/* * 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 #include #include 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