| /* |
| * Copyright (c) 2014 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. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_ |
| |
| #include <assert.h> |
| #include <string.h> |
| |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| |
| namespace webrtc { |
| |
| static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) { |
| switch (layout) { |
| case AudioProcessing::kMono: |
| case AudioProcessing::kMonoAndKeyboard: |
| return 1; |
| case AudioProcessing::kStereo: |
| case AudioProcessing::kStereoAndKeyboard: |
| return 2; |
| } |
| assert(false); |
| return -1; |
| } |
| |
| // Helper to encapsulate a contiguous data buffer with access to a pointer |
| // array of the deinterleaved channels. |
| template <typename T> |
| class ChannelBuffer { |
| public: |
| ChannelBuffer(int samples_per_channel, int num_channels) |
| : data_(new T[samples_per_channel * num_channels]), |
| channels_(new T*[num_channels]), |
| samples_per_channel_(samples_per_channel), |
| num_channels_(num_channels) { |
| memset(data_.get(), 0, sizeof(T) * samples_per_channel * num_channels); |
| for (int i = 0; i < num_channels; ++i) |
| channels_[i] = &data_[i * samples_per_channel]; |
| } |
| ~ChannelBuffer() {} |
| |
| void CopyFrom(const void* channel_ptr, int i) { |
| assert(i < num_channels_); |
| memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T)); |
| } |
| |
| T* data() { return data_.get(); } |
| const T* channel(int i) const { |
| assert(i >= 0 && i < num_channels_); |
| return channels_[i]; |
| } |
| T* channel(int i) { |
| const ChannelBuffer<T>* t = this; |
| return const_cast<T*>(t->channel(i)); |
| } |
| T** channels() { return channels_.get(); } |
| |
| int samples_per_channel() { return samples_per_channel_; } |
| int num_channels() { return num_channels_; } |
| int length() { return samples_per_channel_ * num_channels_; } |
| |
| private: |
| scoped_ptr<T[]> data_; |
| scoped_ptr<T*[]> channels_; |
| int samples_per_channel_; |
| int num_channels_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_ |