blob: 17c48144a84410fdbaba796ca78e61c5dfcf24be [file] [log] [blame]
/*
* 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"
#include "webrtc/base/checks.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 + samples_read, sizeof(int16_t), missing_samples,
fp_) < missing_samples) {
// Could not read enough even after rewinding the file.
return false;
}
}
return true;
}
bool InputAudioFile::Seek(int samples) {
if (!fp_) {
return false;
}
// Find file boundaries.
const long current_pos = ftell(fp_);
RTC_CHECK_NE(EOF, current_pos)
<< "Error returned when getting file position.";
RTC_CHECK_EQ(0, fseek(fp_, 0, SEEK_END)); // Move to end of file.
const long file_size = ftell(fp_);
RTC_CHECK_NE(EOF, file_size) << "Error returned when getting file position.";
// Find new position.
long new_pos = current_pos + sizeof(int16_t) * samples; // Samples to bytes.
RTC_CHECK_GE(new_pos, 0)
<< "Trying to move to before the beginning of the file";
new_pos = new_pos % file_size; // Wrap around the end of the file.
// Move to new position relative to the beginning of the file.
RTC_CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET));
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