pwestin@webrtc.org
8d80fa83fc
Fix for STL vector function data not available.
...
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1626004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4190 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-06 21:33:06 +00:00
pwestin@webrtc.org
d30859e58e
Connect ACM with RTP module for audio NACK.
...
Depends on http://review.webrtc.org/1507004/
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1613007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4189 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-06 21:09:01 +00:00
turaj@webrtc.org
a305e9612a
Nack for audio.
...
R=stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1507004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4188 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-06 19:00:09 +00:00
sergeyu@chromium.org
d9c4658756
Fix leaks in DesktopRegion
...
BUG=crbug.com/246870
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1615004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4186 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05 19:24:42 +00:00
fischman@webrtc.org
2b3a29a1fa
Implement DetectNumberOfCores on Android and make it consistent on Linux and Android
...
TEST=Number of cpu cores on Linux and Android is right
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1585007
Patch from Jeremy Mao <yujie.mao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4185 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05 16:37:42 +00:00
pwestin@webrtc.org
db24995680
Wire up Nack for Voe
...
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1614004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4184 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05 15:33:20 +00:00
pbos@webrtc.org
7f1b0ae888
Fix init list for VideoSendStream::Config::Rtp.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1616004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4183 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05 11:39:18 +00:00
pbos@webrtc.org
025f4f152b
Stats+Config moved into VideoSend/ReceiveStreams.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1561006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4182 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05 11:33:21 +00:00
kjellander@webrtc.org
fec34d7afa
Merge webrtc_utility_unittests into modules_unittests.
...
This CL eliminates the webrtc_utility_unittests test target.
NOTICE: Upon committing, this test must be removed from the
Buildbot configuration.
BUG=1843
TEST=trybots passing. Compiled and ran modules_unittests, verified the
AudioFrameOperationsTest test executes and passes.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1584004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4181 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-05 08:58:46 +00:00
andrew@webrtc.org
b2d29bd2fe
Restore relative include paths to libyuv.
...
Required in order to use an externally compiled libyuv. Removed
in https://code.google.com/p/webrtc/source/detail?r=4167
TBR=tnakamura
Review URL: https://webrtc-codereview.appspot.com/1611005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4180 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 23:53:48 +00:00
turaj@webrtc.org
3942f3a985
Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer.
...
bug=issue1847
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1585006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4178 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 21:31:22 +00:00
fbarchard@google.com
16d78bd307
Fix scale.cc build error with mingw64 -m32 gcc
...
BUG=571
TESTED=gcc scale.cc
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1613005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4177 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 19:41:00 +00:00
turaj@webrtc.org
9238de9d49
resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero.
...
Also solve DTMF playout with Opus.
issue=b9050210
Test=Manual by QA Team.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1583004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4176 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 19:18:39 +00:00
sergeyu@chromium.org
3d34f66292
Move screen capturers from chromium to webrtc.
...
R=alexeypa@chromium.org , wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1586005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4175 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 18:51:23 +00:00
fischman@webrtc.org
b7a8f43670
Roll chromium_revision in webrtc 199267:203806
...
This switches the default build system on linux from make to ninja. Details in
https://groups.google.com/a/chromium.org/forum/?fromgroups#!topic/chromium-dev/_Fsv4_XZ_bo
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1607004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4174 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 17:10:24 +00:00
kjellander@webrtc.org
430464c776
Add WebKit/Tools/Scripts to support Android test execution.
...
In https://code.google.com/p/webrtc/source/detail?r=4038 we rolled
chromium_revision past the point where WebKit/Tools/Scripts had its
own DEP in the Chromium DEPS file.
Since Chromium now only have a single WebKit checkout, we need to
pull the Tools/Scripts dir to be able to use the Android test
framework (build/android/run_test.py) since it's depending on modules
in webkitpy.
I have filed http://crbug.com/246529 to get this dependency removed.
BUG=1882
TEST=build/android/run_tests.py executes without any import errors.
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1608004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4173 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 16:29:45 +00:00
stefan@webrtc.org
a817962bab
Refactor padding and rtp header functionality.
...
BUG=1837
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1611004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4172 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 13:47:36 +00:00
stefan@webrtc.org
de98478965
Update the remote bitrate estimator before passing the packet to the RTP module.
...
This solves the problem of reconstructed packets biasing the bandwidth estimate.
TEST=vie_auto_test --automated, trybots
R=mflodman@webrtc.org , solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1594005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4171 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 12:15:40 +00:00
pbos@webrtc.org
6998c8ef7a
Remove XvRenderer.
...
