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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
59 lines
2 KiB
C++
59 lines
2 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 "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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next_index_ = 0;
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return true;
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}
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rtc::ArrayView<const int16_t> AudioLoop::GetNextBlock() {
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// Check that the AudioLoop is initialized.
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if (block_length_samples_ == 0)
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return rtc::ArrayView<const int16_t>();
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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 rtc::ArrayView<const int16_t>(output_ptr, block_length_samples_);
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}
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} // namespace test
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} // namespace webrtc
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