e364ac902f
Specifically, when someone asks for a const pointer to the int16 version of the array, there's no need to invalidate the float version of that array, and vice versa. (But obviously, invalidation still has to happen when someone asks for a non-const pointer.) R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6725 4adac7df-926f-26a2-2b94-8c16560cd09d
81 lines
2.4 KiB
C++
81 lines
2.4 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_COMMON_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
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#include <assert.h>
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#include <string.h>
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
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switch (layout) {
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case AudioProcessing::kMono:
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case AudioProcessing::kMonoAndKeyboard:
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return 1;
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case AudioProcessing::kStereo:
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case AudioProcessing::kStereoAndKeyboard:
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return 2;
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}
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assert(false);
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return -1;
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}
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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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memset(data_.get(), 0, sizeof(T) * samples_per_channel * num_channels);
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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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~ChannelBuffer() {}
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void CopyFrom(const void* channel_ptr, int i) {
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assert(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* channel(int i) const {
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assert(i >= 0 && 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** channels() { return channels_.get(); }
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int samples_per_channel() { return samples_per_channel_; }
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int num_channels() { return num_channels_; }
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int length() { return samples_per_channel_ * num_channels_; }
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private:
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scoped_ptr<T[]> data_;
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scoped_ptr<T*[]> channels_;
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int samples_per_channel_;
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int num_channels_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
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