c9cff24ff0Adding classes to be used for logging data within the engines and the components for offline processing. Data logged with these classes can conveniently be parsed and processed with e.g. Matlab. Review URL: http://webrtc-codereview.appspot.com/95009
stefan@webrtc.org
2011-08-29 07:39:02 +00:00
4094c49ddfTemporarily use digital AGC in WebRTC since Chromium can't support analog AGC. Fix suggested by henrika. Review URL: http://webrtc-codereview.appspot.com/121001
perkj@google.com
2011-08-29 07:36:28 +00:00
c9b75e0a4bremoving the warnings from the voe tests.
xians@google.com
2011-08-29 07:30:16 +00:00
36450af2b3Removing unsupported codecs from ptypes file
henrik.lundin@webrtc.org
2011-08-27 01:25:35 +00:00
92bace1fafHi, This CL will support negotiation of RTCP Mux feature. Earlier we were by default enabling and assuming remote end point will support this feature as well. This will also remove the maintaining of transport channels in WebRtcSession. Its left to cricket::Transport Review URL: http://webrtc-codereview.appspot.com/131005
mallinath@webrtc.org
2011-08-27 00:37:58 +00:00
a057a9561cvideo_coding: Updating protection logic in media optimization utility: 1. Changing protection logic structure: Accepts only one method (not a list) 2. Removed unused code (unreferenced protection methods) 3. Removed inline constructors/destructors. Review URL: http://webrtc-codereview.appspot.com/120005
mikhal@webrtc.org
2011-08-26 21:17:34 +00:00
552f173979video_coding: Moving video metrics computation to a designated file. This is the first stage of a general clean-up to test_util. Will try to divide this clean-up to small changes, so it will be easier to review. Review URL: http://webrtc-codereview.appspot.com/120004
mikhal@webrtc.org
2011-08-26 17:38:09 +00:00
c57f9c38adUsing IAudioEndpointVolume in IsSpeakerMuteAvailable and IsMicrophoneMuteAvailable to be consistent with SpeakerMute and MicrophoneMute APIs. Review URL: http://webrtc-codereview.appspot.com/112007
xians@webrtc.org
2011-08-26 12:28:33 +00:00
59af6f1434Porting Mac keypress detection from GIPS repository. Mac keypress detection was added specifically for GTalk. Review URL: http://webrtc-codereview.appspot.com/124001
zakkhoyt@google.com
2011-08-25 20:30:25 +00:00
102b2270c7First version of the peerconnection client application for Linux. I made several updates to the Windows version as well so that both implementations share a big portion of the code. The underlying PeerConnection notifications have changed a bit since the last update so that there's still a known issue that I plan to fix in my next change:
tommi@webrtc.org
2011-08-25 15:03:52 +00:00
137ece4ac3* Make GetReadyState accessible via the PeerConnection interface. * Update PeerConnection implementations to include "virtual" in the method declarations. * Add a check for a valid signaling thread in webrtcsession.cc. Review URL: http://webrtc-codereview.appspot.com/137001
tommi@webrtc.org
2011-08-25 14:18:25 +00:00
44d356d6dfFix unused variable warning in spatial_resampler.cc
stefan@webrtc.org
2011-08-25 07:53:53 +00:00
1cdc6b5d79This CL adding a factory class which has the responsibility of creating peerconnection objects. This is very basic class doesn't do any reference count, user has the responsibility to delete the object externally.
mallinath@webrtc.org
2011-08-24 23:50:05 +00:00
d1015fe677Replaced regular sleep with a talk_base::Thread::ProcessMessages(..) call so that Posts get some execution time from the main thread. Review URL: http://webrtc-codereview.appspot.com/122007
hellner@google.com
2011-08-24 21:35:09 +00:00
9788e18532* Add PeerConnectionProxy to forward all the API calls to signaling thread. * Use Send instead of Post so that we can report error. Review URL: http://webrtc-codereview.appspot.com/113009
wu@webrtc.org
2011-08-23 23:49:44 +00:00
4482b04207revert r430 to keep webrtc always ready to roll in chromium. r430 will be used when libvpx in chromium is rolled to Cayuga. Review URL: http://webrtc-codereview.appspot.com/119008
wjia@google.com
2011-08-23 23:41:00 +00:00
f9f1deba8fGet ready for libvpx Cayuga (v0.9.7-p1). When building with Chromium, on Windows, only header files are needed; otherwise, libvpx.gyp:libvpx is needed.
