mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
89bbf379ce
Instead of setting a runtime, allow neteq_quality_test to consume a complete file using --runtime_ms -1 Bug: webrtc:10690 Change-Id: I90d35cf31996d9336fef817b9332a2cd1d04e77e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139101 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Pablo Barrera González <barrerap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28134}
53 lines
1.9 KiB
C++
53 lines
1.9 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/constructor_magic.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,
|
|
bool loop_at_end = true)
|
|
: InputAudioFile(file_name, loop_at_end),
|
|
file_rate_hz_(file_rate_hz),
|
|
output_rate_hz_(-1) {}
|
|
ResampleInputAudioFile(const std::string 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) {}
|
|
|
|
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_
|