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/*
* Copyright (c) 2013 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 MEDIA_BASE_AUDIO_SOURCE_H_
#define MEDIA_BASE_AUDIO_SOURCE_H_
#include <cstddef>
#include "absl/types/optional.h"
namespace cricket {
// Abstract interface for providing the audio data.
// TODO(deadbeef): Rename this to AudioSourceInterface, and rename
// webrtc::AudioSourceInterface to AudioTrackSourceInterface.
class AudioSource {
public:
class Sink {
public:
// Callback to receive data from the AudioSource.
virtual void OnData(
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
absl::optional<int64_t> absolute_capture_timestamp_ms) = 0;
// Called when the AudioSource is going away.
virtual void OnClose() = 0;
// Returns the number of channels encoded by the sink. This can be less than
// the number_of_channels if down-mixing occur. A value of -1 means an
// unknown number.
virtual int NumPreferredChannels() const = 0;
protected:
virtual ~Sink() {}
};
// Sets a sink to the AudioSource. There can be only one sink connected
// to the source at a time.
virtual void SetSink(Sink* sink) = 0;
protected:
virtual ~AudioSource() {}
};
} // namespace cricket
#endif // MEDIA_BASE_AUDIO_SOURCE_H_
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