changed MSAP to work with 96000KHZ

content/renderer/media/media_stream_audio_processor.cc

Index: content/renderer/media/media_stream_audio_processor.cc
diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc
index 460ab8baf4e653d6418f99735d229ab97cddd6b6..2c64d936c8c95a959a6f0a7bb8b8e63fb714aaa9 100644
--- a/content/renderer/media/media_stream_audio_processor.cc
+++ b/content/renderer/media/media_stream_audio_processor.cc
@@ -70,6 +70,13 @@ class MediaStreamAudioProcessor::MediaStreamAudioConverter
     int buffer_size = std::max(
         kMaxNumberOfBuffersInFifo * source_params_.frames_per_buffer(),
         kMaxNumberOfBuffersInFifo * sink_params_.frames_per_buffer());
+
+    // Also, FIFO needs to have enough space to store pre-processed data

[henrika (OOO until Aug 14), 2014/06/02 08:23:41]

Is it possible to set the buffer size once only and not twice?

[no longer working on chromium, 2014/06/02 09:15:08]

Yes, I added a max_frame_size as a temp variable to make it possible.

+    // before passing the data to webrtc::AudioProcessing, which requires 10ms
+    // as packet size.
+    buffer_size = std::max(
+        buffer_size,
+        kMaxNumberOfBuffersInFifo * source_params_.sample_rate() / 100);
     fifo_.reset(new media::AudioFifo(source_params_.channels(), buffer_size));
     // TODO(xians): Use CreateWrapper to save one memcpy.
     audio_wrapper_ = media::AudioBus::Create(sink_params_.channels(),