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/*
* 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 MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
#include <string>
#include "absl/strings/string_view.h"
#include "common_audio/resampler/include/resampler.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
namespace webrtc {
namespace test {
// Class for handling a looping input audio file with resampling.
class ResampleInputAudioFile : public InputAudioFile {
public:
ResampleInputAudioFile(absl::string_view file_name,
int file_rate_hz,
bool loop_at_end = true)
: InputAudioFile(file_name, loop_at_end),
file_rate_hz_(file_rate_hz),
output_rate_hz_(-1) {}
ResampleInputAudioFile(absl::string_view file_name,
int file_rate_hz,
int output_rate_hz,
bool loop_at_end = true)
: InputAudioFile(file_name, loop_at_end),
file_rate_hz_(file_rate_hz),
output_rate_hz_(output_rate_hz) {}
ResampleInputAudioFile(const ResampleInputAudioFile&) = delete;
ResampleInputAudioFile& operator=(const ResampleInputAudioFile&) = delete;
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_;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_