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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.
*/
#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include <memory>
#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
bool ResampleInputAudioFile::Read(size_t samples,
int output_rate_hz,
int16_t* destination) {
const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
<< "Frame size and sample rates don't add up to an integer.";
std::unique_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
return false;
resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
size_t output_length = 0;
RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read,
destination, samples, output_length),
0);
RTC_CHECK_EQ(samples, output_length);
return true;
}
bool ResampleInputAudioFile::Read(size_t samples, int16_t* destination) {
RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
return Read(samples, output_rate_hz_, destination);
}
void ResampleInputAudioFile::set_output_rate_hz(int rate_hz) {
output_rate_hz_ = rate_hz;
}
} // namespace test
} // namespace webrtc
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