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/*
* 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 <assert.h>
#include <string.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 <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) {
memset(data_.get(), 0, sizeof(T) * samples_per_channel * num_channels);
for (int i = 0; i < num_channels; ++i)
channels_[i] = &data_[i * samples_per_channel];
}
~ChannelBuffer() {}
void CopyFrom(const void* channel_ptr, int i) {
assert(i < num_channels_);
memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T));
}
T* data() { return data_.get(); }
T* channel(int i) {
assert(i < num_channels_);
return channels_[i];
}
T** channels() { return channels_.get(); }
int samples_per_channel() { return samples_per_channel_; }
int num_channels() { return num_channels_; }
int length() { return samples_per_channel_ * num_channels_; }
private:
scoped_ptr<T[]> data_;
scoped_ptr<T*[]> channels_;
int samples_per_channel_;
int num_channels_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
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