generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
650 Commits 1 Branch 0 Tags 108 MiB
e90265bd1a
Commit Graph

650 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
tommi@webrtc.org
e90265bd1a Commit http://webrtc-codereview.appspot.com/191001/ 
Review URL: http://webrtc-codereview.appspot.com/192001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-30 13:26:14 +00:00
perkj@webrtc.org
e804ee1a80 This patch hooks up PeerConnectionImpl to PeerConnectionSignaling. 
Implements
virtual bool ProcessSignalingMessage(const std::string& msg);
virtual scoped_refptr<StreamCollection> remote_streams();
virtual void CommitStreamChanges();

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/187001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-29 22:27:54 +00:00
wu@webrtc.org
78083bf750 * Add Serialize functions to PeerConnectionMessage. 
* Separated file for PeerConnectionMessage.
* Update to the latest and fix compiling errors
Review URL: http://webrtc-codereview.appspot.com/182002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-29 19:11:52 +00:00
mallinath@webrtc.org
9a1249d9e0 first cut of webrtcsession. Doesn't do much other than creating files and empty function bodies. 
Review URL: http://webrtc-codereview.appspot.com/186002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-29 18:15:21 +00:00
mflodman@webrtc.org
5eec6cf29a Started rewriting video_engine tests to use GUnit. 
- Added comments to the new test.
- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/168002
Patch from Patrik Hoglund <phoglund@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-29 12:24:13 +00:00
perkj@webrtc.org
5045f671d0 Add SignalUpdateSessionDescription to PeerConnectionSignaling. 
This is to allow webrtcsession to setup the mediachannels based on tracks.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/184001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 23:06:46 +00:00
punyabrata@webrtc.org
6b6d08164f Remove 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 17:45:03 +00:00
kma@google.com
c611b1a950 Bit-exact with non-Neon version. 
Review URL: http://webrtc-codereview.appspot.com/180002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 16:03:38 +00:00
andrew@webrtc.org
87d49798ca Add 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 15:04:36 +00:00
bjornv@google.com
0beae6798d Removed 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. 
The VoE auto tests have been updated as well.
Review URL: http://webrtc-codereview.appspot.com/178003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 14:08:19 +00:00
perkj@webrtc.org
2f56ff48a4 Implementation of PcSignaling. A Class to handle signaling between peerconnections. 
Review URL: http://webrtc-codereview.appspot.com/149002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 20:35:37 +00:00
andrew@webrtc.org
18421f2063 Remove unnecessary include from NS interface. 
http://code.google.com/p/webrtc/issues/detail?id=46
Review URL: http://webrtc-codereview.appspot.com/183001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 19:50:52 +00:00
amyfong@webrtc.org
6a23ad5702 Fixed the CameraCap button to say Version, also change the function name inside ChannelDlg.cpp 
Review URL: http://webrtc-codereview.appspot.com/182001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 19:19:10 +00:00
amyfong@webrtc.org
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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 17:46:45 +00:00
mikhal@webrtc.org
848fad23c6 video_coding: Updating media opt test - fixing call to protection callback. 
Review URL: http://webrtc-codereview.appspot.com/179003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 16:30:59 +00:00
xians@google.com
49d025f262 Get the right guid str for GetRecordingDeviceName 
Bug=http://code.google.com/p/webrtc/issues/detail?id=99
Test=none
Review URL: http://webrtc-codereview.appspot.com/183002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 14:43:06 +00:00
andrew@webrtc.org
82f66a776f Return to the WebM git repository for libvpx. 
This slows a warm gclient sync by about 0.3 s on my Linux machine. gclient seems to treat git tags and commit hashes identically, so the readable tag is preferred.
Review URL: http://webrtc-codereview.appspot.com/179002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 10:47:25 +00:00
bjornv@google.com
a2c6ea09b0 Removed a segmentation fault error when processing near_file only. 
Review URL: http://webrtc-codereview.appspot.com/174001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 08:04:45 +00:00
kma@google.com
961885a8bb In 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-26 16:35:25 +00:00
mikhal@webrtc.org
e185e9f68a video_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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 22:02:40 +00:00
turajs@google.com
cf136186f5 Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:49:25 +00:00
turajs@google.com
13335ccd7a Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:47:25 +00:00
turajs@google.com
610f478705 Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@645 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:45:34 +00:00
turajs@google.com
53439d9982 Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@644 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:44:00 +00:00
amyfong@webrtc.org
713f91e12b Fixed vie_autotest_custom_call.cc minor issues. 
1. mirror of local render removed
2. the video device the user selected wasn't what was actually being used when the call is being made
3. fixed mentions of loopback calls
Review URL: http://webrtc-codereview.appspot.com/171001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@643 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 16:41:26 +00:00
mikhal@webrtc.org
105ff39dec video 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@642 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 16:41:11 +00:00
turajs@google.com
496ef8aca8 To fix warnings in test files. 
Review URL: http://webrtc-codereview.appspot.com/169001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 15:45:48 +00:00
bjornv@google.com
8e9e83b530 This CL adds guards against division by zero, that should fix http://b/issue?id=5278531 
In addition a read outside memory event has been detected and removed.
Also an improper noise weighting has been corrected.
Review URL: http://webrtc-codereview.appspot.com/152001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@640 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 12:39:47 +00:00
kjellander@webrtc.org
9e7774f163 Added compare methods for TickInterval class. 
This is useful to be able to sort them using the STL algorithm library.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/173002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 11:33:31 +00:00
kjellander@webrtc.org
dca57bddf8 Adding git ignore file. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/173001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 11:15:35 +00:00
bjornv@google.com
dc743a8bba Replaces a use of log2. 
I've replaced a log2 operation so it works on Windows.
Review URL: http://webrtc-codereview.appspot.com/171002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@637 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 08:13:53 +00:00
leozwang@google.com
90eff6c7c6 Fix compilation error in build-in AEC test 
Review URL: http://webrtc-codereview.appspot.com/164001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 18:02:03 +00:00
wu@webrtc.org
221b522118 Return the number of /dev/video* without trying to open it. 
Consider the case when there're /dev/video0 and /dev/video1. But for somereason the video0 is not in a correct state and can't be open. As a result, current NumberOfDevices will return 1, which is fine. However, we will then never be able to get the device we really want - /dev/video1. Consider the code below, the GetCaptureDevice will fail because it calls into DeviceInfoLinux::GetDeviceName(0, ...) which will again try to open the /dev/video0. So the root cause is the mismatching of the NumberOfDevices and GetDeviceName.

