guoweis@webrtc.org
369a637ac8
Implemented Network::GetBestIP() selection logic as following.
...
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.
ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address
BUG=3808
At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.
R=jiayl@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7200
Committed: https://code.google.com/p/webrtc/source/detail?r=7201
Review URL: https://webrtc-codereview.appspot.com/31369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7216 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 22:37:29 +00:00
glaznev@webrtc.org
3b67f8e0ca
Enable HW video decoding on Qualcomm devices.
...
Parallel decoding and encoding problem is fixed now
(b/16353967), so it is possible to start using Qualcomm
VP8 HW decoder. Bitrate overshoots should be fixed as well.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7215 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 21:25:51 +00:00
henrike@webrtc.org
4a5061fbff
talk/p2p/base: removed unused variable "port_"
...
BUG=N/A
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7212 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 12:33:07 +00:00
andresp@webrtc.org
a74eda1b6f
Split video_capture_module specific implementation (external vs internal capture)
...
into its own targets. Dependencies must link directly with the desired one.
Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.
Targets linking with default/external capture implementation:
- anything dependent on webrtc_test_common
- anything dependent on video_engine_core
Targets linking with internal capture implementation:
- vie_auto_test
- anything dependent on webrtc_test_renderer
GN changes:
- Not many since there is almost no test definitions.
TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 11:50:19 +00:00
andresp@webrtc.org
85ef770d92
Split video engine android initialization into each internal module initialization.
...
This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on.
BUG=3768,3770
R=glaznev@webrtc.org , stefan@webrtc.org
TBR=henrike@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 11:44:51 +00:00
pbos@webrtc.org
ab990ae43a
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
...
Re-lands r7114 after landing r7204 to adress the compile error causing
the rollback in r7151.
BUG=3070
TBR=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 09:02:25 +00:00
buildbot@webrtc.org
6a9b155798
(Auto)update libjingle 75683337-> 75695882
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7206 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 08:08:38 +00:00
glaznev@webrtc.org
a59c501c99
Java VideoRenderer class may be backed by two different native
...
classes depending on type of rendering.
Fix crash in AppRtcDemo by calling correct destructor on exit.
BUG=
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 03:26:59 +00:00
guoweis@webrtc.org
40c2aa36f2
Implemented Network::GetBestIP() selection logic as following.
...
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.
ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address
BUG=3808
At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.
R=jiayl@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7200
Review URL: https://webrtc-codereview.appspot.com/31369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7201 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 20:29:41 +00:00
guoweis@webrtc.org
f8bff762d1
Implemented Network::GetBestIP() selection logic as following.
...
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.
ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address
BUG=3808
At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7200 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 20:17:22 +00:00
pbos@webrtc.org
cddd17c0f8
Recreate VideoStreams when setting resolution.
...
Instead of just changing resolution on the last stream streams are
reallocated to make sure that all streams are updated to match the
new input resolution.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7197 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 16:33:13 +00:00
pbos@webrtc.org
88e85ad64d
Add pbos@webrtc.org (myself) to talk/media/webrtc/.
...
Allows easier reviews of webrtcvideoengine2.cc and landing the new video
API on shorter review cycles.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7196 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 16:14:51 +00:00
buildbot@webrtc.org
80132e4d70
(Auto)update libjingle 75610402-> 75610402
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7194 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 15:24:15 +00:00
kjellander@webrtc.org
595b23c66f
Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..."
...
Breaks Chrome build and prevents rolling WebRTC into Chrome DEPS.
> Enable ipv6 by default for webrtc under a Finch experiment.
>
> BUG=413437 (chromium)
> https://code.google.com/p/chromium/issues/detail?id=413437
>
> Review URL: https://webrtc-codereview.appspot.com/23529005
TBR=guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7190 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 08:58:22 +00:00
andrew@webrtc.org
6ae5a6d7fe
Add a target for the approved subset of rtc_base.
...
rtc_base drags in a bunch of unwieldly dependencies (e.g. nss and
json) not required for standalone webrtc (aka rtc/media). The root of
the problem appears to be that MessageQueue depends on a socket server.
(And since common.h -> logging.h -> thread.h -> messagequeue.h, this
dependency spreads quickly.)
This starts a new target for a "purified" subset of rtc_base. It adds
the files which are already being used, replacing the use of common.h
with checks.h. desktop_capture is a lost cause, and retains its
dependency on the full rtc_base.
