Adds time measurement of WebRtcAudioRenderer::Render

media/audio/audio_output_device.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 "media/audio/audio_output_device.h"
 
 #include "base/basictypes.h"
 #include "base/debug/trace_event.h"
 #include "base/threading/thread_restrictions.h"
 #include "base/time/time.h"
 #include "media/audio/audio_output_controller.h"
 #include "media/base/limits.h"
+#include "media/base/scoped_histogram_timer.h"
 
 namespace media {
 
 // Takes care of invoking the render callback on the audio thread.
 // An instance of this class is created for each capture stream in
 // OnStreamCreated().
 class AudioOutputDevice::AudioThreadCallback
     : public AudioDeviceThread::Callback {
  public:
   AudioThreadCallback(const AudioParameters& audio_parameters,
@@ skipping 260 matching lines @@
   output_bus_ =
       AudioBus::WrapMemory(audio_parameters_, shared_memory_.memory());
 }
 
 // Called whenever we receive notifications about pending data.
 void AudioOutputDevice::AudioThreadCallback::Process(int pending_data) {
   // Negative |pending_data| indicates the browser side stream has stopped.
   if (pending_data < 0)
     return;
 
+  SCOPED_UMA_HISTOGRAM_TIMER("Media.AudioOutputDevice.ProcessTime");

[no longer working on chromium, 2014/09/17 14:54:50]

I would suggest moving the code to webrtc_audio_renderer.cc, and how about
logging it every 10 callbacks (every 100ms)?

[henrika (OOO until Aug 14), 2014/09/18 10:56:47]

I could move it but AFAIK it is not possible to log at a lower frequency due to
the design of SCOPED_UMA_HISTOGRAM_TIMER. It will only measure the elapsed time
int scope and I can't figure out a way to do so and at the same time check a
counter. If so, I would have to add my time measurement manually instead.

[henrika (OOO until Aug 14), 2014/09/18 11:03:16]

Let me see if I can do something without the macro instead.

+
   // Convert the number of pending bytes in the render buffer into milliseconds.
   int audio_delay_milliseconds = pending_data / bytes_per_ms_;
 
   TRACE_EVENT0("audio", "AudioOutputDevice::FireRenderCallback");
 
   // Update the audio-delay measurement then ask client to render audio.  Since
   // |output_bus_| is wrapping the shared memory the Render() call is writing
   // directly into the shared memory.
   render_callback_->Render(output_bus_.get(), audio_delay_milliseconds);
 }
 
 }  // namespace media.