Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

58 lines
1.9 KiB
C++

/*
* 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/audio_loop.h"
#include <assert.h>
#include <stdio.h>
#include <string.h>
namespace webrtc {
namespace test {
bool AudioLoop::Init(const std::string file_name,
size_t max_loop_length_samples,
size_t block_length_samples) {
FILE* fp = fopen(file_name.c_str(), "rb");
if (!fp) return false;
audio_array_.reset(new int16_t[max_loop_length_samples +
block_length_samples]);
size_t samples_read = fread(audio_array_.get(), sizeof(int16_t),
max_loop_length_samples, fp);
fclose(fp);
// Block length must be shorter than the loop length.
if (block_length_samples > samples_read) return false;
// Add an extra block length of samples to the end of the array, starting
// over again from the beginning of the array. This is done to simplify
// the reading process when reading over the end of the loop.
memcpy(&audio_array_[samples_read], audio_array_.get(),
block_length_samples * sizeof(int16_t));
loop_length_samples_ = samples_read;
block_length_samples_ = block_length_samples;
return true;
}
const int16_t* AudioLoop::GetNextBlock() {
// Check that the AudioLoop is initialized.
if (block_length_samples_ == 0) return NULL;
const int16_t* output_ptr = &audio_array_[next_index_];
next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
return output_ptr;
}
} // namespace test
} // namespace webrtc