
No change in functionallity. BUG=webrtc:3146 R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7760 4adac7df-926f-26a2-2b94-8c16560cd09d
125 lines
3.9 KiB
C++
125 lines
3.9 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
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#include <string.h>
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#include "webrtc/base/checks.h"
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#include "webrtc/common_audio/include/audio_util.h"
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namespace webrtc {
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// Helper to encapsulate a contiguous data buffer with access to a pointer
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// array of the deinterleaved channels.
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template <typename T>
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class ChannelBuffer {
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public:
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ChannelBuffer(int samples_per_channel, int num_channels)
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: data_(new T[samples_per_channel * num_channels]),
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channels_(new T*[num_channels]),
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samples_per_channel_(samples_per_channel),
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num_channels_(num_channels) {
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Initialize();
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}
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ChannelBuffer(const T* data, int samples_per_channel, int num_channels)
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: data_(new T[samples_per_channel * num_channels]),
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channels_(new T*[num_channels]),
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samples_per_channel_(samples_per_channel),
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num_channels_(num_channels) {
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Initialize();
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memcpy(data_.get(), data, length() * sizeof(T));
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}
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ChannelBuffer(const T* const* channels, int samples_per_channel,
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int num_channels)
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: data_(new T[samples_per_channel * num_channels]),
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channels_(new T*[num_channels]),
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samples_per_channel_(samples_per_channel),
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num_channels_(num_channels) {
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Initialize();
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for (int i = 0; i < num_channels_; ++i)
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CopyFrom(channels[i], i);
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}
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~ChannelBuffer() {}
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void CopyFrom(const void* channel_ptr, int i) {
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DCHECK_LT(i, num_channels_);
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memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T));
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}
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T* data() { return data_.get(); }
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const T* data() const { return data_.get(); }
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const T* channel(int i) const {
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DCHECK_GE(i, 0);
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DCHECK_LT(i, num_channels_);
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return channels_[i];
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}
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T* channel(int i) {
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const ChannelBuffer<T>* t = this;
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return const_cast<T*>(t->channel(i));
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}
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T* const* channels() { return channels_.get(); }
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const T* const* channels() const { return channels_.get(); }
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int samples_per_channel() const { return samples_per_channel_; }
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int num_channels() const { return num_channels_; }
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int length() const { return samples_per_channel_ * num_channels_; }
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private:
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void Initialize() {
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memset(data_.get(), 0, sizeof(T) * length());
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for (int i = 0; i < num_channels_; ++i)
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channels_[i] = &data_[i * samples_per_channel_];
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}
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scoped_ptr<T[]> data_;
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scoped_ptr<T*[]> channels_;
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const int samples_per_channel_;
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const int num_channels_;
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};
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// One int16_t and one float ChannelBuffer that are kept in sync. The sync is
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// broken when someone requests write access to either ChannelBuffer, and
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// reestablished when someone requests the outdated ChannelBuffer. It is
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// therefore safe to use the return value of ibuf_const() and fbuf_const()
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// until the next call to ibuf() or fbuf(), and the return value of ibuf() and
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// fbuf() until the next call to any of the other functions.
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class IFChannelBuffer {
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public:
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IFChannelBuffer(int samples_per_channel, int num_channels);
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ChannelBuffer<int16_t>* ibuf();
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ChannelBuffer<float>* fbuf();
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const ChannelBuffer<int16_t>* ibuf_const() const;
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const ChannelBuffer<float>* fbuf_const() const;
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int num_channels() const { return ibuf_.num_channels(); }
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int samples_per_channel() const { return ibuf_.samples_per_channel(); }
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private:
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void RefreshF() const;
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void RefreshI() const;
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mutable bool ivalid_;
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mutable ChannelBuffer<int16_t> ibuf_;
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mutable bool fvalid_;
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mutable ChannelBuffer<float> fbuf_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
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