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147 Commits

Author SHA1 Message Date
pbos@webrtc.org
058b1f17ac Remove GetReceiveBandwidthEstimatorStats. 
Removes unnecessary non-standard stats that we don't really make use of.

BUG=
R=pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47379004

Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 08:55:16 +00:00
hbos@webrtc.org
4aef5fef18 Add thread checks to the CaptureManager. 
It looks like it is being used single threadedly, except that in some cases it is created and/or destroyed in threads other than the one running its operations. As such, CaptureManager() contains 'thread_checker_.DetachFromThread()' and ~CaptureManager() does not have a DCHECK.

BUG=
R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36279004

Cr-Commit-Position: refs/heads/master@{#8498}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-25 10:09:45 +00:00
hbos@webrtc.org
1e64263b90 Thread-safe ChannelManager.GetSupportedFormats, used by VideoSource 
VideoSource was using VideoCapturer's GetSupportedFormats in a non-thread safe manner.
Now this is handled to (new method) ChannelManager.GetSupportedFormats.

BUG=
R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42079004

Cr-Commit-Position: refs/heads/master@{#8495}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-25 09:50:22 +00:00
lally@webrtc.org
a747093334 After another round of reviews. 
Cr-Commit-Position: refs/heads/master@{#8483}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8483 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-24 20:20:19 +00:00
lally@webrtc.org
ec97c6516f Attempt on read-only acceptance of -12. 
Cr-Commit-Position: refs/heads/master@{#8477}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-24 20:19:29 +00:00
minyue@webrtc.org
f9b5c1b3d0 Removing CELT. 
CELT is not supported in WebRTC/Libjingle. There are a few left-over in our code base. They are cleaned up in this CL.

BUG=
R=pbos@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36099004

Cr-Commit-Position: refs/heads/master@{#8385}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 12:37:14 +00:00
henrika@webrtc.org
62f6e75673 Refactoring WebRTC Java/JNI audio recording in C++ and Java. 
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC:

- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup

Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).

BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33969004

Cr-Commit-Position: refs/heads/master@{#8325}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-11 08:39:19 +00:00
pkasting@chromium.org
0e81fdf5d2 Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. 
BUG=chromium:82439
TEST=none
R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40569004

Cr-Commit-Position: refs/heads/master@{#8229}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-02 23:54:40 +00:00
honghaiz@google.com
a67ca1a3bb Only report the first rtp packet because it indicates the media has started flowing. 
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37829004

Cr-Commit-Position: refs/heads/master@{#8189}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 19:48:40 +00:00
tommi@webrtc.org
586f2eda0d Change GetStreamBySsrc to not copy StreamParams. 
This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple.  Also, we can use lambdas now :)

BUG=
R=perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 23:00:41 +00:00
jlmiller@webrtc.org
5f93d0a140 Update libjingle license statements at top of talk files for consistency 
BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-20 21:36:13 +00:00
pbos@webrtc.org
c62749fb47 Parallelize MediaRecorder unittests. 
Exchanging static filenames for temporary ones, permitting tests to be
run in parallel without conflicting parallel uses of the same filenames.

TBR=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 -r 100 out/Debug/libjingle_p2p_unittest

Review URL: https://webrtc-codereview.appspot.com/34589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-02 09:01:20 +00:00
pthatcher@webrtc.org
5ad4178137 Move the Jingle-specific network code into webrtc/libjingle. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-23 22:14:15 +00:00
pthatcher@webrtc.org
4c0544ab07 Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository. 
Also, fix the includes and header guards of examples/call.

R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 22:29:55 +00:00
pthatcher@webrtc.org
5647877b2d Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 03:32:59 +00:00
pthatcher@webrtc.org
aacc23465b Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
(This is the 3rd try)

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 20:31:29 +00:00
pthatcher@webrtc.org
f5847d7746 Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well. 
R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 17:09:11 +00:00
pbos@webrtc.org
a9cf079248 Rename external_hmac_ctx_t to ExternalHmacContext. 
_t types are reserved by POSIX.

R=juberti@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/33699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7944 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:12:21 +00:00
pthatcher@webrtc.org
4cb3856a4d Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." 
This reverts r7939 because it broke Chromium and other depedent projects that rely on certain logic remaining in p2p/base/session.cc and not in webrtc/libjingle/session.cc.

