Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(121)

Side by Side Diff: media/renderers/audio_renderer_impl.h

Issue 2522673003: Don't run WSOLA over always muted channels. (Closed)
Patch Set: Fix cast tests. Created 4 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 // Audio rendering unit utilizing an AudioRendererSink to output data. 5 // Audio rendering unit utilizing an AudioRendererSink to output data.
6 // 6 //
7 // This class lives inside three threads during it's lifetime, namely: 7 // This class lives inside three threads during it's lifetime, namely:
8 // 1. Render thread 8 // 1. Render thread
9 // Where the object is created. 9 // Where the object is created.
10 // 2. Media thread (provided via constructor) 10 // 2. Media thread (provided via constructor)
(...skipping 216 matching lines...) Expand 10 before | Expand all | Expand 10 after
227 std::unique_ptr<base::TickClock> tick_clock_; 227 std::unique_ptr<base::TickClock> tick_clock_;
228 228
229 // Memory usage of |algorithm_| recorded during the last 229 // Memory usage of |algorithm_| recorded during the last
230 // HandleDecodedBuffer_Locked() call. 230 // HandleDecodedBuffer_Locked() call.
231 int64_t last_audio_memory_usage_; 231 int64_t last_audio_memory_usage_;
232 232
233 // Sample rate of the last decoded audio buffer. Allows for detection of 233 // Sample rate of the last decoded audio buffer. Allows for detection of
234 // sample rate changes due to implicit AAC configuration change. 234 // sample rate changes due to implicit AAC configuration change.
235 int last_decoded_sample_rate_; 235 int last_decoded_sample_rate_;
236 236
237 // Indicates which channels are muted and can be ignored by the algorithm.
238 std::vector<bool> channel_mask_;
239
237 // After Initialize() has completed, all variables below must be accessed 240 // After Initialize() has completed, all variables below must be accessed
238 // under |lock_|. ------------------------------------------------------------ 241 // under |lock_|. ------------------------------------------------------------
239 base::Lock lock_; 242 base::Lock lock_;
240 243
241 // Algorithm for scaling audio. 244 // Algorithm for scaling audio.
242 double playback_rate_; 245 double playback_rate_;
243 std::unique_ptr<AudioRendererAlgorithm> algorithm_; 246 std::unique_ptr<AudioRendererAlgorithm> algorithm_;
244 247
245 // Simple state tracking variable. 248 // Simple state tracking variable.
246 State state_; 249 State state_;
(...skipping 43 matching lines...) Expand 10 before | Expand all | Expand 10 after
290 293
291 // NOTE: Weak pointers must be invalidated before all other member variables. 294 // NOTE: Weak pointers must be invalidated before all other member variables.
292 base::WeakPtrFactory<AudioRendererImpl> weak_factory_; 295 base::WeakPtrFactory<AudioRendererImpl> weak_factory_;
293 296
294 DISALLOW_COPY_AND_ASSIGN(AudioRendererImpl); 297 DISALLOW_COPY_AND_ASSIGN(AudioRendererImpl);
295 }; 298 };
296 299
297 } // namespace media 300 } // namespace media
298 301
299 #endif // MEDIA_RENDERERS_AUDIO_RENDERER_IMPL_H_ 302 #endif // MEDIA_RENDERERS_AUDIO_RENDERER_IMPL_H_
OLDNEW
« no previous file with comments | « media/filters/audio_renderer_algorithm_unittest.cc ('k') | media/renderers/audio_renderer_impl.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698