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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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