| /* |
| * 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_CHANNEL_BUFFER_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_ |
| |
| #include <string.h> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/common_audio/include/audio_util.h" |
| |
| namespace webrtc { |
| |
| // Helper to encapsulate a contiguous data buffer with access to a pointer |
| // array of the deinterleaved channels. The buffer is zero initialized at |
| // creation. |
| 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) { |
| 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>* t = this; |
| return const_cast<T*>(t->channel(i)); |
| } |
| |
| T* const* channels() { return channels_.get(); } |
| const T* const* channels() const { return channels_.get(); } |
| |
| // Sets the |slice| pointers to the |start_frame| position for each channel. |
| // Returns |slice| for convenience. |
| const T* const* Slice(T** slice, int start_frame) const { |
| DCHECK_LT(start_frame, samples_per_channel_); |
| for (int i = 0; i < num_channels_; ++i) |
| slice[i] = &channels_[i][start_frame]; |
| return slice; |
| } |
| T** Slice(T** slice, int start_frame) { |
| const ChannelBuffer<T>* t = this; |
| return const_cast<T**>(t->Slice(slice, start_frame)); |
| } |
| |
| 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<T[]> data_; |
| scoped_ptr<T*[]> channels_; |
| const int samples_per_channel_; |
| const int num_channels_; |
| }; |
| |
| // One int16_t and one float ChannelBuffer that are kept in sync. The sync is |
| // broken when someone requests write access to either ChannelBuffer, and |
| // reestablished when someone requests the outdated ChannelBuffer. It is |
| // therefore safe to use the return value of ibuf_const() and fbuf_const() |
| // until the next call to ibuf() or fbuf(), and the return value of ibuf() and |
| // fbuf() until the next call to any of the other functions. |
| class IFChannelBuffer { |
| public: |
| IFChannelBuffer(int samples_per_channel, int num_channels); |
| |
| ChannelBuffer<int16_t>* ibuf(); |
| ChannelBuffer<float>* fbuf(); |
| const ChannelBuffer<int16_t>* ibuf_const() const; |
| const ChannelBuffer<float>* fbuf_const() const; |
| |
| int num_channels() const { return ibuf_.num_channels(); } |
| int samples_per_channel() const { return ibuf_.samples_per_channel(); } |
| |
| private: |
| void RefreshF() const; |
| void RefreshI() const; |
| |
| mutable bool ivalid_; |
| mutable ChannelBuffer<int16_t> ibuf_; |
| mutable bool fvalid_; |
| mutable ChannelBuffer<float> fbuf_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_ |