Express audio delay more precisely in frames rather than milliseconds.

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 <utility>
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
@@ skipping 405 matching lines @@
 
 media::OutputDeviceStatus WebRtcAudioRenderer::GetDeviceStatus() {
   DCHECK(thread_checker_.CalledOnValidThread());
   if (!sink_.get())
     return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL;
 
   return sink_->GetDeviceStatus();
 }
 
 int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
-                                uint32_t audio_delay_milliseconds,
+                                uint32_t frames_delayed,
                                 uint32_t frames_skipped) {
   DCHECK(audio_renderer_thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   if (!source_)
     return 0;
 
+  // TODO(grunell): Converting from frames to milliseconds will potentially lose
+  // hundreds of microseconds which may cause audio video drift. Update
+  // this class and all usage of render delay msec -> frames (possibly even
+  // using a double type for frames). See http://crbug.com/586540
+  uint32_t audio_delay_milliseconds = static_cast<double>(frames_delayed) *
+                                      base::Time::kMillisecondsPerSecond /
+                                      sink_params_.sample_rate();
+
   DVLOG(2) << "WebRtcAudioRenderer::Render()";
   DVLOG(2) << "audio_delay_milliseconds: " << audio_delay_milliseconds;
 
   DCHECK_LE(audio_delay_milliseconds, static_cast<uint32_t>(INT_MAX));
   audio_delay_milliseconds_ = static_cast<int>(audio_delay_milliseconds);
 
   // If there are skipped frames, pull and throw away the same amount. We always
   // pull 10 ms of data from the source (see PrepareSink()), so the fifo is only
   // required if the number of frames to drop doesn't correspond to 10 ms.
   if (frames_skipped > 0) {
@@ skipping 31 matching lines @@
 // Called by AudioPullFifo when more data is necessary.
 void WebRtcAudioRenderer::SourceCallback(
     int fifo_frame_delay, media::AudioBus* audio_bus) {
   DCHECK(audio_renderer_thread_checker_.CalledOnValidThread());
   base::TimeTicks start_time = base::TimeTicks::Now();
   DVLOG(2) << "WebRtcAudioRenderer::SourceCallback("
            << fifo_frame_delay << ", "
            << audio_bus->frames() << ")";
 
   int output_delay_milliseconds = audio_delay_milliseconds_;
+  // TODO(grunell): This integer division by sample_rate will cause loss of
+  // partial milliseconds, and may cause avsync drift. http://crbug.com/586540
   output_delay_milliseconds += fifo_frame_delay *
                                base::Time::kMillisecondsPerSecond /
                                sink_params_.sample_rate();
   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_);
@@ skipping 169 matching lines @@
           base::Bind(&WebRtcAudioRenderer::SourceCallback,
                      base::Unretained(this))));
     }
     sink_params_ = new_sink_params;
   }
 
   sink_->Initialize(new_sink_params, this);
 }
 
 }  // namespace content