/* * 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 #include #include "webrtc/base/checks.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 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) { Initialize(); } ChannelBuffer(const T* data, 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) { Initialize(); memcpy(data_.get(), data, length() * sizeof(T)); } ChannelBuffer(const T* const* channels, 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) { Initialize(); for (int i = 0; i < num_channels_; ++i) CopyFrom(channels[i], i); } ~ChannelBuffer() {} void CopyFrom(const void* channel_ptr, int i) { DCHECK_LT(i, num_channels_); memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T)); } T* data() { return data_.get(); } const T* data() const { return data_.get(); } const T* channel(int i) const { DCHECK_GE(i, 0); DCHECK_LT(i, num_channels_); return channels_[i]; } T* channel(int i) { const ChannelBuffer* t = this; return const_cast(t->channel(i)); } T* const* channels() { return channels_.get(); } const T* const* channels() const { return channels_.get(); } int samples_per_channel() const { return samples_per_channel_; } int num_channels() const { return num_channels_; } int length() const { return samples_per_channel_ * num_channels_; } private: void Initialize() { memset(data_.get(), 0, sizeof(T) * length()); for (int i = 0; i < num_channels_; ++i) channels_[i] = &data_[i * samples_per_channel_]; } scoped_ptr data_; scoped_ptr channels_; const int samples_per_channel_; const int num_channels_; }; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_