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gerrit-public.fairphone.software / platform / external / webrtc / 63e6072a4356a478dee8dce6cc87878dd77295cb / pc / test / fakeaudiocapturemodule_unittest.cc
  1. a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  2. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  3. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  4. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  5. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  6. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/test/fakeaudiocapturemodule_unittest.cc]
  7. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  8. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  9. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  10. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/test/fakeaudiocapturemodule_unittest.cc]
  11. 1aee0b5 Remove old methods in AudioTransport, make it pass a gn build by maxmorin · 8 years ago
  12. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 9 years ago
  13. ae69b02 Fix typo in FakeAdmTest.TestProcess name. by Peter Boström · 9 years ago
  14. a26ac92 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ ) by pbos · 9 years ago
  15. da33a8a Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ ) by torbjorng · 9 years ago
  16. f14c47a Remove ignored return code from modules. by Peter Boström · 9 years ago
  17. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  18. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc]
  19. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  20. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  21. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
  22. ee8c6d3 In PeerConnectionTestWrapper, put audio input on a separate thread. by deadbeef · 9 years ago
  23. 728d903 Reformat existing code. There should be no functional effects. by Peter Kasting · 10 years ago
  24. b7e5054 Match existing type usage better. by Peter Kasting · 10 years ago
  25. 8cf9bdb Remove USE_WEBRTC_DEV_BRANCH. by pbos@webrtc.org · 10 years ago
  26. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  27. 1972ff8 Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. by henrik.lundin@webrtc.org · 10 years ago
  28. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 11 years ago
  29. d852434 (Auto)update libjingle 71107853-> 71115715 by buildbot@webrtc.org · 11 years ago
  30. 94454b7 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 11 years ago
  31. cb711f7 Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 11 years ago
  32. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 12 years ago