ed083d4079
This saves the user from having to explicitly check stream_has_voice(). It will allow typing detection to function, which relies on this behaviour. Review URL: http://webrtc-codereview.appspot.com/144004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@621 4adac7df-926f-26a2-2b94-8c16560cd09d
72 lines
2.3 KiB
C++
72 lines
2.3 KiB
C++
/*
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* Copyright (c) 2011 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_MAIN_SOURCE_AUDIO_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
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#include "module_common_types.h"
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#include "typedefs.h"
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namespace webrtc {
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struct AudioChannel;
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struct SplitAudioChannel;
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class AudioBuffer {
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public:
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AudioBuffer(int max_num_channels, int samples_per_channel);
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virtual ~AudioBuffer();
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int num_channels() const;
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int samples_per_channel() const;
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int samples_per_split_channel() const;
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WebRtc_Word16* data(int channel) const;
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WebRtc_Word16* low_pass_split_data(int channel) const;
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WebRtc_Word16* high_pass_split_data(int channel) const;
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WebRtc_Word16* mixed_low_pass_data(int channel) const;
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WebRtc_Word16* low_pass_reference(int channel) const;
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WebRtc_Word32* analysis_filter_state1(int channel) const;
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WebRtc_Word32* analysis_filter_state2(int channel) const;
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WebRtc_Word32* synthesis_filter_state1(int channel) const;
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WebRtc_Word32* synthesis_filter_state2(int channel) const;
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void set_activity(AudioFrame::VADActivity activity);
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AudioFrame::VADActivity activity();
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void DeinterleaveFrom(AudioFrame* audioFrame);
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void InterleaveTo(AudioFrame* audioFrame) const;
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void Mix(int num_mixed_channels);
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void CopyAndMixLowPass(int num_mixed_channels);
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void CopyLowPassToReference();
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private:
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const int max_num_channels_;
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int num_channels_;
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int num_mixed_channels_;
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int num_mixed_low_pass_channels_;
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const int samples_per_channel_;
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int samples_per_split_channel_;
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bool reference_copied_;
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AudioFrame::VADActivity activity_;
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WebRtc_Word16* data_;
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// TODO(andrew): use vectors here.
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AudioChannel* channels_;
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SplitAudioChannel* split_channels_;
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// TODO(andrew): improve this, we don't need the full 32 kHz space here.
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AudioChannel* mixed_low_pass_channels_;
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AudioChannel* low_pass_reference_channels_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
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