The hope is that as additional components are desired they will be
cleaned and added to rtc_base_approved.
BUG=3806
R=andresp@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7188 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 01:03:29 +00:00
glaznev@webrtc.org
996784548d
HW video decoding optimization to better support HD resolution:
...
- Change hw video decoder wrapper to allow to feed multiple input
and query for an output every 10 ms.
- Add an option to decode video frame into an Android surface object. Create
shared with video renderer EGL context and external texture on
video decoder thread.
- Support external texture rendering in Android renderer.
- Support TextureVideoFrame in Java and use it to pass texture from video decoder
to renderer.
- Fix HW encoder and decoder detection code to avoid query codec capabilities
from sw codecs.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7185 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 17:52:42 +00:00
guoweis@webrtc.org
cd309e3168
Enable ipv6 by default for webrtc under a Finch experiment.
...
BUG=413437 (chromium)
https://code.google.com/p/chromium/issues/detail?id=413437
Review URL: https://webrtc-codereview.appspot.com/23529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7184 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 16:31:13 +00:00
pbos@webrtc.org
000d86792d
Make BW checks > 0 in peerconnection_unittest.cc.
...
These checks (> 40k) fail on LSan FYI bots and the purpose of them seem
to be that we're getting non-zero BW reported.
R=stefan@webrtc.org
TBR=jiayl@webrtc.org , solenberg@webrtc.org
BUG=3817,chromium:375154
Review URL: https://webrtc-codereview.appspot.com/29479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7183 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 14:38:07 +00:00
henrike@webrtc.org
7f826350e3
Stop building talk/xmllite since it is no longer used.
...
BUG=3379
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7176 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 08:13:36 +00:00
buildbot@webrtc.org
a42a3ade54
(Auto)update libjingle 75390072-> 75428737
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7174 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-13 01:09:18 +00:00
fbarchard@google.com
7e31197cb2
Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..."
...
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
> Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
>
> Breaks other repos.
>
> TBR=fbarchard@google.com
> BUG=N/A
>
> Review URL: https://webrtc-codereview.appspot.com/23639004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7173 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-13 00:52:42 +00:00
glaznev@webrtc.org
192a54ff2f
Temporary revert maximum video codec resolution back to 1080p.
...
BUG=3757, 3738
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7171 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 16:40:35 +00:00
henrike@webrtc.org
3decd9b776
Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
...
Breaks other repos.
TBR=fbarchard@google.com
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/23639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7170 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 16:31:29 +00:00
buildbot@webrtc.org
ea77334c30
(Auto)update libjingle 75302540-> 75327856
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7160 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:52:48 +00:00
henrike@webrtc.org
1d8f780779
Stop building talk/sound since it is no longer used.
...
BUG=N/A
R=pbos@webrtc.org
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7156 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 17:16:56 +00:00
glaznev@webrtc.org
1d53f64b0f
Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD.
...
webrtc::VideoEngine::SetAndroidObjects and webrtc::VoiceEngine::SetAndroidObjects
are not compatible with WEBRTC_CHROMIUM_BUILD. Since neither VoiceEngine nor VideoEngine
are needed at the time it's better to disable it completely.
BUG=https://crbug.com/412276
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7155 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 16:58:25 +00:00
henrikg@webrtc.org
307d3dbdee
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
...
Speculative revert, seems to be reason for flaky Win FYI bot compile break.
> Expose VideoEncoders with webrtc/video_encoder.h.
>
> Exposes VideoEncoders as part of the public API and provides a factory
> method for creating them.
>
> BUG=3070
> R=mflodman@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/21929004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 09:48:30 +00:00
sprang@webrtc.org
c665dcb205
Revert 7145 "Stop building talk/sound since it is no longer used."
...
> Stop building talk/sound since it is no longer used.
>
> BUG=N/A
> R=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/22319004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7148 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 08:29:53 +00:00
henrik.lundin@webrtc.org
1972ff8a6e
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
...
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions). I've removed some of
these.
This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class. Removed "virtual" in those
cases.
BUG=none
TEST=none
R=andrew@webrtc.org , henrik.lundin@webrtc.org , mallinath@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 06:20:28 +00:00
henrike@webrtc.org
4c876453c8
Stop building talk/sound since it is no longer used.
...
BUG=N/A
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7145 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 22:18:04 +00:00
glaznev@webrtc.org
3472dcd7b0
Fix frame rate selection for Android camera.
