Add generic interfaces for the sinks of the media stream audio track

content/public/renderer/media_stream_audio_sink.h

Index: content/public/renderer/media_stream_audio_sink.h
diff --git a/content/public/renderer/media_stream_audio_sink.h b/content/public/renderer/media_stream_audio_sink.h
new file mode 100644
index 0000000000000000000000000000000000000000..079c086d859045c26757ec45a6c3e6ce555df9c1
--- /dev/null
+++ b/content/public/renderer/media_stream_audio_sink.h
@@ -0,0 +1,58 @@
+// 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/common/content_export.h"
+
+namespace blink {
+class WebMediaStreamTrack;
+}
+
+namespace content {
+
+class CONTENT_EXPORT MediaStreamAudioSink {

[no longer working on chromium, 2013/11/27 13:36:20]

it will inherit MediaStreamSink after Per lands his CL.

+  // Callback on delivering the audio interleaved data.
+  // |audio_data| is the pointer to the audio data.
+  // |sample_rate| is the sample frequency of audio data.
+  // |number_of_channels| is the number of channels reflecting the order of
+  // surround sound channels.
+  // |number_of_frames| is the number of audio freams in the buffer.

[perkj_chrome, 2013/11/27 14:56:40]

spelling

[no longer working on chromium, 2013/11/27 16:27:57]

Done.

+  // |channels| contains a vector of WebRtc VoE channels.
+  // |audio_delay_milliseconds| is recording delay value.
+  // |current_volume| is current microphone volume, in range of |0, 255].
+  // |need_audio_processing| indicates if the audio needs WebRtc AEC/NS/AGC
+  // audio processing.
+  // |key_pressed| is the flag to help typing detection.
+  // Note, |channels|, |audio_delay_milliseconds|, |current_volume|,
+  // |need_audio_processing| and |key_pressed| are needed by WebRtc, for other
+  // clients, those variables can be ignore.
+  // TODO(xians): Remove those WebRtc specific variables after moving the

[no longer working on chromium, 2013/11/27 13:36:20]

I hope these comments and TODO can explain why we need to keep these webrtc
specific variables for the short term.
And we are working hard to move the APM to chrome now, after this is done, the
callback interface will be much cleaned.

+  // APM from WebRtc to Chrome.
+  // The return value is the new microphone volume, in the range of |0, 255].
+  // When the volume does not need to be updated, it returns 0.
+  virtual int OnData(const int16* audio_data,
+                     int sample_rate,
+                     int number_of_channels,
+                     int number_of_frames,
+                     const std::vector<int>& channels,
+                     int audio_delay_milliseconds,
+                     int current_volume,
+                     bool need_audio_processing,
+                     bool key_pressed) = 0;
+
+  // Callback called when the forma of the audio stream has changed, and
+  // it will be called on the same thread as calling OnData().
+  virtual void OnSetFormat(const media::AudioParameters& params) = 0;
+
+ protected:
+  virtual ~MediaStreamAudioSink() {}
+};
+
+CONTENT_EXPORT void AddToAudioTrack(MediaStreamAudioSink* sink,
+                                    const blink::WebMediaStreamTrack& track);
+
+CONTENT_EXPORT void RemoveFromAudioTrack(
+    MediaStreamAudioSink* sink, const blink::WebMediaStreamTrack& track);
+
+}  // namespace content