ddbb8a2c24
Select "processing" rates based on the input and output sampling rates. Resample the input streams to those rates, and if necessary to the output rate. - Remove deprecated stream format APIs. - Remove deprecated device sample rate APIs. - Add a ChannelBuffer class to help manage deinterleaved channels. - Clean up the splitting filter state. - Add a unit test which verifies the output against known-working native format output. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
76 lines
2.3 KiB
C++
76 lines
2.3 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 <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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T* channel(int i) {
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assert(i < num_channels_);
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return channels_[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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