BUG=
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 02:28:25 +00:00
pthatcher@webrtc.org
536f999e58 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
This is an un-revert of r7992 and r7993.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 01:22:02 +00:00
pbos@webrtc.org
e728ee03ba Remove or rename typedefs with _t prefixes. 
_t prefixes are reserved for additional typenames in POSIX.

R=henrik.lundin@webrtc.org, hta@webrtc.org, stefan@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/36559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 13:43:55 +00:00
pthatcher@webrtc.org
f050791ba0 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." 
This reverts r7992.

It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 22:28:03 +00:00
pthatcher@webrtc.org
4afb59903c Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7922 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 21:37:37 +00:00
pthatcher@webrtc.org
e2b7585bc2 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. 
R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 21:09:08 +00:00
pbos@webrtc.org
18a3896bd2 Revert r7886:7887. 
Broke build steps in other code that uses securetunnelsessionclient.cc
and others.

TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/36439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 07:03:04 +00:00
pthatcher@webrtc.org
dee76f3b89 Move the obvious/easy Jingle-specific code into webrtc/libjingle. 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 21:04:42 +00:00
pthatcher@webrtc.org
40b276ea7b Cleanup little things found when refactoring. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/33519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 02:44:30 +00:00
magjed@webrtc.org
35c1ace185 Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..." 
Reason for revert is failed testcases:
WebRtcVideoEngineExtendedTestFake.ResetSimulcastSendCodecOnNewFrameSize
WebRtcVideoEngineExtendedTestFake.MultipleSendStreamsDifferentFormats

> WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
> 
> BUG=3936
> R=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/30039004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 16:21:49 +00:00
magjed@webrtc.org
52da44b7e6 WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution 
BUG=3936
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 15:43:11 +00:00
henrik.lundin@webrtc.org
f85dbce041 Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" 
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.

BUG=3951
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 12:25:00 +00:00
henrike@webrtc.org
269fb4bc90 move xmpp and p2p to webrtc 
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-28 22:20:11 +00:00
tommi@webrtc.org
f15dee6980 Check if a datachannel in the current local description is an sctp channel before assuming rtp. 
When generating an offer from a local description when 'sctp' is not explicitly set in the
media session options, we were generating an offer with an RTP datachannel even though the
channel in the local description was already sctp.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 22:15:04 +00:00
pthatcher@webrtc.org
2e7ee4b28b Fix the SrtpFilter crash caused by two local offers. 
BUG=http://crbug.com/421774
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 16:10:29 +00:00
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." 
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 22:03:39 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. 
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 17:30:28 +00:00
buildbot@webrtc.org
81ddc78536 (Auto)update libjingle 77701902-> 77709729 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 22:39:24 +00:00
buildbot@webrtc.org
1ecbe45c7e (Auto)update libjingle 77689511-> 77696841 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 20:29:28 +00:00
jiayl@webrtc.org
742922b313 Make the media content send only if offerToReceive is false while local streams exist. 
We previously do not add the media content if offerToReceive is false.

BUG=3833
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 21:32:43 +00:00
jiayl@webrtc.org
7dfb7fa189 Reland disallowing blocking calls on the worker thread. 
This fixed the issue that invoking the call when the thread is not started.

BUG=3559
R=juberti@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/24769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 22:45:55 +00:00
pbos@webrtc.org
34f2a9ea72 Initialize SSL in unittest_main.cc. 
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-28 11:36:45 +00:00
thorcarpenter@google.com
a21d071607 Reverting part of 
https://webrtc-codereview.appspot.com/15089004/diff/140001/talk/session/media/channelmanager.cc?context=10&column_width=80
because of a major regression hanging the executable on start.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-26 17:19:14 +00:00
pbos@webrtc.org
05305116d6 Explicitly initialize SSL for tests. 
Adding missing SSL initialization/cleanups in
TransportDescriptionFactoryTest and MediaSessionTest.

These being missing prevent these tests from being run individually
without other tests preceding them that initialize SSL.

BUG=3860
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-25 15:50:26 +00:00
jiayl@webrtc.org
3987b6de50 Fix a problem in Thread::Send. 
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.

BUG=3559
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 17:14:05 +00:00
pbos@webrtc.org
d60d79a145 Thread annotation of rtc::CriticalSection. 
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.

This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.

R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 07:10:57 +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
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
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
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
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
buildbot@webrtc.org
fa4535b270 (Auto)update libjingle 74694022-> 74696326 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 16:49:04 +00:00