webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

51 lines
1.7 KiB
C++

/*
* 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 "common_audio/resampler/include/resampler.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "rtc_base/constructormagic.h"
#include "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 // MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_