Revert of 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
new file mode 100644
index 0000000000000000000000000000000000000000..999644f4ce161946e55e59645d29d043b723c0c8
--- /dev/null
+++ b/webrtc/audio/audio_sink.h
@@ -0,0 +1,53 @@
+/*
+ *  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_