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git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d

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niklase@google.com
2011-05-30 11:22:19 +00:00
parent 01813fe945
commit 77ae29bc81
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modules/audio_processing/main/source/audio_buffer.h Normal file
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/*
* Copyright (c) 2011 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_MAIN_SOURCE_AUDIO_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
#include "typedefs.h"
namespace webrtc {
struct AudioChannel;
struct SplitAudioChannel;
class AudioFrame;
class AudioBuffer {
public:
AudioBuffer(WebRtc_Word32 max_num_channels, WebRtc_Word32 samples_per_channel);
virtual ~AudioBuffer();
WebRtc_Word32 num_channels() const;
WebRtc_Word32 samples_per_channel() const;
WebRtc_Word32 samples_per_split_channel() const;
WebRtc_Word16* data(WebRtc_Word32 channel) const;
WebRtc_Word16* low_pass_split_data(WebRtc_Word32 channel) const;
WebRtc_Word16* high_pass_split_data(WebRtc_Word32 channel) const;
WebRtc_Word16* mixed_low_pass_data(WebRtc_Word32 channel) const;
WebRtc_Word16* low_pass_reference(WebRtc_Word32 channel) const;
WebRtc_Word32* analysis_filter_state1(WebRtc_Word32 channel) const;
WebRtc_Word32* analysis_filter_state2(WebRtc_Word32 channel) const;
WebRtc_Word32* synthesis_filter_state1(WebRtc_Word32 channel) const;
WebRtc_Word32* synthesis_filter_state2(WebRtc_Word32 channel) const;
void DeinterleaveFrom(AudioFrame* audioFrame);
void InterleaveTo(AudioFrame* audioFrame) const;
void Mix(WebRtc_Word32 num_mixed_channels);
void CopyAndMixLowPass(WebRtc_Word32 num_mixed_channels);
void CopyLowPassToReference();
private:
const WebRtc_Word32 max_num_channels_;
WebRtc_Word32 num_channels_;
WebRtc_Word32 num_mixed_channels_;
WebRtc_Word32 num_mixed_low_pass_channels_;
const WebRtc_Word32 samples_per_channel_;
WebRtc_Word32 samples_per_split_channel_;
bool reference_copied_;
WebRtc_Word16* data_;
// TODO(ajm): Prefer to make these vectors if permitted...
AudioChannel* channels_;
SplitAudioChannel* split_channels_;
// TODO(ajm): improve this, we don't need the full 32 kHz space here.
AudioChannel* mixed_low_pass_channels_;
AudioChannel* low_pass_reference_channels_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_