Merge 123292 - Increase the buffer size in AudioRendererImpl to fix muted playback rate

content/renderer/media/audio_renderer_impl.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/audio_renderer_impl.h"
 
 #include <math.h>
 
 #include <algorithm>
 
 #include "base/bind.h"
 #include "content/common/child_process.h"
 #include "content/common/media/audio_messages.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_buffers_state.h"
 #include "media/audio/audio_util.h"
 #include "media/base/filter_host.h"
 
-// We define GetBufferSizeForSampleRate() instead of using
-// GetAudioHardwareBufferSize() in audio_util because we're using
-// the AUDIO_PCM_LINEAR flag, instead of AUDIO_PCM_LOW_LATENCY,
-// which the audio_util functions assume.
-//
-// See: http://code.google.com/p/chromium/issues/detail?id=103627
-// for a more detailed description of the subtleties.
-static size_t GetBufferSizeForSampleRate(int sample_rate) {
-  // kNominalBufferSize has been tested on Windows, Mac OS X, and Linux
-  // using the low-latency audio codepath (SyncSocket implementation)
-  // with the AUDIO_PCM_LINEAR flag.
-  const size_t kNominalBufferSize = 2048;
-
-  if (sample_rate <= 48000)
-    return kNominalBufferSize;
-  else if (sample_rate <= 96000)
-    return kNominalBufferSize * 2;
-  return kNominalBufferSize * 4;
-}
-
 AudioRendererImpl::AudioRendererImpl(media::AudioRendererSink* sink)
     : AudioRendererBase(),
       bytes_per_second_(0),
       stopped_(false),
       sink_(sink),
       is_initialized_(false) {
 }
 
 AudioRendererImpl::~AudioRendererImpl() {
 }
@@ skipping 35 matching lines @@
                                       sample_rate,
                                       bits_per_channel,
                                       0);
 
   bytes_per_second_ = audio_parameters_.GetBytesPerSecond();
 
   DCHECK(sink_.get());
 
   if (!is_initialized_) {
     sink_->Initialize(
-        GetBufferSizeForSampleRate(sample_rate),
+        media::SelectSamplesPerPacket(sample_rate),
         audio_parameters_.channels,
         audio_parameters_.sample_rate,
         audio_parameters_.format,
         this);
 
     sink_->Start();
     is_initialized_ = true;
     return true;
   }
 
@@ skipping 142 matching lines @@
 
 void AudioRendererImpl::OnError() {
   host()->DisableAudioRenderer();
 }
 
 void AudioRendererImpl::OnRenderEndOfStream() {
   // TODO(enal): schedule callback instead of polling.
   if (base::Time::Now() >= earliest_end_time_)
     SignalEndOfStream();
 }