Commit Graph

24 Commits

Author SHA1 Message Date
Thiago Farina
91543731c3 Do not define POSIX.
It breaks integration with upstream re2 library on Chromium.

Without patching re2 library, with this define, it produces the
following error:

../../third_party/re2/re2/re2.h:254:5: error: expected identifier
    POSIX, // POSIX syntax, leftmost-longest match

As we define POSIX on the command line, the C preprocessor changes
RE2::POSIX to nothing and thus break the compilation. :(

See chromium-dev mailing list for this discussion in
https://groups.google.com/a/chromium.org/d/topic/chromium-dev/UXCHnX7pV44/discussion

BUG=None
TEST=ninja -C out/Debug, everything compiles as before
R=sergeyu@chromium.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46049004

Patch from Thiago Farina <tfarina@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#9032}
2015-04-20 11:14:22 +00:00
Thiago Farina
cb76b89572 Cleanup: Move json.h into rtc namespace.
This should fix the TODO in that header.

BUG=None
TEST=ninja -C out/Debug still compiles everything.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47919004

Patch from Thiago Farina <tfarina@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8921}
2015-04-02 09:59:23 +00:00
guoweis@webrtc.org
00c509ad1c Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

Tested with peerconnection_client on windows, AppRTCDemo on Mac.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8660

Committed: https://code.google.com/p/webrtc/source/detail?r=8661

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8705}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8705 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-12 21:38:19 +00:00
braveyao@webrtc.org
9bfa5f0405 In r8605, DTLS is enabled by default for native webrtc. So we have to disable it explicitly in peerconnection example for loopback test.
BUG=4386
TEST=Manual Test
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44599004

Cr-Commit-Position: refs/heads/master@{#8677}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8677 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 03:21:41 +00:00
braveyao@webrtc.org
c68e0c9dfe Fix cpplint warning in the previous cl to peerconnection client example.
BUG=3872
TEST=Manual Test + AutoTest
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40949004

Cr-Commit-Position: refs/heads/master@{#8525}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8525 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 09:51:43 +00:00
decurtis@webrtc.org
bfa3c7253f Don't call g_thread_init on glib >=2.31.0
g_thread_init() is deprecated in glib 2.31.0 and later. This will call
g_thread_ini() only when compiling against older versions of glib.

BUG=1971,chromium:253566
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40019004

Cr-Commit-Position: refs/heads/master@{#8400}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8400 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 21:23:13 +00:00
braveyao@webrtc.org
8820ac7cc4 peerconnectin_server: missing comma in sprintfn() in r8128
BUG=4244
TEST=Manual Test
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37079004

Cr-Commit-Position: refs/heads/master@{#8213}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8213 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 09:58:45 +00:00
braveyao@webrtc.org
a742cb1f37 Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off.
BUG=3872
TEST=Manual Test
R=jiayl@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36989004

Cr-Commit-Position: refs/heads/master@{#8193}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8193 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 04:23:39 +00:00
jlmiller@webrtc.org
b40c7bb53c Change sprintf use in talk samples to snprintf
BUG=2301
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8128 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 18:49:06 +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
tommi@webrtc.org
2c13f659c7 Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7763 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-28 10:37:31 +00:00
perkj@webrtc.org
c2dd5ee2c0 Prepare for removal of PeerConnectionObserver::OnError.
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 11:31:29 +00:00
thakis@chromium.org
44010f3e52 win: Replace custom assert() macro with regular assert.h
The current code got added in libjingle r103; I don't see a good reason for it.
Things still build with plain old assert.h.

The custom assert was wrong: __debugbreak() is documented to return void,
so doing `cond ? true : __debugbreak()` shouldn't build (and it doesn't in
clang-cl). It's possible to make it build by writing
`cond ? true : (__debugbreak(), true)`, but just using the regular header
seems like a much better fix.

BUG=chromium:82385
Review URL: https://webrtc-codereview.appspot.com/19139004/


git-svn-id: http://webrtc.googlecode.com/svn/trunk@7007 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 03:00:15 +00:00
buildbot@webrtc.org
a09a99950e (Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 17:26:08 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
tommi@webrtc.org
2adc51c86e Handle the case if an unusually long peer name is provided in the peerconnection example.
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6687 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 08:56:07 +00:00
kjellander@webrtc.org
0402515d35 Implement command line flags for peerconnection client example on Windows
Adding the flags and functionality for 'autoconnect', 'autocall', 'server',
'port', and 'help' like in the linux example.

BUG=3459
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13609004

Patch from Vicken Simonian <vsimon@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 16:28:13 +00:00
mallinath@webrtc.org
a0d3067575 Use CreatePeerConnection method which accepts port_allocator.
Other method will be removed, in a different CL.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20369006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-26 00:00:15 +00:00
henrike@webrtc.org
b8395ebe14 (Auto)update libjingle 62293974-> 62364298
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-28 21:57:22 +00:00
jiayl@webrtc.org
14d80793a8 PeerConnectionClient needs to initialize SSL.
BUG=2911
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5531 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 00:41:59 +00:00
sergeyu@chromium.org
a23f0ca4ba Update talk to 56619788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3839005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 22:48:52 +00:00
wu@webrtc.org
97077a3ab2 Update libjingle to 55618622.
Update libyuv to r826.

TEST=try bots
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-25 21:18:33 +00:00
henrike@webrtc.org
28654cbc22 Update talk folder to revision=49713299.
TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1848004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-22 21:07:49 +00:00
henrike@webrtc.org
28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00