Fixing the webrtc local audio renderer timer between Stop() and Start().

content/renderer/media/webrtc_local_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_local_audio_renderer.h"
 
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/message_loop_proxy.h"
 #include "base/synchronization/lock.h"
@@ skipping 10 matching lines @@
     media::AudioBus* audio_bus, int audio_delay_milliseconds) {
   base::AutoLock auto_lock(thread_lock_);
 
   if (!playing_) {
     audio_bus->Zero();
     return 0;
   }
 
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render");
 
-  base::Time now = base::Time::Now();
-  total_render_time_ += now - last_render_time_;
-  last_render_time_ = now;
-
   DCHECK(loopback_fifo_.get() != NULL);
 
   // Provide data by reading from the FIFO if the FIFO contains enough
   // to fulfill the request.
   if (loopback_fifo_->frames() >= audio_bus->frames()) {
     loopback_fifo_->Consume(audio_bus, 0, audio_bus->frames());
   } else {
     audio_bus->Zero();
     // This warning is perfectly safe if it happens for the first audio
     // frames. It should not happen in a steady-state mode.
@@ skipping 22 matching lines @@
   // Push captured audio to FIFO so it can be read by a local sink.
   if (loopback_fifo_) {
     if (loopback_fifo_->frames() + number_of_frames <=
         loopback_fifo_->max_frames()) {
       scoped_ptr<media::AudioBus> audio_source = media::AudioBus::Create(
           number_of_channels, number_of_frames);
       audio_source->FromInterleaved(audio_data,
                                     audio_source->frames(),
                                     sizeof(audio_data[0]));
       loopback_fifo_->Push(audio_source.get());
+
+      base::Time now = base::Time::Now();
+      total_render_time_ += now - last_render_time_;
+      last_render_time_ = now;
     } else {
       DVLOG(1) << "FIFO is full";
     }
   }
 }
 
 void WebRtcLocalAudioRenderer::SetCaptureFormat(
     const media::AudioParameters& params) {
   audio_params_ = params;
 }
@@ skipping 131 matching lines @@
   if (!sink_)
     return base::TimeDelta();
   return total_render_time();
 }
 
 bool WebRtcLocalAudioRenderer::IsLocalRenderer() const {
   return true;
 }
 
 }  // namespace content