| /* |
| * 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_API_CALL_AUDIO_SINK_H_ |
| #define WEBRTC_API_CALL_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_API_CALL_AUDIO_SINK_H_ |