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https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
58 lines
1.9 KiB
C++
58 lines
1.9 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
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#include <assert.h>
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#include <stdio.h>
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#include <string.h>
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namespace webrtc {
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namespace test {
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bool AudioLoop::Init(const std::string file_name,
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size_t max_loop_length_samples,
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size_t block_length_samples) {
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FILE* fp = fopen(file_name.c_str(), "rb");
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if (!fp) return false;
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audio_array_.reset(new int16_t[max_loop_length_samples +
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block_length_samples]);
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size_t samples_read = fread(audio_array_.get(), sizeof(int16_t),
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max_loop_length_samples, fp);
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fclose(fp);
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// Block length must be shorter than the loop length.
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if (block_length_samples > samples_read) return false;
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// Add an extra block length of samples to the end of the array, starting
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// over again from the beginning of the array. This is done to simplify
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// the reading process when reading over the end of the loop.
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memcpy(&audio_array_[samples_read], audio_array_.get(),
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block_length_samples * sizeof(int16_t));
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loop_length_samples_ = samples_read;
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block_length_samples_ = block_length_samples;
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return true;
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}
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const int16_t* AudioLoop::GetNextBlock() {
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// Check that the AudioLoop is initialized.
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if (block_length_samples_ == 0) return NULL;
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const int16_t* output_ptr = &audio_array_[next_index_];
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next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
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return output_ptr;
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}
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} // namespace test
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} // namespace webrtc
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