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/*
* Copyright (c) 2013 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 "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
namespace webrtc {
namespace test {
InputAudioFile::InputAudioFile(const std::string file_name) {
fp_ = fopen(file_name.c_str(), "rb");
}
InputAudioFile::~InputAudioFile() { fclose(fp_); }
bool InputAudioFile::Read(size_t samples, int16_t* destination) {
if (!fp_) {
return false;
}
size_t samples_read = fread(destination, sizeof(int16_t), samples, fp_);
if (samples_read < samples) {
// Rewind and read the missing samples.
rewind(fp_);
size_t missing_samples = samples - samples_read;
if (fread(destination, sizeof(int16_t), missing_samples, fp_) <
missing_samples) {
// Could not read enough even after rewinding the file.
return false;
}
}
return true;
}
void InputAudioFile::DuplicateInterleaved(const int16_t* source, size_t samples,
size_t channels,
int16_t* destination) {
// Start from the end of |source| and |destination|, and work towards the
// beginning. This is to allow in-place interleaving of the same array (i.e.,
// |source| and |destination| are the same array).
for (int i = static_cast<int>(samples - 1); i >= 0; --i) {
for (int j = static_cast<int>(channels - 1); j >= 0; --j) {
destination[i * channels + j] = source[i];
}
}
}
} // namespace test
} // namespace webrtc
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