blob: c0af3546b0478762a997f4cf063768e8c4daeee2 [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_MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
#include <string>
#include "webrtc/base/constructormagic.h"
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/typedefs.h"
namespace webrtc {
namespace test {
// Class for handling a looping input audio file with resampling.
class ResampleInputAudioFile : public InputAudioFile {
public:
ResampleInputAudioFile(const std::string file_name, int file_rate_hz)
: InputAudioFile(file_name),
file_rate_hz_(file_rate_hz),
output_rate_hz_(-1) {}
ResampleInputAudioFile(const std::string file_name,
int file_rate_hz,
int output_rate_hz)
: InputAudioFile(file_name),
file_rate_hz_(file_rate_hz),
output_rate_hz_(output_rate_hz) {}
bool Read(size_t samples, int output_rate_hz, int16_t* destination);
bool Read(size_t samples, int16_t* destination) override;
void set_output_rate_hz(int rate_hz);
private:
const int file_rate_hz_;
int output_rate_hz_;
Resampler resampler_;
RTC_DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_