Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.

webrtc/audio/audio_sink.h

Index: webrtc/audio/audio_sink.h
diff --git a/webrtc/audio/audio_sink.h b/webrtc/audio/audio_sink.h
deleted file mode 100644
index 999644f4ce161946e55e59645d29d043b723c0c8..0000000000000000000000000000000000000000
--- a/webrtc/audio/audio_sink.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_AUDIO_SINK_H_
-#define WEBRTC_AUDIO_AUDIO_SINK_H_
-
-#if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
-// Avoid conflict with format_macros.h.
-#define __STDC_FORMAT_MACROS
-#endif
-
-#include <inttypes.h>
-#include <stddef.h>
-
-namespace webrtc {
-
-// Represents a simple push audio sink.
-class AudioSinkInterface {
- public:
-  virtual ~AudioSinkInterface() {}
-
-  struct Data {
-    Data(int16_t* data,
-         size_t samples_per_channel,
-         int sample_rate,
-         size_t channels,
-         uint32_t timestamp)
-        : data(data),
-          samples_per_channel(samples_per_channel),
-          sample_rate(sample_rate),
-          channels(channels),
-          timestamp(timestamp) {}
-
-    int16_t* data;               // The actual 16bit audio data.
-    size_t samples_per_channel;  // Number of frames in the buffer.
-    int sample_rate;             // Sample rate in Hz.
-    size_t channels;             // Number of channels in the audio data.
-    uint32_t timestamp;          // The RTP timestamp of the first sample.
-  };
-
-  virtual void OnData(const Data& audio) = 0;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_AUDIO_AUDIO_SINK_H_