Add chromium support for MediaStreamAudioDestinationNode

content/renderer/media/webrtc_audio_capturer.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/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
@@ skipping 82 matching lines @@
   base::AutoLock auto_lock(lock_);
   for (SinkList::iterator it = sinks_.begin(); it != sinks_.end(); ++it) {
     if (sink == *it) {
       sinks_.erase(it);
       break;
     }
   }
 }
 
 void WebRtcAudioCapturer::SetCapturerSource(
-    const scoped_refptr<media::AudioCapturerSource>& source) {
+    const scoped_refptr<media::AudioCapturerSource>& source,
+    media::ChannelLayout channel_layout,
+    float sample_rate) {
   DVLOG(1) <<  "SetCapturerSource()";
   scoped_refptr<media::AudioCapturerSource> old_source;
   {
     base::AutoLock auto_lock(lock_);
     if (source_ == source)
       return;
 
     source_.swap(old_source);
     source_ = source;
   }
 
   // Detach the old source from normal recording.
-  if (old_source)
+  if (old_source) {
     old_source->Stop();
 
+    // Dispatch the new parameters both to the sink(s) and to the new source.
+    // The idea is to get rid of any dependency of the microphone parameters
+    // which would normally be used by default.
+
+    int buffer_size = GetBufferSizeForSampleRate(sample_rate);
+    if (!buffer_size) {
+      DLOG(ERROR) << "Unsupported sample-rate: " << sample_rate;
+      return;
+    }
+
+    params_.Reset(params_.format(),
+                  channel_layout,
+                  sample_rate,
+                  16,  // ignored since the audio stack uses float32.
+                  buffer_size);
+
+    buffer_.reset(new int16[params_.frames_per_buffer() * params_.channels()]);
+
+    for (SinkList::const_iterator it = sinks_.begin();
+         it != sinks_.end(); ++it) {
+      (*it)->SetCaptureFormat(params_);
+    }
+  }
+
   if (source)
     source->Initialize(params_, this, this);
 }
 
 void WebRtcAudioCapturer::SetStopCallback(
     const base::Closure& on_device_stopped_cb) {
   DVLOG(1) <<  "WebRtcAudioCapturer::SetStopCallback()";
   base::AutoLock auto_lock(lock_);
   on_device_stopped_cb_ = on_device_stopped_cb;
 }
@@ skipping 76 matching lines @@
     return false;
   }
 
   params_.Reset(format, channel_layout, sample_rate, 16, buffer_size);
 
   buffer_.reset(new int16[params_.frames_per_buffer() * params_.channels()]);
 
   // Create and configure the default audio capturing source. The |source_|
   // will be overwritten if the client call the source calls
   // SetCapturerSource().
-  SetCapturerSource(AudioDeviceFactory::NewInputDevice());
+  SetCapturerSource(
+      AudioDeviceFactory::NewInputDevice(), channel_layout, sample_rate);
 
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
                             channel_layout, media::CHANNEL_LAYOUT_MAX);
 
   return true;
 }
 
 void WebRtcAudioCapturer::ProvideInput(media::AudioBus* dest) {
   base::AutoLock auto_lock(lock_);
   DCHECK(loopback_fifo_.get() != NULL);
@@ skipping 144 matching lines @@
   // Inform the local renderer about the stopped device.
   // The renderer can then save resources by not asking for more data from
   // the stopped source. We are on the IO thread but the callback task will
   // be posted on the message loop of the main render thread thanks to
   // usage of BindToLoop() when the callback was initialized.
   if (!on_device_stopped_cb_.is_null())
     on_device_stopped_cb_.Run();
 }
 
 }  // namespace content