Commit Graph

30 Commits

Author SHA1 Message Date
pthatcher@webrtc.org
3ee4fe5a94 Re-land: Add API to get negotiated SSL ciphers
This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.

The previously approved CL https://webrtc-codereview.appspot.com/26009004/ was reverted in https://webrtc-codereview.appspot.com/40689004/ due to compilation issues while rolling into Chromium.
As the new method has landed in Chromium in https://crrev.com/bc321c76ace6e1d5a03440e554ccb207159802ec, this should be safe to land here now.

BUG=3976
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37209004

Cr-Commit-Position: refs/heads/master@{#8343}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8343 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 22:35:30 +00:00
tommi@webrtc.org
2bf0e90c9d Revert 8275 "This CL adds an API to the SSL stream adapters and ..."
I'm reverting the patch due to compilation issues. It would be great if we could make sure Chromium is ready for the patch before we land it in WebRTC.
As is, we're trying to roll webrtc into Chromium and we can't (this is not the only reason though).  I might reland this after the roll, depending on how that goes though.
Here's an example failure:

e:\b\build\slave\win_gn\build\src\jingle\glue\channel_socket_adapter_unittest.cc(77) : error C2259: 'jingle_glue::MockTransportChannel' : cannot instantiate abstract class
        due to following members:
        'bool cricket::TransportChannel::GetSslCipher(std::string *)' : is abstract
        e:\b\build\slave\win_gn\build\src\third_party\webrtc\p2p\base\transportchannel.h(107) : see declaration of 'cricket::TransportChannel::GetSslCipher'
ninja: build stopped: subcommand failed.

> This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
> 
> BUG=3976
> R=davidben@chromium.org, juberti@webrtc.org, pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26009004

TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40689004

Cr-Commit-Position: refs/heads/master@{#8282}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8282 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-07 11:13:18 +00:00
pthatcher@webrtc.org
1d11c8202b This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
BUG=3976
R=davidben@chromium.org, juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26009004

Cr-Commit-Position: refs/heads/master@{#8275}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8275 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 19:47:39 +00:00
tnakamura@webrtc.org
db1ebf6c0c Add jakehilton@gmail.com to AUTHORS
BUG=3918
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34889004

Cr-Commit-Position: refs/heads/master@{#8172}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8172 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 19:15:48 +00:00
pthatcher@webrtc.org
0ba1533fdb Added support for an Origin header in STUN messages.
For WebRTC there are instances where it may be desirable to provide
information to the STUN/TURN server about the website that initiated
a peer connection. This modification allows an origin string to be
included in the MediaConstraints object provided by the browser, which
is then passed as a STUN header in communications with the server.
A separate change will be submitted to the Chromium project that
uses and is dependent on this change, implementing IETF draft
http://tools.ietf.org/html/draft-johnston-tram-stun-origin-02

Originally a patch from skobalt@gmail.com.

(https://webrtc-codereview.appspot.com/12839005/edit)

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8035 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-10 00:47:02 +00:00
tkchin@webrtc.org
ee9d61ce45 This fixes a small memory leak (found using Xcode/Instruments on iOS) in
the ObjC bindings of PeerConnection. The generated session description has
to be released by the recipient

BUG=3985
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28959004

Patch from Matthias Liebig <matthias.gcode@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 22:01:53 +00:00
tkchin@webrtc.org
c569a49a3d Unit tests for SSLAdapter
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17309004

Patch from Manish Jethani <manish.jethani@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7269 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 05:56:44 +00:00
henrike@webrtc.org
31c285b333 Update AUTHORS file.
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:12:59 +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
andrew@webrtc.org
d798095a37 replace inline assembly WebRtcNsx_PrepareSpectrumNeon by intrinsics.
The modification only uses the unique part of the spectrum (as is done for the C and asm code). It passes
byte to byte conformance test, and the single function performance
(if not specified, the code is compiled by GCC 4.6) on different
platforms:

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| C                          |      100% |      100% |       100% |
| Neon asm                   |       18% |       14% |        19% |
| Neon inline asm            |       31% |       25% |        27% |
| Neon intrinsic (GCC 4.6)   |       33% |       27% |        42% |
| Neon intrinscis (GCC 4.8)  |       17% |       14% |        19% |
| Neon intrinsics (LLVM 3.3) |       15% |       13% |        18% |

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13739004

Patch from Joe Yu <joe.yu@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6920 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 17:46:45 +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
fischman@webrtc.org
7c82adae61 AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial.
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16379004

Patch from Bridger Maxwell <bridgeyman@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6028 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 00:17:47 +00:00
henrike@webrtc.org
ceffdbc371 Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord()
R=henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6018 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:11:07 +00:00
fischman@webrtc.org
82387e4608 Add ability to receive calls for iOS
BUG=2701
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7989005

Patch from Sajid Hussain <shussain@temasys.com.sg>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 18:47:11 +00:00
andrew@webrtc.org
a9bdee6690 Add Christophe Dumez to AUTHORS.
Copied from Chromium's AUTHORS.

