Upstream version 10.39.225.0

[platform/framework/web/crosswalk.git] / src / content / renderer / media / audio_renderer_mixer_manager.cc

diff --git a/src/content/renderer/media/audio_renderer_mixer_manager.cc b/src/content/renderer/media/audio_renderer_mixer_manager.cc
index 0044d25..4a62d1a 100644 (file)
--- a/src/content/renderer/media/audio_renderer_mixer_manager.cc
+++ b/src/content/renderer/media/audio_renderer_mixer_manager.cc
@@ -21,15 +21,18 @@ AudioRendererMixerManager::AudioRendererMixerManager(
 }
 
 AudioRendererMixerManager::~AudioRendererMixerManager() {
-  DCHECK(mixers_.empty());
+  // References to AudioRendererMixers may be owned by garbage collected
+  // objects.  During process shutdown they may be leaked, so, transitively,
+  // |mixers_| may leak (i.e., may be non-empty at this time) as well.
 }
 
 media::AudioRendererMixerInput* AudioRendererMixerManager::CreateInput(
-    int source_render_view_id) {
+    int source_render_view_id, int source_render_frame_id) {
   return new media::AudioRendererMixerInput(
       base::Bind(
           &AudioRendererMixerManager::GetMixer, base::Unretained(this),
-          source_render_view_id),
+          source_render_view_id,
+          source_render_frame_id),
       base::Bind(
           &AudioRendererMixerManager::RemoveMixer, base::Unretained(this),
           source_render_view_id));
@@ -42,7 +45,11 @@ void AudioRendererMixerManager::SetAudioRendererSinkForTesting(
 
 media::AudioRendererMixer* AudioRendererMixerManager::GetMixer(
     int source_render_view_id,
+    int source_render_frame_id,
     const media::AudioParameters& params) {
+  // Effects are not passed through to output creation, so ensure none are set.
+  DCHECK_EQ(params.effects(), media::AudioParameters::NO_EFFECTS);
+
   const MixerKey key(source_render_view_id, params);
   base::AutoLock auto_lock(mixers_lock_);
 
@@ -52,9 +59,9 @@ media::AudioRendererMixer* AudioRendererMixerManager::GetMixer(
     return it->second.mixer;
   }
 
-  // On ChromeOS and Linux we can rely on the playback device to handle
-  // resampling, so don't waste cycles on it here.
-#if defined(OS_CHROMEOS) || defined(OS_LINUX)
+  // On ChromeOS we can rely on the playback device to handle resampling, so
+  // don't waste cycles on it here.
+#if defined(OS_CHROMEOS)
   int sample_rate = params.sample_rate();
 #else
   int sample_rate = hardware_config_->GetOutputSampleRate();
@@ -62,10 +69,10 @@ media::AudioRendererMixer* AudioRendererMixerManager::GetMixer(
 
   // Create output parameters based on the audio hardware configuration for
   // passing on to the output sink.  Force to 16-bit output for now since we
-  // know that works well for WebAudio and WebRTC.
+  // know that works everywhere; ChromeOS does not support other bit depths.
   media::AudioParameters output_params(
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY, params.channel_layout(),
-      sample_rate, 16, hardware_config_->GetOutputBufferSize());
+      sample_rate, 16, hardware_config_->GetHighLatencyBufferSize());
 
   // If we've created invalid output parameters, simply pass on the input params
   // and let the browser side handle automatic fallback.
@@ -79,7 +86,7 @@ media::AudioRendererMixer* AudioRendererMixerManager::GetMixer(
   } else {
     mixer = new media::AudioRendererMixer(
         params, output_params, AudioDeviceFactory::NewOutputDevice(
-            source_render_view_id));
+            source_render_view_id, source_render_frame_id));
   }
 
   AudioRendererMixerReference mixer_reference = { mixer, 1 };