...
- Android camera supports multiple fps values for a single video
resolution - change video source default video format selection
to pick up best available fps.
- Change fps range calculation to better match target fps value.
BUG=2622
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7142 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 19:24:57 +00:00
henrike@webrtc.org
b2efb6771c
Put base tests in webrtc_tests.gyp
...
BUG=N/A
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 17:28:19 +00:00
jiayl@webrtc.org
b6d69282f5
Enable shared socket for TurnPort.
...
In AllocationSequence::OnReadPacket, we now hand the packet to both the TurnPort and StunPort if the remote address matches the server address.
TESTED=AppRtc loopback call generates both turn and stun candidates.
BUG=1746
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7138 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 16:31:34 +00:00
buildbot@webrtc.org
5d639b3ef3
(Auto)update libjingle 75141932-> 75179475
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 07:57:12 +00:00
jiayl@webrtc.org
7d4891d3f1
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
...
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.
BUG=2108
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7068
Review URL: https://webrtc-codereview.appspot.com/16309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 21:43:15 +00:00
fbarchard@google.com
54cf1505e2
ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that.
...
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7121 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 18:34:53 +00:00
jiayl@webrtc.org
22406fcc9b
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
...
BUG=3570
R=juberti@webrtc.org , mallinath@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7070
Review URL: https://webrtc-codereview.appspot.com/20999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7120 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 15:44:05 +00:00
mallinath@webrtc.org
3d81b1b22a
Relanding https://code.google.com/p/webrtc/source/detail?r=7093 , after it got
...
reverted due to some internal compile failures.
In this CL changes are done in portallocator_unittest.cc, in particular to EXPECT_EQ checking in new tests.
Original patch committed in https://code.google.com/p/webrtc/source/detail?r=7093
TBR=juberti@webrtc.org
BUG=1179
Review URL: https://webrtc-codereview.appspot.com/22329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7118 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 14:38:10 +00:00
andresp@webrtc.org
4d19e05ab2
Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own.
...
This needs to happen sooner or later as if webrtc/base/checks.h happens to be included transitively here it would collide.
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7115 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 11:45:44 +00:00
pbos@webrtc.org
b420191743
Expose VideoEncoders with webrtc/video_encoder.h.
...
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.
BUG=3070
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 10:40:56 +00:00
henrike@webrtc.org
8b0b21161a
Revert 7093: "Implementing ICE Transports type handling in libjingle transport."
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TBR=mallinath@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/28419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7112 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 22:46:28 +00:00
pbos@webrtc.org
7118e61669
Finish work queue in SctpDataMediaChannelTest.
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Always finishing the work queue prevents memory leak detected in
LeakSanitizer (packet is deleted on the receiver side).
R=jiayl@webrtc.org
BUG=3608,chromium:375154
Review URL: https://webrtc-codereview.appspot.com/28399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7110 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 21:44:07 +00:00
jiayl@webrtc.org
0e52772aa9
Fix a bot-breaking memory leak from early returning in ParseMediaDescription.
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BUG=3791
R=henrike@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7109 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 21:43:43 +00:00
jiayl@webrtc.org
c172320bd2
Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android.
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This reverts commit r7068.
TBR=kjellander@webrtc.org
BUG=2108
Review URL: https://webrtc-codereview.appspot.com/23539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 20:44:36 +00:00
buildbot@webrtc.org
fd42f9dd6f
(Auto)update libjingle 74955991-> 75042522
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7106 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 19:45:36 +00:00
mallinath@webrtc.org
7256d31d28
Implementing ICE Transports type handling in libjingle transport.
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BUG=1179
R=juberti@webrtc.org , bemasc@webrtc.org , jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7093 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 04:08:44 +00:00
thorcarpenter@google.com
cc060563f3
Remove unnecessary include from testutils.cc.
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7090 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 21:19:00 +00:00
buildbot@webrtc.org
992febb997
(Auto)update libjingle 74873066-> 74873164
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:39:08 +00:00
thorcarpenter@google.com
a3344cfda4
Fix webrtcvideoframe tests.
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R=fbarchard@google.com , harryjin@google.com , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7088 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:34:13 +00:00
jiayl@webrtc.org
ddb85ab85b
Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07
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- SDP sctpmap attribute replaced with fmtp attribute
- SDP sctp-port attribute is newly added
BUG=3592
R=jiayl@webrtc.org , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7087 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:31:56 +00:00