Line data    Source code

       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 WEBRTC_API_CALL_AUDIO_SINK_H_
      12             : #define WEBRTC_API_CALL_AUDIO_SINK_H_
      13             : 
      14             : #if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
      15             : // Avoid conflict with format_macros.h.
      16             : #define __STDC_FORMAT_MACROS
      17             : #endif
      18             : 
      19             : #include <inttypes.h>
      20             : #include <stddef.h>
      21             : 
      22             : namespace webrtc {
      23             : 
      24             : // Represents a simple push audio sink.
      25             : class AudioSinkInterface {
      26             :  public:
      27             :   virtual ~AudioSinkInterface() {}
      28             : 
      29             :   struct Data {
      30           0 :     Data(int16_t* data,
      31             :          size_t samples_per_channel,
      32             :          int sample_rate,
      33             :          size_t channels,
      34             :          uint32_t timestamp)
      35           0 :         : data(data),
      36             :           samples_per_channel(samples_per_channel),
      37             :           sample_rate(sample_rate),
      38             :           channels(channels),
      39           0 :           timestamp(timestamp) {}
      40             : 
      41             :     int16_t* data;               // The actual 16bit audio data.
      42             :     size_t samples_per_channel;  // Number of frames in the buffer.
      43             :     int sample_rate;             // Sample rate in Hz.
      44             :     size_t channels;             // Number of channels in the audio data.
      45             :     uint32_t timestamp;          // The RTP timestamp of the first sample.
      46             :   };
      47             : 
      48             :   virtual void OnData(const Data& audio) = 0;
      49             : };
      50             : 
      51             : }  // namespace webrtc
      52             : 
      53             : #endif  // WEBRTC_API_CALL_AUDIO_SINK_H_