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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 // AudioRendererAlgorithm buffers and transforms audio data. The owner of | 5 // AudioRendererAlgorithm buffers and transforms audio data. The owner of |
6 // this object provides audio data to the object through EnqueueBuffer() and | 6 // this object provides audio data to the object through EnqueueBuffer() and |
7 // requests data from the buffer via FillBuffer(). The owner also sets the | 7 // requests data from the buffer via FillBuffer(). The owner also sets the |
8 // playback rate, and the AudioRendererAlgorithm will stretch or compress the | 8 // playback rate, and the AudioRendererAlgorithm will stretch or compress the |
9 // buffered audio as necessary to match the playback rate when fulfilling | 9 // buffered audio as necessary to match the playback rate when fulfilling |
10 // FillBuffer() requests. | 10 // FillBuffer() requests. |
11 // | 11 // |
12 // This class is *not* thread-safe. Calls to enqueue and retrieve data must be | 12 // This class is *not* thread-safe. Calls to enqueue and retrieve data must be |
13 // locked if called from multiple threads. | 13 // locked if called from multiple threads. |
14 // | 14 // |
15 // AudioRendererAlgorithm uses the Waveform Similarity Overlap and Add (WSOLA) | 15 // AudioRendererAlgorithm uses the Waveform Similarity Overlap and Add (WSOLA) |
16 // algorithm to stretch or compress audio data to meet playback speeds less than | 16 // algorithm to stretch or compress audio data to meet playback speeds less than |
17 // or greater than the natural playback of the audio stream. The algorithm | 17 // or greater than the natural playback of the audio stream. The algorithm |
18 // preserves local properties of the audio, therefore, pitch and harmonics are | 18 // preserves local properties of the audio, therefore, pitch and harmonics are |
19 // are preserved. See audio_renderer_algorith.cc for a more elaborate | 19 // are preserved. See audio_renderer_algorith.cc for a more elaborate |
20 // description of the algorithm. | 20 // description of the algorithm. |
21 // | 21 // |
22 // Audio at very low or very high playback rates are muted to preserve quality. | |
23 // | |
24 | 22 |
25 #ifndef MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_ | 23 #ifndef MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_ |
26 #define MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_ | 24 #define MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_ |
27 | 25 |
28 #include "base/memory/ref_counted.h" | 26 #include "base/memory/ref_counted.h" |
29 #include "base/memory/scoped_ptr.h" | 27 #include "base/memory/scoped_ptr.h" |
30 #include "media/audio/audio_parameters.h" | 28 #include "media/audio/audio_parameters.h" |
31 #include "media/base/audio_buffer.h" | 29 #include "media/base/audio_buffer.h" |
32 #include "media/base/audio_buffer_queue.h" | 30 #include "media/base/audio_buffer_queue.h" |
33 | 31 |
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77 void IncreaseQueueCapacity(); | 75 void IncreaseQueueCapacity(); |
78 | 76 |
79 // Returns the number of frames left in |audio_buffer_|, which may be larger | 77 // Returns the number of frames left in |audio_buffer_|, which may be larger |
80 // than QueueCapacity() in the event that EnqueueBuffer() delivered more data | 78 // than QueueCapacity() in the event that EnqueueBuffer() delivered more data |
81 // than |audio_buffer_| was intending to hold. | 79 // than |audio_buffer_| was intending to hold. |
82 int frames_buffered() { return audio_buffer_.frames(); } | 80 int frames_buffered() { return audio_buffer_.frames(); } |
83 | 81 |
84 // Returns the samples per second for this audio stream. | 82 // Returns the samples per second for this audio stream. |
85 int samples_per_second() { return samples_per_second_; } | 83 int samples_per_second() { return samples_per_second_; } |
86 | 84 |
87 // Is the sound currently muted? | |
88 bool is_muted() { return muted_; } | |
89 | |
90 private: | 85 private: |
91 // Within |search_block_|, find the block of data that is most similar to | 86 // Within |search_block_|, find the block of data that is most similar to |
92 // |target_block_|, and write it in |optimal_block_|. This method assumes that | 87 // |target_block_|, and write it in |optimal_block_|. This method assumes that |
93 // there is enough data to perform a search, i.e. |search_block_| and | 88 // there is enough data to perform a search, i.e. |search_block_| and |
94 // |target_block_| can be extracted from the available frames. | 89 // |target_block_| can be extracted from the available frames. |
95 void GetOptimalBlock(); | 90 void GetOptimalBlock(); |
96 | 91 |
97 // Read a maximum of |requested_frames| frames from |wsola_output_|. Returns | 92 // Read a maximum of |requested_frames| frames from |wsola_output_|. Returns |
98 // number of frames actually read. | 93 // number of frames actually read. |
99 int WriteCompletedFramesTo( | 94 int WriteCompletedFramesTo( |
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133 | 128 |
134 // Sample rate of audio stream. | 129 // Sample rate of audio stream. |
135 int samples_per_second_; | 130 int samples_per_second_; |
136 | 131 |
137 // Used by algorithm to scale output. | 132 // Used by algorithm to scale output. |
138 float playback_rate_; | 133 float playback_rate_; |
139 | 134 |
140 // Buffered audio data. | 135 // Buffered audio data. |
141 AudioBufferQueue audio_buffer_; | 136 AudioBufferQueue audio_buffer_; |
142 | 137 |
143 // True if the audio should be muted. | |
144 bool muted_; | |
145 | |
146 // If muted, keep track of partial frames that should have been skipped over. | |
147 double muted_partial_frame_; | |
148 | |
149 // How many frames to have in the queue before we report the queue is full. | 138 // How many frames to have in the queue before we report the queue is full. |
150 int capacity_; | 139 int capacity_; |
151 | 140 |
152 // Book keeping of the current time of generated audio, in frames. This | 141 // Book keeping of the current time of generated audio, in frames. This |
153 // should be appropriately updated when out samples are generated, regardless | 142 // should be appropriately updated when out samples are generated, regardless |
154 // of whether we push samples out when FillBuffer() is called or we store | 143 // of whether we push samples out when FillBuffer() is called or we store |
155 // audio in |wsola_output_| for the subsequent calls to FillBuffer(). | 144 // audio in |wsola_output_| for the subsequent calls to FillBuffer(). |
156 // Furthermore, if samples from |audio_buffer_| are evicted then this | 145 // Furthermore, if samples from |audio_buffer_| are evicted then this |
157 // member variable should be updated based on |playback_rate_|. | 146 // member variable should be updated based on |playback_rate_|. |
158 // Note that this member should be updated ONLY by calling UpdateOutputTime(), | 147 // Note that this member should be updated ONLY by calling UpdateOutputTime(), |
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212 // searched for a block (|optimal_block_|) that is most similar to | 201 // searched for a block (|optimal_block_|) that is most similar to |
213 // |target_block_|. | 202 // |target_block_|. |
214 scoped_ptr<AudioBus> target_block_; | 203 scoped_ptr<AudioBus> target_block_; |
215 | 204 |
216 DISALLOW_COPY_AND_ASSIGN(AudioRendererAlgorithm); | 205 DISALLOW_COPY_AND_ASSIGN(AudioRendererAlgorithm); |
217 }; | 206 }; |
218 | 207 |
219 } // namespace media | 208 } // namespace media |
220 | 209 |
221 #endif // MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_ | 210 #endif // MEDIA_FILTERS_AUDIO_RENDERER_ALGORITHM_H_ |
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