Support for remote audio into tracks

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
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index 0000000000000000000000000000000000000000..b5a02ca013750710ad2ca1ce6d2cd690fd8cd75b
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+/*
+ *  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_
+
+#include <inttypes.h>
+#include <stddef.h>
+
+namespace webrtc {
+
+// Represents a simple push audio sink.
+class AudioSink {

[the sun, 2015/12/11 13:52:01]

I thought the style guide said interface classes must end in "Interface"?

[tommi (sloooow) - chröme, 2015/12/11 14:22:57]

Done.

+ public:
+  virtual ~AudioSink() {}
+
+  struct Data {
+    Data(uint8_t* data,
+         size_t samples_per_channel,
+         int sample_rate,
+         int channels,
+         uint32_t timestamp)
+        : data(data),
+          samples_per_channel(samples_per_channel),
+          sample_rate(sample_rate),
+          channels(channels),
+          timestamp(timestamp) {}
+
+    uint8_t* data;               // The actual audio data.
+    int bits_per_sample = 16;    // Bits per sample (currently always 16bit).

[the sun, 2015/12/11 13:52:01]

Either this should be an enum (float, int16, int32, interleave, deinterleave...)
or, more likely, we should just pass floats, always. Make bits_per_sample const
or hard code in your AudioSink->AudioTrackSinkInterface adapter.

[tommi (sloooow) - chröme, 2015/12/11 14:22:57]

I'm mimicking what's being done in Chromium here but maybe I should just skip
that and have audio data fixed as int16_t* for now.  I agree that it would be
better to switch to floats and have a better way to represent the format of the
audio, but I think it's a little bit outside the scope of this cl.

[the sun, 2015/12/11 16:32:04]

Agreed.

+    size_t samples_per_channel;  // Number of frames in the buffer.
+    int sample_rate;             // Sample rate in Hz.
+    int 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_