AnotherFoxGuy/webrtc-audio-processing
1
0
Fork 0
Code Pull Requests Activity

Update common_audio

Browse Source 
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Moved src/ to webrtc/ to easily diff against the third_party/webrtc
   in the chromium tree

 * ARM/NEON/MIPS support is not yet hooked up

 * Tests have not been copied
This commit is contained in:
Arun Raghavan
2015-10-13 12:16:16 +05:30
parent 9413986e79
commit c4fb4e38de
230 changed files with 11201 additions and 8656 deletions
Download Patch File Download Diff File Expand all files Collapse all files

71
webrtc/modules/audio_processing/audio_buffer.h Normal file
View File

@ -0,0 +1,71 @@
/*
* 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 "module_common_types.h"
#include "typedefs.h"
namespace webrtc {
struct AudioChannel;
struct SplitAudioChannel;
class AudioBuffer {
public:
AudioBuffer(int max_num_channels, int samples_per_channel);
virtual ~AudioBuffer();
int num_channels() const;
int samples_per_channel() const;
int samples_per_split_channel() const;
WebRtc_Word16* data(int channel) const;
WebRtc_Word16* low_pass_split_data(int channel) const;
WebRtc_Word16* high_pass_split_data(int channel) const;
WebRtc_Word16* mixed_low_pass_data(int channel) const;
WebRtc_Word16* low_pass_reference(int channel) const;
WebRtc_Word32* analysis_filter_state1(int channel) const;
WebRtc_Word32* analysis_filter_state2(int channel) const;
WebRtc_Word32* synthesis_filter_state1(int channel) const;
WebRtc_Word32* synthesis_filter_state2(int channel) const;
void set_activity(AudioFrame::VADActivity activity);
AudioFrame::VADActivity activity();
void DeinterleaveFrom(AudioFrame* audioFrame);
void InterleaveTo(AudioFrame* audioFrame) const;
void Mix(int num_mixed_channels);
void CopyAndMixLowPass(int num_mixed_channels);
void CopyLowPassToReference();
private:
const int max_num_channels_;
int num_channels_;
int num_mixed_channels_;
int num_mixed_low_pass_channels_;
const int samples_per_channel_;
int samples_per_split_channel_;
bool reference_copied_;
AudioFrame::VADActivity activity_;
WebRtc_Word16* data_;
// TODO(andrew): use vectors here.
AudioChannel* channels_;
SplitAudioChannel* split_channels_;
// TODO(andrew): 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_