bbc1f10187Changed modules/audio_processing/utility/Android.mk, to correct a build error in Android with the change from version r674. Review URL: http://webrtc-codereview.appspot.com/197003
kma@webrtc.org
2011-10-05 18:09:02 +00:00
487e401a27Moving creation of sessiondescriptions to webrtcsession. Fixing defect durin close down in peerconnectionmanager.
perkj@webrtc.org
2011-10-05 17:15:36 +00:00
bf39ff4271Some general optimization in NS. No big effort in introducing new style. Speed improved ~2%. Bit exact. Will introduce mulpty-and-accumulate and sqrt_floor next, which increase speed another 2% or so.
kma@webrtc.org
2011-10-05 17:10:06 +00:00
a58224f9f0Introduced a SPL inline function (multiple-accumulate), for preformance in ARMv7. It's used in quite some occations over many modules. Review URL: http://webrtc-codereview.appspot.com/178004
kma@webrtc.org
2011-10-05 16:44:11 +00:00
cb4ab65dfcMoved creation of objects to the signaling thread. Fixed defect of not initializing remote_media_streams in peerconnection_impl.cc Fixed defect in glare case of peerconnectionsignaling.cc
perkj@webrtc.org
2011-10-04 17:54:34 +00:00
ae7a0522c5video_coding robustness: Updating hybrid mode's settings 1. Disabling adjustment factor - temporary update. 2. Enabling a windowed filtered loss for the hybrid mode. Review URL: http://webrtc-codereview.appspot.com/192003
mikhal@webrtc.org
2011-10-03 22:54:34 +00:00
1b6ff7adbeConnecting PeerConnectionImpl with WebrtcSession and MediaStreamHandlers. This cl connects PeerConnectionImpl with WebrtcSession and MediaStreamHandlers.
perkj@webrtc.org
2011-10-03 22:50:04 +00:00
666f56bd41MediaStreamHandler implements eventhandlers for streams and tracks. Sets local and remote renderer and capture device.
perkj@webrtc.org
2011-10-03 21:55:17 +00:00
ed6d555775* Add the crypto serialize and deserialize. * Populate candidates test data.
wu@webrtc.org
2011-10-03 21:13:29 +00:00
ee2c391c15more webrtc session changes. Transport and TransportChannel handling is complete. Need work on session state. Review URL: http://webrtc-codereview.appspot.com/183005
mallinath@webrtc.org
2011-10-03 20:33:06 +00:00
99239d5a41First compiling version of peerconnection_client_dev using the new Peerconnection API. Links but does not work since the new peerconnection is under development. I would like to commit a version with as few changes as possible to the old peerconnection_client but using the new PeerConnection API.
perkj@webrtc.org
2011-10-03 15:59:40 +00:00
f458916145Returning errors if any of the Init() settings in VoE fail.
andrew@webrtc.org
2011-10-03 15:22:28 +00:00
5b91464edfAllow an aggregated partition to spill over to a new packet.
stefan@webrtc.org
2011-10-03 10:26:12 +00:00
5eec6cf29aStarted rewriting video_engine tests to use GUnit.
mflodman@webrtc.org
2011-09-29 12:24:13 +00:00
5045f671d0Add SignalUpdateSessionDescription to PeerConnectionSignaling. This is to allow webrtcsession to setup the mediachannels based on tracks.
perkj@webrtc.org
2011-09-28 23:06:46 +00:00
6b6d08164fRemove assert "currentVoEMicLevel <= kMaxVolumeLevel". We ran into an issue on a Linux system where the currentVoEMicLevel was in fact greater than the kMaxVolumeLevel. Therefore we are removing this assert and capping the currentMicLevel to the maxVolumeLevel when this case is detected. Review URL: http://webrtc-codereview.appspot.com/180001
punyabrata@webrtc.org
2011-09-28 17:45:03 +00:00
87d49798caAdd patterns for root_files (src/build/ and non-recursive contents of ./ and src/), common_audio, and audio_processing to WATCHLISTS. Review URL: http://webrtc-codereview.appspot.com/185001
andrew@webrtc.org
2011-09-28 15:04:36 +00:00
0beae6798dRemoved level estimator calls, since it is not supported. There are still one place left; used within SetRTPAudioLevelIndicationStatus(). The error return value of level_estimator() has no effect there.
