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Author SHA1 Message Date
kjellander@webrtc.org
d292b9c9da Unit tests now compile and run at all platforms. 
Cosmetic changes to mocks.h.

Review URL: http://webrtc-codereview.appspot.com/253003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 16:34:52 +00:00
niklas.enbom@webrtc.org
87c50f02c6 Adding author 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@870 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 12:19:22 +00:00
niklas.enbom@webrtc.org
3a9680b01b Adding author 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 09:48:06 +00:00
niklas.enbom@webrtc.org
0ba31331a8 Aligning license file with file header 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@868 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 09:31:39 +00:00
henrik.lundin@webrtc.org
895870b68f Adding marker bit to RTPanalyze results 
Review URL: http://webrtc-codereview.appspot.com/254005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 08:44:42 +00:00
mikhal@webrtc.org
bb8dfbdee2 updating vpm unit_test following r858 
Review URL: http://webrtc-codereview.appspot.com/255005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 22:07:16 +00:00
turaj@webrtc.org
7395d3d8e9 Addressing issue 115 http://code.google.com/p/webrtc/issues/detail?id=115 
Review URL: http://webrtc-codereview.appspot.com/261002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:33:06 +00:00
turaj@webrtc.org
fac5316856 Address the problem that iSAC could not go 16 kHz. It was addressed in P4 but not moved to svn. 
Review URL: http://webrtc-codereview.appspot.com/261001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:32:46 +00:00
turaj@webrtc.org
9116cf7c9b Have a guard on computing nrg to avoid wrap-around. This is discovered in a release test. During entropy coding of spectrum the value of "nrg" was too large and after shifting it became negative, resulting in decoder error. 
Review URL: http://webrtc-codereview.appspot.com/239016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:29:34 +00:00
mflodman@webrtc.org
29d75b3f7d Only allow increasing capture time. 
Review URL: http://webrtc-codereview.appspot.com/259001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:10:49 +00:00
andrew@webrtc.org
18ee6ec8e9 Use __inline in NS-fixed. 
The use of "inline" was failing to build on Windows.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/255003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:07:46 +00:00
andrew@webrtc.org
3119ecfec8 Fix audioproc build errors on Windows. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/254003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:00:18 +00:00
mikhal@webrtc.org
c4ab8706f4 video_processing: Adding logic to avoid a memcpy when not required 
Review URL: http://webrtc-codereview.appspot.com/255002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 16:44:24 +00:00
punyabrata@webrtc.org
0ab521f754 Resolving a crash related to strncopy followed by a strcat 
call. strncopy will not explicity copy or add a "\0" therefore
strcat did not know where to append the "\n" which was causing
an out of bounds crash.
Because we are checking the length, strcpy should be good enough
as it also copies the "\0". Please note that that I am pre-emptively
adding 2 instead of 1 to the length to take into account of the \n
that will be added later.
Review URL: http://webrtc-codereview.appspot.com/253004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 15:19:44 +00:00
perkj@webrtc.org
36a992b030 Merge streamparams and mediasession from libjingle and made necessary changes in peerconnection. 
-Removed ssrc from tracks.
-Updated PeerConnectionMessage parsing and serialization.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/239020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 11:46:56 +00:00
kjellander@webrtc.org
d6837709cf Fixing VPMUnitTest compilation error on Windows. 
It tried to include Visual Leak Detector which is not a tool that is installed/configured by default in the build.

Review URL: http://webrtc-codereview.appspot.com/257002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 01:51:10 +00:00
henrike@webrtc.org
b37c628ae4 Fixes crash due to r841. 
Review URL: http://webrtc-codereview.appspot.com/256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 23:53:04 +00:00
kma@webrtc.org
e9f909b575 Move the SetAndroidObjects to VideoCaptureFactory so that ViE can get access to it. 
Review URL: http://webrtc-codereview.appspot.com/244002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 22:24:57 +00:00
andrew@webrtc.org
f1a45d77fb Add missing <stdlib.h> to data_log test. 
BUG=
TEST=system_wrappers_unittests

Review URL: http://webrtc-codereview.appspot.com/256002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 21:44:54 +00:00
andrew@webrtc.org
3134aacd6b Use fileutils for the audio file in voe_auto_test. 
BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/250010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 21:31:07 +00:00
kma@webrtc.org
27957508a3 Changed Android makefile to make the lastest video render code run. 
Review URL: http://webrtc-codereview.appspot.com/247005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 21:29:50 +00:00
kjellander@webrtc.org
84736882ad Fixing system_wrappers unittests. 
Not complete, but enough to include them in the build again.

Review URL: http://webrtc-codereview.appspot.com/241008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 20:44:24 +00:00
henrike@webrtc.org
8885d22399 Review URL: http://webrtc-codereview.appspot.com/239015 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 20:35:13 +00:00
andrew@webrtc.org
1e10bb32b9 Remove global std::strings from fileutils. 
This is forbidden by the style guide and can cause the static
initialization order fiasco.

BUG=
TEST=test_support_unittests

Review URL: http://webrtc-codereview.appspot.com/248006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 20:22:02 +00:00
andrew@webrtc.org
2c74bab8b9 Remove unneeded assert and tracing. 
This is related to r840.

BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/239019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 19:54:20 +00:00
amyfong@webrtc.org
299e2c9ea4 vie_autotest_custom_call.cc - fixed VieAutotestDevcoderObserver to use const int for videoChannel for IncomingCodecChanged, RequestNewKeyFrame 
- this caused vie_auto_test to fail for Windows (but fine for Linux & Mac).
Review URL: http://webrtc-codereview.appspot.com/253001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 19:10:26 +00:00
henrike@webrtc.org
4d8c81878e The implementation before this change list keeps the ownership of memory that is used by peer connection instances in the peer connection manager. This means that if the peer connection manager is deleted before all peer connections it has created, these peer connections will be pointing to invalid memory. 
The solution in this CL is to create a bundle of the memory that needs to be alive as long as there are any peer connections or peer connection manager instances. This bundle is scoped reference counted so that it is deleted only when there are no references to it. This enables the peer connection and manager to be deleted in any order.
Review URL: http://webrtc-codereview.appspot.com/246003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 18:00:10 +00:00
kjellander@webrtc.org
177bb523bd Fixing system_wrappers unittests. 
Not complete, but enough to include them in the build again.

Review URL: http://webrtc-codereview.appspot.com/241008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 17:10:01 +00:00
henrike@webrtc.org
066f9e5a2f Ray, please verify that this cl fixes the issue. Once the verification has been made, please review: 
Henrik A: VoE
Andrew: audio_conference_mixer

Thanks!
Review URL: http://webrtc-codereview.appspot.com/241010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 23:15:47 +00:00
henrike@webrtc.org
731ecba47d Review URL: http://webrtc-codereview.appspot.com/251002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@840 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 22:49:24 +00:00
braveyao@webrtc.org
1f6d740571 This CL is about to manually reset the ShutdownRenderEvent at StopPlayout(). 
It could happen that if you want to restart playout, the new sponsored Render thread would catch this event
if the previous Render thread quits before this event is set.
With this modification, the device plugging out/in during talking would be supported well.
Review URL: http://webrtc-codereview.appspot.com/248002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 21:30:30 +00:00
wu@webrtc.org
88e0a34815 Remove duplicated code. 
Review URL: http://webrtc-codereview.appspot.com/251001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 17:29:44 +00:00
stefan@webrtc.org
f960211f8b Fixes two jitter buffer bugs related to NACK. 
Avoid decoding delta frames after a Flush() and after requesting
a key frame due to full NACK list.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 16:00:49 +00:00
perkj@webrtc.org
35a12cdf60 Fix comment. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 15:09:03 +00:00
perkj@webrtc.org
8129752c3b Add refcount and scoped_refptr. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 15:08:54 +00:00
perkj@webrtc.org
94cfde7c66 Removed scoped_refptr from libjingle.gyp 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 14:26:41 +00:00
perkj@webrtc.org
7e08613bda Move refcount and scoped_refptr to merge with libjingle. Deleted scoped_refptr_msg.h. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 14:26:25 +00:00
bjornv@webrtc.org
250cd6f41b Added a VAD unit test to common_audio. At this stage it runs through the API calls, but should later be complemented with tests on a file. 
Review URL: http://webrtc-codereview.appspot.com/243002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 12:45:58 +00:00
stefan@webrtc.org
eb65860720 Reverts the workaround in r823 and solves a macro bug. 
The macro bug caused frames to be dropped after being grabbed
for decoding.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 12:25:34 +00:00
tina.legrand@webrtc.org
8b1f621e3a Updated gypi for tests to work on osx. 
Review URL: http://webrtc-codereview.appspot.com/250002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 08:57:34 +00:00
andrew@webrtc.org
dfbebb916c Add a documented_interfaces watchlist. 
BUG=
TEST=watchlists.py

Review URL: http://webrtc-codereview.appspot.com/244013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 22:33:27 +00:00
amyfong@webrtc.org
ca4666b75c vie wintest added hybrid protection mode 
also fixed Max Framerate to reflect its actually the min framerate
Review URL: http://webrtc-codereview.appspot.com/244010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 21:16:40 +00:00
amyfong@webrtc.org
1e7e60b739 Fixed issue build failling due to vie_autotest_custom_call calling GetBandwidthUsage, which was 
changed in r822.
Review URL: http://webrtc-codereview.appspot.com/240014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 20:53:30 +00:00
amyfong@webrtc.org
51e1bb4e1a vie_autotest_customcall added encoder/decoder observer, maxBitrate set, print call statistics, enable kTraceAll 
When creating a new custom call, now able to set start bit rate (default is 1000)

The following modify call options were added
  9. Toggle Encoder Observer
 10. Toggle Decoder Observer
 12. Print Call Statistics

Also set the trace filter to kTraceAll

File defaults new call VGA (640x480)
Review URL: http://webrtc-codereview.appspot.com/239012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 18:16:35 +00:00
mikhal@webrtc.org
5200a05500 video_coding/jitter_buffer Updating condition on which we return a frame. 
Review URL: http://webrtc-codereview.appspot.com/240011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:54:54 +00:00
mikhal@webrtc.org
30f6376802 VP8: Updating codec version: VP8 version will now return the libvpx version used. 
Review URL: http://webrtc-codereview.appspot.com/247009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:45:00 +00:00
stefan@webrtc.org
2d28aff785 Workaround for an issue where frames are grabbed for decoding prematurely. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/240013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:13:18 +00:00
stefan@webrtc.org
fbea4e555d Solves two bandwidth estimation issues and measures the sent video bitrate. 
Issues solved:
1. Possible overflow when reducing the bandwidth estimate at the send-side
2. A burst of loss reports could make us reduce the rate way too far since
   we reduced the rate relative the current estimate and not the actual
   rate sent.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/244011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:08:29 +00:00
mflodman@webrtc.org
7e4269e9ee Changed VP8 qp min and added noise reduction. 
Review URL: http://webrtc-codereview.appspot.com/248003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 12:59:47 +00:00
mflodman@webrtc.org
8fc663b3ae Don't trigger false ViE SetReceiveCodec warning. 
Review URL: http://webrtc-codereview.appspot.com/250001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-26 11:30:52 +00:00