diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/audio_loop.cc')
-rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/audio_loop.cc | 57 |
1 files changed, 0 insertions, 57 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/audio_loop.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/audio_loop.cc deleted file mode 100644 index 94ea5bef015..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/tools/audio_loop.cc +++ /dev/null @@ -1,57 +0,0 @@ -/* - * 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/neteq4/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 |