R=ch.dumez@samsung.com

Review URL: https://webrtc-codereview.appspot.com/5559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5341 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-06 19:43:21 +00:00
fischman@webrtc.org
af320fd2f7 The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation.
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6019004

Patch from Rafael Lopez Diez <rafalopezdiez@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5309 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 21:33:27 +00:00
fischman@webrtc.org
eb7def234e Fix compilation errors on Fedora 20.
peerconnection_jni.cc: syscall() comes from <unistd.h>
RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it
rtp_payload_registry_unittest.cc: avoid narrowing int to uint32.

BUG=2700
R=andrew@webrtc.org, fischman@webrtc.org, henrik.lundin@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5019004

Patch from Victor Costan <costan@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 21:34:30 +00:00
henrike@webrtc.org
7b2f955e56 Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README.
BUG=1935
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1805005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 18:04:56 +00:00
mflodman@webrtc.org
efdf778d3f BUG=1351
Propose of this CL: Close the camera properly on MacOS in order to allow other apps to use it.

Changes in this CL:
1. video_capture_qtkit_info_objc.mm _captureDevicesInfo is never released. I have found this memory leak using Instruments from XCode. The patch is releasing it in dealloc.

2. In video_capture_qtkit_objc.h:
a) _captureDeviceName is not needed. Is allocated in the class but never used.
b) I don't see the role of the  NSAutoreleasePool. also if you use it you have to release it when the class is destroyed. Otherwise you will leak memory. Libjingle has for each thread a pool on mac os.

3. In video_capture_qtkit_objc.mm
a) the camera is not stopped properly . See the changes from dealloc. NOTE : If you don't call [[_captureVideoDeviceInput device] close] other apps will not be able to use the camera since you are not closing your app

b) Removed QTCaptureDevice* videoDevice = (QTCaptureDevice*)[_captureDevices objectAtIndex:0]; I don't know why this because the desired camera is opened in setCaptureDeviceById and can be different than position 0 in the camera array. At this moment if you have two cameras and user want to pick the one on index 1 the app also locks the one on 0 .

Other changes I have done to improve (and are not in this CL):
a) I have set the FPS properly to the desired. I have succeeded to reduce the CPU with 3 % doing this. The current code for setting FPS is commented in webrtc
b) I have removed _rLock from the equation. I don't know if it's good or not but I hadn't understood what exactly we are trying to protect with this. Anyway in the current implementation is never released.

Review URL: https://webrtc-codereview.appspot.com/1097014

Patch from Silviu Caragea <silviu.cpp@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3886 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-23 11:57:56 +00:00
phoglund@webrtc.org
17b867ae00 compile fix for get_nprocs() with uClibc
BUG=

Review URL: https://webrtc-codereview.appspot.com/1150006
Patch from Mostyn Bramley-Moore <mostynb@opera.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3598 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-04 15:09:03 +00:00
andrew@webrtc.org
5140e24037 MIPS optimizations for Signal Processing Library patch01
Review URL: https://webrtc-codereview.appspot.com/1028004
Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3557 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-21 20:12:21 +00:00
andrew@webrtc.org
73a702c979 This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware.
Review URL: https://webrtc-codereview.appspot.com/1061007
Patch from Gil Osher <gil.osher@vonage.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3437 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-30 21:18:31 +00:00
andrew@webrtc.org
bcb717428f .gitignore: Add *.mk, created as part of ChromiumOS build
Contributed by Josh Triplett <josh.triplett@intel.com>

BUG=None
TEST=Build Chromium and ChromiumOS from source, and run "repo status",
     with and without this change.

Review URL: https://webrtc-codereview.appspot.com/1000006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3334 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-04 21:25:42 +00:00
pwestin@webrtc.org
1b6da28047 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests.
Landing of 573005 On behalf of an1kumar@gmail.com

TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1002008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3322 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-21 17:46:24 +00:00
niklas.enbom@webrtc.org
ef629299e9 Landing http://review.webrtc.org/914006/
Review URL: https://webrtc-codereview.appspot.com/930007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3030 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-31 14:35:11 +00:00
turaj@webrtc.org
01ad75888a ilbc: Mark untouched input arrays as const
Review URL: https://webrtc-codereview.appspot.com/662004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2490 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-03 21:35:46 +00:00
andrew@webrtc.org
e22beabaf1 [MIPS] Adding support for MIPS architecture for WebRTC.
Small change to typedefs.h to enable MIPS Little Endian port.

TBR=niklas.enbom@webrtc.org
BUG=https://code.google.com/p/chromium/issues/detail?id=130022
TEST=make chrome

Review URL: https://webrtc-codereview.appspot.com/679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2451 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-27 22:24:43 +00:00
niklas.enbom@webrtc.org
87c50f02c6 Adding author
git-svn-id: http://webrtc.googlecode.com/svn/trunk@870 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 12:19:22 +00:00
niklas.enbom@webrtc.org
3a9680b01b Adding author
git-svn-id: http://webrtc.googlecode.com/svn/trunk@869 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 09:48:06 +00:00
niklase@google.com
da159d6be6 git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:51:34 +00:00