
Removed the zero out memset in this change: https://review.webrtc.org/24469004/ assuming it was unneeded. Dr. Memory taught me that assupmtion was invalid. linux_memcheck try runs might have caught this, if they weren't flaking out on unrelated stuff. TBR=claguna@google.com Review URL: https://webrtc-codereview.appspot.com/28429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7113 4adac7df-926f-26a2-2b94-8c16560cd09d
112 lines
3.3 KiB
C++
112 lines
3.3 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/base/checks.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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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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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
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