 @@ const int kLowQpThresholdVp8 = 60;
 const int kHighQpThresholdVp8 = 70;
 const int kLowVarianceThreshold = 1;
 const int kHighVarianceThreshold = 2;
+
+// How large window we use to calculate the framerate/bitrate.
+const int kRateStatisticsWindowSizeMs = 1000;
 }  // namespace
 
 ReceiveStatisticsProxy::ReceiveStatisticsProxy(
@@ -66,7 +71,8 @@ ReceiveStatisticsProxy::ReceiveStatisticsProxy(
       render_fps_tracker_(100, 10u),
       render_pixel_tracker_(100, 10u),
       freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs),
-      first_report_block_time_ms_(-1) {
+      first_report_block_time_ms_(-1),
+      avg_rtt_ms_(0) {
   stats_.ssrc = config_.rtp.remote_ssrc;
   for (auto it : config_.rtp.rtx)
     rtx_stats_[it.second.ssrc] = StreamDataCounters();
@@ -118,6 +124,17 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
                  << freq_offset_stats.ToString();
   }
 
+  if (stats_.frame_counts.key_frames > 0 ||
+      stats_.frame_counts.delta_frames > 0) {
+    float num_key_frames = stats_.frame_counts.key_frames;
+    float num_total_frames =
+        stats_.frame_counts.key_frames + stats_.frame_counts.delta_frames;
+    int key_frames_permille =
+        (num_key_frames * 1000.0f / num_total_frames + 0.5f);
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
+                              key_frames_permille);
+  }
+
   int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
   if (qp != -1)
     RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
@@ -129,15 +146,12 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
   if (decode_ms != -1)
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_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 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) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
@@ -264,8 +278,25 @@ void ReceiveStatisticsProxy::QualitySample() {
   qp_sample_.Reset();
 }
 
+void ReceiveStatisticsProxy::UpdateFrameAndBitrate(int64_t now_ms) const {
+  int64_t old_frames_ms = now_ms - kRateStatisticsWindowSizeMs;
+  while (!frame_window_.empty() &&
+         frame_window_.begin()->first < old_frames_ms) {
+    frame_window_accumulated_bytes_ -= frame_window_.begin()->second;
+    frame_window_.erase(frame_window_.begin());
+  }
+
+  int framerate =
+      (frame_window_.size() * 1000 + 500) / kRateStatisticsWindowSizeMs;
+  size_t bitrate_bps =
+      frame_window_accumulated_bytes_ * 8000 / kRateStatisticsWindowSizeMs;
+  stats_.network_frame_rate = framerate;
+  stats_.total_bitrate_bps = bitrate_bps;
+}
+
 VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
   rtc::CritScope lock(&crit_);
+  UpdateFrameAndBitrate(clock_->TimeInMilliseconds());
   return stats_;
 }
 
@@ -283,8 +314,6 @@ void ReceiveStatisticsProxy::OnIncomingRate(unsigned int framerate,
                                             unsigned int bitrate_bps) {
   rtc::CritScope lock(&crit_);
   QualitySample();

[philipel, 2017/01/11 16:12:34]

This function is still called about every second from the VideoReceiver::Process
thread.

This is the work Viktor did to detect bad call quality and we don't use it yet,
so maybe we want to clean up even more in VideoReceiver::Process?

[stefan-webrtc, 2017/01/12 14:16:15]

Ouch... Why is it called here? Can we move it to somewhere else where we have
periodic calls? Or just add a todo to clean up

[philipel, 2017/01/12 16:41:35]

I will take a look at it in one of my clean up CLs.

-  stats_.network_frame_rate = framerate;
-  stats_.total_bitrate_bps = bitrate_bps;
 }
 
 void ReceiveStatisticsProxy::OnDecoderTiming(int decode_ms,
@@ -293,8 +322,7 @@ void ReceiveStatisticsProxy::OnDecoderTiming(int decode_ms,
                                              int target_delay_ms,
                                              int jitter_buffer_ms,
                                              int min_playout_delay_ms,
-                                             int render_delay_ms,
-                                             int64_t rtt_ms) {
+                                             int render_delay_ms) {
   rtc::CritScope lock(&crit_);
   stats_.decode_ms = decode_ms;
   stats_.max_decode_ms = max_decode_ms;
@@ -309,7 +337,7 @@ void ReceiveStatisticsProxy::OnDecoderTiming(int decode_ms,
   current_delay_counter_.Add(current_delay_ms);
   // Network delay (rtt/2) + target_delay_ms (jitter delay + decode time +
   // render delay).
-  delay_counter_.Add(target_delay_ms + rtt_ms / 2);
+  delay_counter_.Add(target_delay_ms + avg_rtt_ms_ / 2);
 }
 
 void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
@@ -413,6 +441,20 @@ void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
                                                    uint32_t frameRate) {
 }
 
+void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
+                                             size_t size_bytes) {
+  rtc::CritScope lock(&crit_);
+  if (is_keyframe)
+    ++stats_.frame_counts.key_frames;
+  else
+    ++stats_.frame_counts.delta_frames;
+
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  frame_window_accumulated_bytes_ += size_bytes;
+  frame_window_.insert(std::make_pair(now_ms, size_bytes));
+  UpdateFrameAndBitrate(now_ms);
+}
+
 void ReceiveStatisticsProxy::OnFrameCountsUpdated(
     const FrameCounts& frame_counts) {
   rtc::CritScope lock(&crit_);
@@ -454,4 +496,10 @@ void ReceiveStatisticsProxy::SampleCounter::Reset() {
   sum = 0;
 }
 
+void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms,
+                                         int64_t max_rtt_ms) {
+  rtc::CritScope lock(&crit_);
+  avg_rtt_ms_ = avg_rtt_ms;
+}
+
 }  // namespace webrtc