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Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc')
-rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc | 57 |
1 files changed, 57 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc new file mode 100644 index 00000000000..2d2a7e3dd4a --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc @@ -0,0 +1,57 @@ +/* + * 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 |