wjia@google.com
2011-08-23 23:08:30 +00:00
dec6aa57f3This CL will remove sending any signal after calling Close and RemoveStream. I am thinking to remove Close method at all, since application can directly delete the object if it wants to end the call with all active streams. Will send that change later in a different CL. Review URL: http://webrtc-codereview.appspot.com/119004
mallinath@webrtc.org
2011-08-23 22:17:03 +00:00
a386fc0a8bFixes build warnings due to unused variables.
hellner@google.com
2011-08-23 21:26:09 +00:00
87c9b74b11* Use the current thread as the signaling thread and worker thread to keep the unit test simple and easier to debug. * I also merged the issue 113007.
wu@webrtc.org
2011-08-23 20:57:29 +00:00
9d64705deaThe method AudioDeviceWindowsWave::RecProc can use uninitialized variables t1 and t2.
xians@webrtc.org
2011-08-23 09:14:56 +00:00
5895ea1573Fixes volume problem controls, happening with some Logitech headsets. Originally submitted as gips p4 depot CL 38122. Review URL: http://webrtc-codereview.appspot.com/116008
punyabrata@webrtc.org
2011-08-22 22:46:38 +00:00
9695e75fbdResolve a crash related to pulseAudio where we need to check if pa_context_get_source_info_by_name/pa_context_get_sink_info_by_name has early failure and returns NULL,then to avoid WaitForOperationCompletion from crashing, paOperation must be checked to ensure it is not NULL.
punyabrata@google.com
2011-08-22 22:35:14 +00:00
288c8698cbOptimization of 'cftmdl': * scalar optimization, vectorization. * 1.7% AEC overall speedup for the straight C path. * 9.2% AEC overall speedup for the SSE2 path. Review URL: http://webrtc-codereview.appspot.com/109008
cduvivier@google.com
2011-08-22 21:55:33 +00:00
eb29a9789d* Remove the previous renderer before set a new one. * Allow to unregister a renderer by giving a NULL point. Review URL: http://webrtc-codereview.appspot.com/123001
wu@webrtc.org
2011-08-22 15:58:03 +00:00
adb23827c1Fix windows build.
tommi@webrtc.org
2011-08-20 13:43:44 +00:00
f81f9f8c2aAdd -Werror and -Wextra to the Linux build.
andrew@webrtc.org
2011-08-19 22:56:22 +00:00
9139fddf0eOptimize ssim_8x8 for SSE2.
frkoenig@google.com
2011-08-19 22:33:08 +00:00
412889a2a9Some cleanup in test app This CL is to keep track of work and demonstrate a way to do ndk build in case it's needed. But ndk-build doesn't work yet because of many reasons, issues will be addressed in future if it's needed. Some minor changes in source files to make them pass compiler. Review URL: http://webrtc-codereview.appspot.com/107004
leozwang@google.com
2011-08-19 15:34:34 +00:00
54e4691e20change from ./test/data/voice_engine/audio_long16.pcm to ../../test/data/voice_engine/audio_long16.pcm Review URL: http://webrtc-codereview.appspot.com/115003
xians@google.com
2011-08-19 08:22:14 +00:00
765c918677Changes based on the review comments. * Rename WebRTCSession to WebRtcSession. * Add comments to the signal. Review URL: http://webrtc-codereview.appspot.com/114009
wu@webrtc.org
2011-08-19 00:14:23 +00:00
771ca422dfFixed assert error in media_opt_util that may have caused index for look-up table to be out of range. Review URL: http://webrtc-codereview.appspot.com/112005
marpan@google.com
2011-08-16 20:51:04 +00:00
stefan@webrtc.org