changed MSAP to work with 96000KHZ

content/renderer/media/media_stream_audio_processor.cc

 // Copyright 2013 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/media_stream_audio_processor.h"
 
 #include "base/command_line.h"
 #include "base/debug/trace_event.h"
 #include "base/metrics/histogram.h"
 #include "content/public/common/content_switches.h"
@@ skipping 52 matching lines @@
     // render thread and used in the audio thread, for example, the
     // |MediaStreamAudioProcessor::capture_converter_|.
     thread_checker_.DetachFromThread();
     audio_converter_.AddInput(this);
     // Create and initialize audio fifo and audio bus wrapper.
     // The size of the FIFO should be at least twice of the source buffer size
     // or twice of the sink buffer size.
     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(),
                                              sink_params_.frames_per_buffer());
   }
 
   virtual ~MediaStreamAudioConverter() {
     audio_converter_.RemoveInput(this);
   }
 
@@ skipping 408 matching lines @@
 
   StopAecDump();
 
   if (playout_data_source_)
     playout_data_source_->RemovePlayoutSink(this);
 
   audio_processing_.reset();
 }
 
 }  // namespace content