generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
4,966 Commits 1 Branch 0 Tags 108 MiB
eec3843596
Commit Graph

4966 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
stefan@webrtc.org
61b4abf1f8 Proper use of frame rate argument in generic_codec_test. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@686 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:40:21 +00:00
mikhal@webrtc.org
e06be4f678 video coding tests: Adding ssimFrame to interface 
Review URL: http://webrtc-codereview.appspot.com/188004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@685 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:54:43 +00:00
mikhal@webrtc.org
ae7a0522c5 video_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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:54:34 +00:00
perkj@webrtc.org
1b6ff7adbe Connecting PeerConnectionImpl with WebrtcSession and MediaStreamHandlers. 
This cl connects PeerConnectionImpl with WebrtcSession and MediaStreamHandlers.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/190005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:50:04 +00:00
perkj@webrtc.org
666f56bd41 MediaStreamHandler implements eventhandlers for streams and tracks. 
Sets local and remote renderer and capture device.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/192002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 21:55:17 +00:00
wu@webrtc.org
236fcaa89a Interface changes after we have the Serialize and Deserialize. 
Review URL: http://webrtc-codereview.appspot.com/186004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 21:34:19 +00:00
wu@webrtc.org
ed6d555775 * Add the crypto serialize and deserialize. 
* Populate candidates test data.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 21:13:29 +00:00
mallinath@webrtc.org
ee2c391c15 more webrtc session changes. Transport and TransportChannel handling is complete. Need work on session state. 
Review URL: http://webrtc-codereview.appspot.com/183005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@679 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 20:33:06 +00:00
marpan@google.com
f1f3fb33b5 Update to rate-mismatch factor in media_opt_util. 
Review URL: http://webrtc-codereview.appspot.com/193003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 19:09:45 +00:00
perkj@webrtc.org
99239d5a41 First 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.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/183003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@677 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 15:59:40 +00:00
andrew@webrtc.org
f458916145 Returning errors if any of the Init() settings in VoE fail. 
There's no reason to try to continue if these simple settings fail; better to know about it immediately.

Also, readjusting the indentation to avoid breaking strings over several lines. This bends GStyle a bit, but it's well worth it to avoid the common "forgot to add a space" error.
Review URL: http://webrtc-codereview.appspot.com/173003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@676 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 15:22:28 +00:00
stefan@webrtc.org
5b91464edf Allow an aggregated partition to spill over to a new packet. 
Adds support for the case where the partition 0 and parts of partition 1
are transmitted in packet 1, and the end of partition 2 is transmitted
in packet 2.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 10:26:12 +00:00
bjornv@google.com
1ba3dbecbb Adds possibility to log delay estimates in AEC. 
Review URL: http://webrtc-codereview.appspot.com/178001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 08:18:10 +00:00
stefan@webrtc.org
f72c36763f Reverting changelist 666 since it broke the build on Mac. 
TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/187003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 07:37:41 +00:00
andrew@webrtc.org
6d169f2474 Fix Mac build error in vie_auto_test introduced in r666. 
COCOA_RENDERING was undefined. Committing without review.
Review URL: http://webrtc-codereview.appspot.com/191002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 06:00:42 +00:00
wu@webrtc.org
c93e36346b * Add Deserize for PeerConnectionMessage 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/189001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@671 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-30 18:08:51 +00:00
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