Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(120)

Side by Side Diff: talk/app/webrtc/test/fakeaudiocapturemodule.h

Issue 1613643004: Remove mutable from rtc::CriticalSection members. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months 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
« no previous file with comments | « no previous file | talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 260 matching lines...) Expand 10 before | Expand all | Expand 10 after
271 // Buffer for samples to send to the webrtc::AudioTransport. 271 // Buffer for samples to send to the webrtc::AudioTransport.
272 char send_buffer_[kNumberSamples * kNumberBytesPerSample]; 272 char send_buffer_[kNumberSamples * kNumberBytesPerSample];
273 273
274 // Counter of frames received that have samples of high enough amplitude to 274 // Counter of frames received that have samples of high enough amplitude to
275 // indicate that the frames are not faked somewhere in the audio pipeline 275 // indicate that the frames are not faked somewhere in the audio pipeline
276 // (e.g. by a jitter buffer). 276 // (e.g. by a jitter buffer).
277 int frames_received_; 277 int frames_received_;
278 278
279 // Protects variables that are accessed from process_thread_ and 279 // Protects variables that are accessed from process_thread_ and
280 // the main thread. 280 // the main thread.
281 mutable rtc::CriticalSection crit_; 281 rtc::CriticalSection crit_;
282 // Protects |audio_callback_| that is accessed from process_thread_ and 282 // Protects |audio_callback_| that is accessed from process_thread_ and
283 // the main thread. 283 // the main thread.
284 rtc::CriticalSection crit_callback_; 284 rtc::CriticalSection crit_callback_;
285 }; 285 };
286 286
287 #endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ 287 #endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
OLDNEW
« no previous file with comments | « no previous file | talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698