Fix wrong timing when audio is muted.

media/audio/audio_output_mixer.cc

Index: media/audio/audio_output_mixer.cc
===================================================================
--- media/audio/audio_output_mixer.cc	(revision 140286)
+++ media/audio/audio_output_mixer.cc	(working copy)
@@ -16,6 +16,9 @@
 
 namespace media {
 
+// Align audio data to 64-byte boundary. That is current max AVX can use.
+const size_t kMixerDataAlignment = 64;
+
 AudioOutputMixer::AudioOutputMixer(AudioManager* audio_manager,
                                    const AudioParameters& params,
                                    const base::TimeDelta& close_delay)
@@ -26,8 +29,11 @@
                    weak_this_.GetWeakPtr(),
                    &AudioOutputMixer::ClosePhysicalStream),
       pending_bytes_(0) {
-  // TODO(enal): align data.
-  mixer_data_.reset(new uint8[params_.GetBytesPerBuffer()]);
+  mixer_data_.reset(

[DaleCurtis, 2012/06/04 20:48:12]

Drive by: if you don't mind a 32 byte alignment you could back this with a
DecoderBuffer which uses FFmpeg alignment and does the right thing on say ARM
platforms. Maybe I should have named it AlignedBuffer :)

+      new uint8[params_.GetBytesPerBuffer() + kMixerDataAlignment - 1]);
+  aligned_mixer_data_ = reinterpret_cast<uint8*>(
+      (reinterpret_cast<size_t>(mixer_data_.get()) + kMixerDataAlignment - 1) &
+      ~(kMixerDataAlignment - 1));
 }
 
 AudioOutputMixer::~AudioOutputMixer() {
@@ -197,8 +203,6 @@
 
     // No need to mix muted stream.
     double volume = proxy_data->volume;
-    if (volume == 0.0)
-      continue;
 
     // Different handling for first and all subsequent streams.
     if (first_stream) {
@@ -212,7 +216,7 @@
       if (actual_size < max_size)
         memset(dest + actual_size, 0, max_size - actual_size);
       first_stream = false;
-      actual_dest = mixer_data_.get();
+      actual_dest = aligned_mixer_data_;
       actual_total_size = actual_size;
     } else {
       media::MixStreams(dest,