andrew@webrtc.org
3ce62fcfe4
Move merge_libs targets to their own gyp.
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The main reason is to depend on all ("*") targets in voice_engine.gyp and video_engine.gyp. We don't want the merge_lib targets building by default, since they do funny stuff like delete some libraries.
Review URL: http://webrtc-codereview.appspot.com/191003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@699 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 01:03:18 +00:00
kma@webrtc.org
af57de006a
Some code style changes in audio_processing/ns/main/source/ by Astyle,
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with a little manual modification.
Review URL: http://webrtc-codereview.appspot.com/201002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@698 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 23:36:01 +00:00
mallinath@webrtc.org
fa41d807a8
Fixes session state transition and registering observer.
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Review URL: http://webrtc-codereview.appspot.com/203001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@697 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 22:49:59 +00:00
henrik.lundin@webrtc.org
01ca01f6e6
Adding neteq_tests to modules tests
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Also moving neteq_tests.gyp and renaming to gypi. Cleaning up a
little in neteq_tests.gypi.
Review URL: http://webrtc-codereview.appspot.com/191004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@696 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 20:38:19 +00:00
mallinath@webrtc.org
29787c71a0
Changes to WebRtcSession after Provider(s) interface addition.
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Review URL: http://webrtc-codereview.appspot.com/201001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@695 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 18:52:26 +00:00
kma@webrtc.org
bbc1f10187
Changed modules/audio_processing/utility/Android.mk, to correct a build error in
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Android with the change from version r674.
Review URL: http://webrtc-codereview.appspot.com/197003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@694 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 18:09:02 +00:00
perkj@webrtc.org
487e401a27
Moving creation of sessiondescriptions to webrtcsession.
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Fixing defect durin close down in peerconnectionmanager.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/193004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@693 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 17:15:36 +00:00
kma@webrtc.org
bf39ff4271
Some general optimization in NS.
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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.
Note: In function WebRtcNsx_DataAnalysis, did the block separation because I found one "if" case is more frequent than "else" within a for loop; rest is kind of code re-aligning.
Review URL: http://webrtc-codereview.appspot.com/181002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@692 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 17:10:06 +00:00
kma@webrtc.org
a58224f9f0
Introduced a SPL inline function (multiple-accumulate), for preformance in ARMv7.
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It's used in quite some occations over many modules.
Review URL: http://webrtc-codereview.appspot.com/178004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@691 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 16:44:11 +00:00
perkj@webrtc.org
cb4ab65dfc
Moved creation of objects to the signaling thread.
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Fixed defect of not initializing remote_media_streams in peerconnection_impl.cc
Fixed defect in glare case of peerconnectionsignaling.cc
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/196001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@690 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 17:54:34 +00:00
mallinath@webrtc.org
bafca109db
Temp hook in WebRtcSession to VideoChannel.
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Review URL: http://webrtc-codereview.appspot.com/195001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@689 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 17:45:21 +00:00
stefan@webrtc.org
4b6f747373
Fixes a newly introduced bug in the jitter buffer where buffer reallocation
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causes corrupt pointers.
Review URL: http://webrtc-codereview.appspot.com/186003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@688 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 06:58:39 +00:00
stefan@webrtc.org
93d216c23f
Fixed bug in jitter buffer which caused the missingFrames bit to never be set.
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Also updated the VP8 wrapper to return fully concealed frames (for rendering).
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/190003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@687 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 06:48:11 +00:00
stefan@webrtc.org
61b4abf1f8
Proper use of frame rate argument in generic_codec_test.
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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
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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
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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.
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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.
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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.
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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.
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* 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.
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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.
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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.
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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.
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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.
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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.
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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.
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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
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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/
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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.
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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.
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- 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.
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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.
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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.
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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.
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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.
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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.
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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.
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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
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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
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* 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.
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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
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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.
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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.
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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.
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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.
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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
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@647 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-23 21:49:25 +00:00