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734 Commits

Author SHA1 Message Date
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." 
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. 
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. 
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. 
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 
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. 
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. 
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 
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. 
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 
- 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
buildbot@webrtc.org
af5fa95258 (Auto)update libjingle 74857067-> 74860820 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 13:03:50 +00:00
buildbot@webrtc.org
7e3bd3d7de (Auto)update libjingle 74851128-> 74857067 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 11:45:42 +00:00
buildbot@webrtc.org
bc6fa1876e (Auto)update libjingle 74825992-> 74851128 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7082 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 11:08:01 +00:00
buildbot@webrtc.org
818b7b3ac9 (Auto)update libjingle 74825084-> 74825992 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 00:14:03 +00:00
jiayl@webrtc.org
dfbcf8161e Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice. 
BUG=3778
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 00:01:12 +00:00
henrike@webrtc.org
f1427c6731 Revert 7070 "TurnPort should retry allocation with a new address on error 
STUN_ERROR_ALLOCATION_MISMATCH."

TBR=jiayl@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/15359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 22:21:33 +00:00
glaznev@webrtc.org
4b234044d5 Reduce maximum video resolution for Android. 
HW video encoder and decoder can not be initialized
with 3840x2160 resolution.

BUG=3757,3738
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 19:50:07 +00:00
jiayl@webrtc.org
574f2f60fe TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. 
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 19:11:34 +00:00
jiayl@webrtc.org
52055a276d 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

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 17:12:25 +00:00