xref: /aosp_15_r20/external/webrtc/api/call/audio_sink.h (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1 /*
2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef API_CALL_AUDIO_SINK_H_
12 #define API_CALL_AUDIO_SINK_H_
13 
14 #include <stddef.h>
15 #include <stdint.h>
16 
17 namespace webrtc {
18 
19 // Represents a simple push audio sink.
20 class AudioSinkInterface {
21  public:
~AudioSinkInterface()22   virtual ~AudioSinkInterface() {}
23 
24   struct Data {
DataData25     Data(const int16_t* data,
26          size_t samples_per_channel,
27          int sample_rate,
28          size_t channels,
29          uint32_t timestamp)
30         : data(data),
31           samples_per_channel(samples_per_channel),
32           sample_rate(sample_rate),
33           channels(channels),
34           timestamp(timestamp) {}
35 
36     const int16_t* data;         // The actual 16bit audio data.
37     size_t samples_per_channel;  // Number of frames in the buffer.
38     int sample_rate;             // Sample rate in Hz.
39     size_t channels;             // Number of channels in the audio data.
40     uint32_t timestamp;          // The RTP timestamp of the first sample.
41   };
42 
43   virtual void OnData(const Data& audio) = 0;
44 };
45 
46 }  // namespace webrtc
47 
48 #endif  // API_CALL_AUDIO_SINK_H_
49