webrtc/api/call/audio_sink.h
Mirko Bonadei 9d9c2d5795 Make header files self contained.
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.

Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
2022-10-08 08:38:36 +00:00

48 lines
1.4 KiB
C++

/*
* 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 API_CALL_AUDIO_SINK_H_
#define API_CALL_AUDIO_SINK_H_
#include <stddef.h>
#include <stdint.h>
namespace webrtc {
// Represents a simple push audio sink.
class AudioSinkInterface {
public:
virtual ~AudioSinkInterface() {}
struct Data {
Data(const 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) {}
const 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 // API_CALL_AUDIO_SINK_H_