bjornv@google.com
2011-09-28 14:08:19 +00:00
2d08d43206* Added modification of Start Bit Rate to vie_auto_test_custom_call * Added minor spacing and ":" for user input during vie_auto_test_custom_call * Changed the default Video Port to 11111 and Audio Port to be 11113 to bring it inline with the WindowsTest application for ViE Review URL: http://webrtc-codereview.appspot.com/181001
amyfong@webrtc.org
2011-09-27 17:46:45 +00:00
961885a8bbIn spl, introduced function WebRtcSpl_Sat32To16(), and changed file resample_by_2.c, both for optimization in ARMv7. Review URL: http://webrtc-codereview.appspot.com/140010
kma@google.com
2011-09-26 16:35:25 +00:00
e185e9f68avideo_coding: updates to jitter buffer logic: Make sure that every frame is inserted only once to the list. Review URL: http://webrtc-codereview.appspot.com/165001
mikhal@webrtc.org
2011-09-23 22:02:40 +00:00
713f91e12bFixed vie_autotest_custom_call.cc minor issues.
amyfong@webrtc.org
2011-09-23 16:41:26 +00:00
105ff39decvideo coding: updating offline tests. Additional clean-up to the offline test: Placing test callbacks in a designated file. Review URL: http://webrtc-codereview.appspot.com/167002
mikhal@webrtc.org
2011-09-23 16:41:11 +00:00
8e9e83b530This CL adds guards against division by zero, that should fix http://b/issue?id=5278531
bjornv@google.com
2011-09-23 12:39:47 +00:00
9e7774f163Added compare methods for TickInterval class. This is useful to be able to sort them using the STL algorithm library.
kjellander@webrtc.org
2011-09-23 11:33:31 +00:00
221b522118Return the number of /dev/video* without trying to open it.
wu@webrtc.org
2011-09-21 16:57:15 +00:00
c389aa2615Fix the bad video issue on Window client by increasing the rtp recv buffer size. Send buffer doesn't really matter, just to keep the same as talk does.
ronghuawu@google.com
2011-09-21 16:53:45 +00:00
65e6ab31ebTemporary log2 remove to build in chrome
bjornv@google.com
2011-09-21 11:56:46 +00:00
3be70ca17eAdded mute, hold and typing detect to voe_cmd_test to increase functionality in the voe_cmd_test application.
amyfong@webrtc.org
2011-09-20 23:41:06 +00:00
a1930427afWhen WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER is defined, we should never try to use _ptrCaptureDeviceInfo. Review URL: http://webrtc-codereview.appspot.com/167001
wu@webrtc.org
2011-09-20 17:38:57 +00:00
679e64d1fcCleaning up of Peerconnection API. Removing RemoteMediaStream. Adding one universal implementation of MediaStream that is used for both remote and local media streams. Removed AudioDevice and VideoDevice since VideoCaptureModule and AudioDeviceModule now is reference counted. Changes LocalAudioTrackImpl and LocalVideoTrackImpl to AudioTrackImpl and VideoTrackImpl so they can be used to repressent both remote and local tracks. Renamed files to a better name. Review URL: http://webrtc-codereview.appspot.com/151001
perkj@webrtc.org
2011-09-20 08:21:22 +00:00
c49db5ea48The files included in devicemanager.h/cc still have some conflict with chromium. Let's keep the devicemanager mods for now and I will see how can we solve this next. Review URL: http://webrtc-codereview.appspot.com/166001
wu@webrtc.org
2011-09-20 00:40:52 +00:00
cb99f78653* Update to use libjingle r85. * Remove (most of) local libjingle mods. Only webrtcvideoengine and webrtcvoiceengine are left now, because the refcounted module has not yet been released to libjingle, so I can't submit the changes to libjingle at the moment. * Update the peerconnection client sample app. Review URL: http://webrtc-codereview.appspot.com/151004
wu@webrtc.org
2011-09-19 21:59:33 +00:00
86b85db67eAdd missing intrinsic casts for VS 2005.
andrew@webrtc.org
2011-09-19 18:48:25 +00:00
b47d4b287dThis CL includes a move of the fixed point delay estimator from aecm to apm/utility. There has also been a code change that makes it possible to enable/disable the far end alignment, so that we save complexity when used as a quality metrics. Review URL: http://webrtc-codereview.appspot.com/135014
bjornv@google.com
2011-09-15 12:27:36 +00:00
29fd9a5f30Removing warnings in all NetEQ test targets
henrik.lundin@webrtc.org
2011-09-15 08:25:45 +00:00
mallinath@webrtc.org