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| 1 /* | 1 /* | 
| 2  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | 2  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | 
| 3  * | 3  * | 
| 4  *  Use of this source code is governed by a BSD-style license | 4  *  Use of this source code is governed by a BSD-style license | 
| 5  *  that can be found in the LICENSE file in the root of the source | 5  *  that can be found in the LICENSE file in the root of the source | 
| 6  *  tree. An additional intellectual property rights grant can be found | 6  *  tree. An additional intellectual property rights grant can be found | 
| 7  *  in the file PATENTS.  All contributing project authors may | 7  *  in the file PATENTS.  All contributing project authors may | 
| 8  *  be found in the AUTHORS file in the root of the source tree. | 8  *  be found in the AUTHORS file in the root of the source tree. | 
| 9  */ | 9  */ | 
| 10 | 10 | 
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| 116   void CopyFrom(const float* const* data, const StreamConfig& stream_config); | 116   void CopyFrom(const float* const* data, const StreamConfig& stream_config); | 
| 117   void CopyTo(const StreamConfig& stream_config, float* const* data); | 117   void CopyTo(const StreamConfig& stream_config, float* const* data); | 
| 118   void CopyLowPassToReference(); | 118   void CopyLowPassToReference(); | 
| 119 | 119 | 
| 120   // Splits the signal into different bands. | 120   // Splits the signal into different bands. | 
| 121   void SplitIntoFrequencyBands(); | 121   void SplitIntoFrequencyBands(); | 
| 122   // Recombine the different bands into one signal. | 122   // Recombine the different bands into one signal. | 
| 123   void MergeFrequencyBands(); | 123   void MergeFrequencyBands(); | 
| 124 | 124 | 
| 125  private: | 125  private: | 
|  | 126   FRIEND_TEST_ALL_PREFIXES(AudioBufferTest, | 
|  | 127                            SetNumChannelsSetsChannelBuffersNumChannels); | 
| 126   // Called from DeinterleaveFrom() and CopyFrom(). | 128   // Called from DeinterleaveFrom() and CopyFrom(). | 
| 127   void InitForNewData(); | 129   void InitForNewData(); | 
| 128 | 130 | 
| 129   // The audio is passed into DeinterleaveFrom() or CopyFrom() with input | 131   // The audio is passed into DeinterleaveFrom() or CopyFrom() with input | 
| 130   // format (samples per channel and number of channels). | 132   // format (samples per channel and number of channels). | 
| 131   const size_t input_num_frames_; | 133   const size_t input_num_frames_; | 
| 132   const size_t num_input_channels_; | 134   const size_t num_input_channels_; | 
| 133   // The audio is stored by DeinterleaveFrom() or CopyFrom() with processing | 135   // The audio is stored by DeinterleaveFrom() or CopyFrom() with processing | 
| 134   // format. | 136   // format. | 
| 135   const size_t proc_num_frames_; | 137   const size_t proc_num_frames_; | 
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| 155   std::unique_ptr<IFChannelBuffer> input_buffer_; | 157   std::unique_ptr<IFChannelBuffer> input_buffer_; | 
| 156   std::unique_ptr<IFChannelBuffer> output_buffer_; | 158   std::unique_ptr<IFChannelBuffer> output_buffer_; | 
| 157   std::unique_ptr<ChannelBuffer<float> > process_buffer_; | 159   std::unique_ptr<ChannelBuffer<float> > process_buffer_; | 
| 158   std::vector<std::unique_ptr<PushSincResampler>> input_resamplers_; | 160   std::vector<std::unique_ptr<PushSincResampler>> input_resamplers_; | 
| 159   std::vector<std::unique_ptr<PushSincResampler>> output_resamplers_; | 161   std::vector<std::unique_ptr<PushSincResampler>> output_resamplers_; | 
| 160 }; | 162 }; | 
| 161 | 163 | 
| 162 }  // namespace webrtc | 164 }  // namespace webrtc | 
| 163 | 165 | 
| 164 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ | 166 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ | 
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