MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc/peer_connection_remote_audio_source.h

Index: content/renderer/media/webrtc/peer_connection_remote_audio_source.h
diff --git a/content/renderer/media/webrtc/peer_connection_remote_audio_source.h b/content/renderer/media/webrtc/peer_connection_remote_audio_source.h
new file mode 100644
index 0000000000000000000000000000000000000000..cfa459f590d19a022c81da7c73f9793b7ae0bee7
--- /dev/null
+++ b/content/renderer/media/webrtc/peer_connection_remote_audio_source.h
@@ -0,0 +1,73 @@
+// Copyright 2015 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.
+
+#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
+#define CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
+
+#include "base/memory/ref_counted.h"
+#include "content/renderer/media/media_stream_audio_track.h"
+#include "content/renderer/media/media_stream_audio_source.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
+
+namespace content {
+
+// PeerConnectionRemoteAudioTrack is a WebRTC specific implementation of an
+// audio track whose data is sourced from a PeerConnection.
+class PeerConnectionRemoteAudioTrack : public MediaStreamAudioTrack {
+ public:
+  explicit PeerConnectionRemoteAudioTrack(
+      const scoped_refptr<webrtc::AudioTrackInterface>& track_interface);
+  ~PeerConnectionRemoteAudioTrack() final;
+
+  // If |track| is an instance of PeerConnectionRemoteAudioTrack, return a
+  // type-casted pointer to it.  Otherwise, return null.
+  static PeerConnectionRemoteAudioTrack* From(MediaStreamAudioTrack* track);
+
+  webrtc::AudioTrackInterface* track_interface() const {
+    return track_interface_.get();
+  }
+
+  // MediaStreamAudioTrack override.
+  void SetEnabled(bool enabled) final;
+  void* GetClassIdentifier() const final;
+
+ private:
+  const scoped_refptr<webrtc::AudioTrackInterface> track_interface_;
+};
+
+// Represents the audio provided by the receiving end of a PeerConnection.
+class PeerConnectionRemoteAudioSource
+    : NON_EXPORTED_BASE(public MediaStreamAudioSource),
+      NON_EXPORTED_BASE(protected webrtc::AudioTrackSinkInterface) {
+ public:
+  explicit PeerConnectionRemoteAudioSource(
+      const scoped_refptr<webrtc::AudioTrackInterface>& track_interface);
+  ~PeerConnectionRemoteAudioSource() final;
+
+ protected:
+  // MediaStreamAudioSource overrides.
+  scoped_ptr<MediaStreamAudioTrack> CreateMediaStreamAudioTrack(
+      const std::string& id) final;
+  void DoStopSource() final;
+  bool EnsureSourceIsStarted() final;
+
+  // webrtc::AudioTrackSinkInterface implementation.
+  void OnData(const void* audio_data, int bits_per_sample, int sample_rate,
+              size_t number_of_channels, size_t number_of_frames) final;
+
+ private:
+  // Interface to the implementation that calls OnData().
+  const scoped_refptr<webrtc::AudioTrackInterface> track_interface_;
+
+  // True if |this| has been registered as a sink via |track_interface_|.
+  bool is_started_;
+
+  // Buffer for converting from interleaved int16 PCM samples to the planar
+  // float format.  Only used on the thread that calls OnData().
+  scoped_ptr<media::AudioBus> audio_bus_;
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_