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libtgvoip/webrtc_dsp/modules/audio_processing/include/audio_generator.h

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/*
* Copyright (c) 2018 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 MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_H_
#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_H_
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
// This class is used as input sink for the APM, for diagnostic purposes.
// Generates an infinite audio signal, [-1, 1] floating point values, in frames
// of fixed channel count and sample rate.
class AudioGenerator {
public:
virtual ~AudioGenerator() {}
// Fill |audio| with the next samples of the audio signal.
virtual void FillFrame(AudioFrameView<float> audio) = 0;
// Return the number of channels output by the AudioGenerator.
virtual size_t NumChannels() = 0;
// Return the sample rate output by the AudioGenerator.
virtual size_t SampleRateHz() = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_H_