blob: 564eda1f03b7b2e25deb240b7334ebedd62f7d2b [file] [log] [blame]
/*
* Copyright (c) 2014 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_COMMON_AUDIO_AUDIO_CONVERTER_H_
#define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
// TODO(ajm): Move channel buffer to common_audio.
#include "webrtc/base/constructormagic.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/scoped_vector.h"
namespace webrtc {
class PushSincResampler;
// Format conversion (remixing and resampling) for audio. Only simple remixing
// conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
// upmix from mono (i.e. |src_channels == 1|).
//
// The source and destination chunks have the same duration in time; specifying
// the number of frames is equivalent to specifying the sample rates.
class AudioConverter {
public:
AudioConverter(int src_channels, int src_frames,
int dst_channels, int dst_frames);
void Convert(const float* const* src,
int src_channels,
int src_frames,
int dst_channels,
int dst_frames,
float* const* dest);
private:
const int src_channels_;
const int src_frames_;
const int dst_channels_;
const int dst_frames_;
scoped_ptr<ChannelBuffer<float>> downmix_buffer_;
ScopedVector<PushSincResampler> resamplers_;
DISALLOW_COPY_AND_ASSIGN(AudioConverter);
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_