One test renderer per platform is sufficient, multiple code paths are
bad.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1612004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4170 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 11:56:06 +00:00
stefan@webrtc.org
8ad3ec9722
Fix build error introduced with r4168.
...
TBR=mflodman@webrtc.org
BUG=1837
Review URL: https://webrtc-codereview.appspot.com/1610004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4169 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 09:52:46 +00:00
stefan@webrtc.org
c3cc375499
Add support for padding in pacer.
...
This improves pacer-based padding by making sure it limits padding according to:
- Never pad more than 800 kbps.
- Padding + media should not go above a given target bitrate.
Also adds appropriate unittests to make sure we reach the given targets.
BUG=1837
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1582005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4168 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 09:36:56 +00:00
pbos@webrtc.org
c69ae69d0b
Include files from webrtc/.. paths in common_video/
...
BUG=1662
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1546004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4167 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 09:02:37 +00:00
pbos@webrtc.org
ba7f6a8614
Include files from webrtc/.. paths in tools/
...
BUG=1662
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1547004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4166 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 08:14:10 +00:00
kjellander@webrtc.org
5156c94f89
Disable neteq_unittests on Win x64 in code.
...
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.
BUG=1460
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1595004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4165 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 05:47:24 +00:00
kjellander@webrtc.org
b6e49aa3f2
Disable audio_decoder_unittests on Win x64 in code.
...
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.
BUG=1459
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1594004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4164 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 05:47:04 +00:00
kjellander@webrtc.org
6eba2774c9
Disable audio_coding_unittests on Win x64 in code.
...
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.
BUG=1458
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1593004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4163 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 05:46:37 +00:00
fischman@webrtc.org
e001b57d84
Do not hold a lock when calling VCMReceiveCallback::FrameToRender.
...
DecodedImageCallback is allowed to be called on a thread different from decoding thread. To avoid the deadlock in VCMDecodedFrameCallback::Decoded, VCMDecodedFrameCallback::_critSect should not be held while calling VCMReceiveCallback::FrameToRender.
BUG=1832
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1570004
Patch from Wu-Cheng Li <wuchengli@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4162 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 03:29:37 +00:00
sergeyu@chromium.org
3ee13e4ac2
Optimized DesktopRegion implementation.
...
Now DestktopRegion can merge overlapping rectangles.
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1526004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4161 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 00:38:39 +00:00
fischman@webrtc.org
34a77354a8
Removed unused class members to enable clang=1 android build.
...
BUG=https://code.google.com/p/webrtc/issues/detail?id=1275
TESTED=video_demo_apk builds with clang=1
R=niklas.enbom@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1605004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4160 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 00:37:21 +00:00
mikhal@webrtc.org
6eb0f6a4d9
Setting SSRC in vie_loopback_test
...
BUG=1822
R=pwestin@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1603004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4159 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-03 22:54:40 +00:00
andrew@webrtc.org
0a38432ea5
Fix error in mixing test for supported sample rates.
...
With the switch to an arbitrary resampler, we now support these strange
rates.
TBR=turaj
Review URL: https://webrtc-codereview.appspot.com/1604004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4158 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-03 22:52:09 +00:00
wu@webrtc.org
fa64a595ad
Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling.
...
This makes it easier for the users of the interface, i.e. doesn't need to remember the id in order to disable audio level indication later.
BUG=1828
TEST=unit tests
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1598005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4157 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-03 21:27:57 +00:00
andrew@webrtc.org
c1eb560a5c
Replace the old resampler with SincResampler in the voice engine signal path.
...
* The old resampler was found to have a wraparound bug.
* Remove support for the old resampler from PushResampler.
* Use PushResampler in AudioCodingModule.
* The old resampler must still be removed from the file utility.
BUG=webrtc:1867,webrtc:827
TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio
R=henrika@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1590004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-03 19:00:29 +00:00
andrew@webrtc.org
31c5f1c91a
Remove ancient and unused CNG test.
...
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1585005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4154 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-03 16:07:07 +00:00
mikhal@webrtc.org
2b3a86554f
Revert 4149 "bug fixes for extremely large images - 10000x10000 ..."
...
> bug fixes for extremely large images - 10000x10000 and 100000 pixel wide.
> BUG=none
> TEST=libyuv unittest with manual LIBYUV_WIDTH=1000000
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1584008
TBR=fbarchard@google.com
Review URL: https://webrtc-codereview.appspot.com/1591005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4152 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 22:59:38 +00:00
niklas.enbom@webrtc.org
b35d2e3abc
Add dummy audio NACK APIs
...
