Adds time measurement of WebRtcAudioRenderer::Render

content/renderer/media/webrtc_audio_renderer.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
@@ skipping 11 matching lines @@
 
 #if defined(OS_WIN)
 #include "base/win/windows_version.h"
 #include "media/audio/win/core_audio_util_win.h"
 #endif
 
 namespace content {
 
 namespace {
 
+// We add a UMA histogram measuring the execution time of the Render() method
+// every |kNumCallbacksBetweenRenderTimeHistograms| callback. Assuming 10ms
+// between each callback leads to one UMA update each 100ms.
+const int kNumCallbacksBetweenRenderTimeHistograms = 10;
+
 // This is a simple wrapper class that's handed out to users of a shared
 // WebRtcAudioRenderer instance.  This class maintains the per-user 'playing'
 // and 'started' states to avoid problems related to incorrect usage which
 // might violate the implementation assumptions inside WebRtcAudioRenderer
 // (see the play reference count).
 class SharedAudioRenderer : public MediaStreamAudioRenderer {
  public:
   // Callback definition for a callback that is called when when Play(), Pause()
   // or SetVolume are called (whenever the internal |playing_state_| changes).
   typedef base::Callback<
@@ skipping 141 matching lines @@
       session_id_(session_id),
       media_stream_(media_stream),
       source_(NULL),
       play_ref_count_(0),
       start_ref_count_(0),
       audio_delay_milliseconds_(0),
       fifo_delay_milliseconds_(0),
       sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                    media::CHANNEL_LAYOUT_STEREO, sample_rate, 16,
                    frames_per_buffer,
-                   GetCurrentDuckingFlag(source_render_frame_id)) {
+                   GetCurrentDuckingFlag(source_render_frame_id)),
+      render_callback_count_(0) {
   WebRtcLogMessage(base::StringPrintf(
       "WAR::WAR. source_render_view_id=%d"
       ", session_id=%d, sample_rate=%d, frames_per_buffer=%d, effects=%i",
       source_render_view_id,
       session_id,
       sample_rate,
       frames_per_buffer,
       sink_params_.effects()));
 }
 
@@ skipping 110 matching lines @@
 }
 
 void WebRtcAudioRenderer::Play() {
   DVLOG(1) << "WebRtcAudioRenderer::Play()";
   DCHECK(thread_checker_.CalledOnValidThread());
 
   if (playing_state_.playing())
     return;
 
   playing_state_.set_playing(true);
+  render_callback_count_ = 0;
 
   OnPlayStateChanged(media_stream_, &playing_state_);
 }
 
 void WebRtcAudioRenderer::EnterPlayState() {
   DVLOG(1) << "WebRtcAudioRenderer::EnterPlayState()";
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK_GT(start_ref_count_, 0) << "Did you forget to call Start()?";
   base::AutoLock auto_lock(lock_);
   if (state_ == UNINITIALIZED)
@@ skipping 99 matching lines @@
 }
 
 void WebRtcAudioRenderer::OnRenderError() {
   NOTIMPLEMENTED();
   LOG(ERROR) << "OnRenderError()";
 }
 
 // Called by AudioPullFifo when more data is necessary.
 void WebRtcAudioRenderer::SourceCallback(
     int fifo_frame_delay, media::AudioBus* audio_bus) {
+  base::TimeTicks start_time = base::TimeTicks::Now() ;
   DVLOG(2) << "WebRtcAudioRenderer::SourceCallback("
            << fifo_frame_delay << ", "
            << audio_bus->frames() << ")";
 
   int output_delay_milliseconds = audio_delay_milliseconds_;
   output_delay_milliseconds += fifo_delay_milliseconds_;
   DVLOG(2) << "output_delay_milliseconds: " << output_delay_milliseconds;
 
   // We need to keep render data for the |source_| regardless of |state_|,
   // otherwise the data will be buffered up inside |source_|.
   source_->RenderData(audio_bus, sink_params_.sample_rate(),
                       output_delay_milliseconds,
                       &current_time_);
 
   // Avoid filling up the audio bus if we are not playing; instead
   // return here and ensure that the returned value in Render() is 0.
   if (state_ != PLAYING)
     audio_bus->Zero();
+
+  if (++render_callback_count_ == kNumCallbacksBetweenRenderTimeHistograms) {
+    base::TimeDelta elapsed = base::TimeTicks::Now() - start_time;
+    render_callback_count_ = 0;
+    UMA_HISTOGRAM_TIMES("WebRTC.AudioRenderTimes", elapsed);
+  }
 }
 
 void WebRtcAudioRenderer::UpdateSourceVolume(
     webrtc::AudioSourceInterface* source) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   // Note: If there are no playing audio renderers, then the volume will be
   // set to 0.0.
   float volume = 0.0f;
 
@@ skipping 67 matching lines @@
       if (RemovePlayingState(source, state))
         EnterPauseState();
     } else if (AddPlayingState(source, state)) {
       EnterPlayState();
     }
     UpdateSourceVolume(source);
   }
 }
 
 }  // namespace content