Since we will open the device in DeviceInfoLinux::GetDeviceName anyway, I think we should return the number of /dev/video* in DeviceInfoLinux::NumberOfDevices without trying to open it. Otherwise the DeviceInfoLinux::NumberOfDevices should return more information like which /dev/video* is valid which is not.

bool found = false;
for (int i = 0; i < vie_capture->NumberOfCaptureDevices(); ++i) {
  if (vie_capture->GetCaptureDevice(i, ...) == 0) {
    found = true;
    break;
  }
}
Review URL: http://webrtc-codereview.appspot.com/148004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@635 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 16:57:15 +00:00
ronghuawu@google.com
c389aa2615 Fix 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.

The same fix is submitted to libjingle for reivew. But I think it's worth to fix it here too as
it may take while for webrtc to get from libjingle. This patch is slightly different then that
one as I don't want to add the webrtcvideoengine.h back to webrtc.
Review URL: http://webrtc-codereview.appspot.com/166002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 16:53:45 +00:00
bjornv@google.com
65e6ab31eb Temporary log2 remove to build in chrome 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 11:56:46 +00:00
amyfong@webrtc.org
3be70ca17e Added mute, hold and typing detect to voe_cmd_test to increase functionality in the voe_cmd_test application. 
Typing Detect is applicable only for Mac.  
Review URL: http://webrtc-codereview.appspot.com/156002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@632 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 23:41:06 +00:00
wu@webrtc.org
a1930427af When WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER is defined, we should never try to use _ptrCaptureDeviceInfo. 
Review URL: http://webrtc-codereview.appspot.com/167001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 17:38:57 +00:00
leozwang@google.com
657f483c26 Fix compilation error 
Review URL: http://webrtc-codereview.appspot.com/162003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@630 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 16:41:20 +00:00
leozwang@google.com
ec5e87614e Enable OPENELSE defination when compile voice engine 
Review URL: http://webrtc-codereview.appspot.com/150005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 16:41:09 +00:00
pwestin@webrtc.org
741da942ec Added support for new RTCP message REMB (remote estimated max bitrate) 
Review URL: http://webrtc-codereview.appspot.com/149001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 13:52:04 +00:00
perkj@webrtc.org
679e64d1fc Cleaning 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@627 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 08:21:22 +00:00
wu@webrtc.org
c49db5ea48 The 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 00:40:52 +00:00
wu@webrtc.org
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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 21:59:33 +00:00
andrew@webrtc.org
86b85db67e Add missing intrinsic casts for VS 2005. 
Allows re-enabling SSE optimization on Windows.
Review URL: http://webrtc-codereview.appspot.com/161003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 18:48:25 +00:00
leozwang@google.com
522f42bb80 Add kPlatformAndroid to platform check function 
Review URL: http://webrtc-codereview.appspot.com/161002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@622 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 17:39:05 +00:00
andrew@webrtc.org
ed083d4079 Modify the _vadActivity member of the AudioFrame passed to AudioProcessing. 
This saves the user from having to explicitly check stream_has_voice(). It will allow typing detection to function, which relies on this behaviour.
Review URL: http://webrtc-codereview.appspot.com/144004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 15:28:51 +00:00
andrew@webrtc.org
94c7413b0d Allow echo metrics to be enabled in process_test. 
Review URL: http://webrtc-codereview.appspot.com/155002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 15:17:57 +00:00
henrik.lundin@webrtc.org
4c36d3b424 Fixing windows warnings in rtp_utility 
Adding explicit casting to bool to avoid warnings when compiling
in windows.

Review URL: http://webrtc-codereview.appspot.com/140002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 08:16:20 +00:00
stefan@webrtc.org
dba7a3abd6 Updating WATCHLIST with a video_coding watch and adding myself to it. 
Review URL: http://webrtc-codereview.appspot.com/159001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@618 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 07:50:48 +00:00
andrew@webrtc.org
67812a4621 Temporarily disabling SSE2 on Windows again until we can build on VS 2005. 
Skipping review because the build is broken on Windows.
Review URL: http://webrtc-codereview.appspot.com/156003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 02:28:49 +00:00