R=pwestin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1579006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4151 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 21:13:52 +00:00
hclam@chromium.org
b1bba167f4
Prevent excessive logging in jitter buffer
...
Jitter buffer logs a message when it is going to recycle frames. This adds a
lot of noise even in normal operation. This change make sure only critical
cases are logged.
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1580007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4150 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 18:52:16 +00:00
fbarchard@google.com
85f28650d5
bug fixes for extremely large images - 10000x10000 and 100000 pixel wide.
...
BUG=none
TEST=libyuv unittest with manual LIBYUV_WIDTH=1000000
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1584008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4149 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 18:00:36 +00:00
fbarchard@google.com
a6494e6902
roll libyuv to r711 for scaler fix to webrtc unittests that scale up and down and check for fairly similar results.
...
BUG=none
TEST=try bots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1575005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4147 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 16:54:42 +00:00
tnakamura@webrtc.org
694cdc6e84
Revert 4104 "Refactor jitter buffer to use separate lists for de..."
...
Reason - leading suspect of video frame corruption tracked in http://b/9216252
Note that if this turns out to not be the cause, be sure to re-revert both this change and r4145.
> Refactor jitter buffer to use separate lists for decodable and incomplete frames.
>
> This changes the design of the jitter buffer to keeping track of decodable frames from the point when packets are inserted in the buffer, instead of searching for decodable frames when they are needed.
>
> To accomplish this the frame_list_, which previously contained all frames (incomplete or complete, continuous or not), is split into a list of decodable_frames_ (complete, continuous) and a list of incomplete_frames_ (either incomplete or non-continuous). These frame lists are updated every time a packet is inserted.
>
> This is another step in the direction of doing most of the work in the jitter buffer only once, when packets are inserted, instead of doing it every time we look for a frame or try to get a nack list.
>
> BUG=1798
> TEST=vie_auto_test, trybots
> R=mikhal@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1522005
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1586007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4146 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 16:09:48 +00:00
tnakamura@webrtc.org
4d9c07ad6d
Revert 4127 "Switch frame list implementation to std::map."
...
We want to revert r4104 for b/9216252, but because r4127 was built on top of r4104, we need to revert r4127 first. We'll un/re-revert this if we discover that r4104 is not to blame.
> Switch frame list implementation to std::map.
>
> This reduces the complexity of insert and find (by timestamp) from linear to logarithmic, which has a big impact on large frame lists.
>
> BUG=1726
> TEST=trybots, vie_auto_test --automated
> R=mikhal@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1561005
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1590005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4145 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 16:06:01 +00:00
braveyao@webrtc.org
5ed7051799
Apprtc: not to start the call until we get Turn response.
...
BUG=1795
Test=Manual Test
R=fischman@webrtc.org , juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/1528004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4144 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-31 06:29:41 +00:00
andrew@webrtc.org
f9f39d59d4
Add a drover.properties file for reference.
...
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1318005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4142 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-30 18:15:54 +00:00
andrew@webrtc.org
eed919d95d
MIPS optimizations for the following functions:
...
WebRtcSpl_ComplexBitReverse, WebRtcSpl_ComplexFFT, WebRtcSpl_ComplexIFFT, WebRtcSpl_DownsampleFast and WebRtcSpl_FilterARFastQ12.
Also, moved the common table used in complex_fft functions to a separate header file (webrtc/common_audio/signal_processing/include/complex_fft_tables.h).
R=andrew@webrtc.org , kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1126004
Patch from Ljubomir Papuga <lpapuga@mips.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4141 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-30 16:38:36 +00:00
mikhal@webrtc.org
adc64a7216
VCM/Timing: Setting clear names to members & methods
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R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1524004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4140 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-30 16:20:18 +00:00
vikasmarwaha@webrtc.org
fddf6be339
Updated apprtc to use new TURN format for chrome versions M28 & above.
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R=dutton@google.com , juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/1563004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4139 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 22:13:19 +00:00
jiayl@webrtc.org
046bc448d5
Fixes the frameRate stats by grouping the frames by timestamp.
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R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1536004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4138 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 16:33:46 +00:00
pbos@webrtc.org
4213633a4d
Use int for FPS instead of size_t.
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BUG=
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1578005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4136 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 15:13:12 +00:00
pbos@webrtc.org
a048d7cb0a
Include files from webrtc/.. paths in rtp_rtcp/
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BUG=1662
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1557004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 14:27:38 +00:00
stefan@webrtc.org
eea2622350
Correctly set SSRCs for extra send RTP modules.
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Fixes a regression introduced in r4096.
BUG=1845
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1585004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4134 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 14:07:54 +00:00