13 Commits

Author SHA1 Message Date
perkj
c9022f5086 Delete empty API files and cleaned up includes.
TBR=glaznev@webrtc.org

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1809053002

Cr-Commit-Position: refs/heads/master@{#12039}
2016-03-17 16:57:30 +00:00
Stefan Holmer
55d6e7ca5f Disable tests due to issue 5659.
TBR=kjellander@webrtc.org
BUG=webrtc:5659

Review URL: https://codereview.webrtc.org/1809103002 .

Cr-Commit-Position: refs/heads/master@{#12035}
2016-03-17 15:26:54 +00:00
kwiberg
2bbff996a6 Helpers in peer connection unit tests: Use scoped_ptr instead of raw pointers
A handful of helpers were using SessionDescriptionInterface** output
arguments to return ownership. Chenge them to either use a
rtc::scoped_ptr<SessionDescriptionInterface>* output parameter, or to
simply return a rtc::scoped_ptr<SessionDescriptionInterface>. Not
using raw pointers for things you own is good in general; it will also
be very convenient when scoped_ptr is gone, since unique_ptr doesn't
have .accept() or .use() methods.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1798173002

Cr-Commit-Position: refs/heads/master@{#12021}
2016-03-16 18:03:08 +00:00
torbjorng
43166b8adf Add IsAcceptableCipher, use instead of GetDefaultCipher.
The old code insists on exact cipher suite matches with hardwired expectations. It does this matching parameterized with key type (RSA vs ECDSA) and TLS version (DTLS vs TLS and version 1.0 vs 1.2).

This CL changes things to check against a white-list of cipher suites, with the check parameterized with key type (again RSA vs ECDSA). Then separately checks TLS version since the old implicit check of TLS version by means of resulting cipher suite was too blunt.

Using a white list for cipher suites isn't perfect, but it is safe and requires minimal maintenance. It allows compatibility with not just one exact version of underlying crypto lib, but any version with reasonable defaults.

The CL also re-enables critical tests which had to be disabled recently to allow a boringssl roll.

BUG=webrtc:5634

Review URL: https://codereview.webrtc.org/1774583002

Cr-Commit-Position: refs/heads/master@{#11951}
2016-03-11 08:06:55 +00:00
hta
aac2dea765 Changed defaults for CreateAnswer in non-constraint mode
This CL also adds control flag in webrtcsession_unittests
that says whether to prefer constraints APIs or non-constraints APIs, and uses it in the test that was needed
to uncover the bug.

BUG=webrtc:4906

Review URL: https://codereview.webrtc.org/1775033002

Cr-Commit-Position: refs/heads/master@{#11947}
2016-03-10 21:36:02 +00:00
hta
6b4f839c53 Adds a test for an one-way media PeerConnection.
This involves changing a few verification functions for frames
received so that they always accept the result if there's no stream.

BUG=

Review URL: https://codereview.webrtc.org/1772353002

Cr-Commit-Position: refs/heads/master@{#11937}
2016-03-10 08:24:37 +00:00
perkj
a3ede6c510 Renamed VideoSourceInterface to VideoTrackSourceInterface.
Moved VideoSourceInterface to MediaStreamInterface.h
Renamed VideoSourceTest to VideoCapturerTrackSourceTest
Renamed VideoSource to VideoCaptureTrackSource and cl lint and cl format.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1770003002 .

Cr-Commit-Position: refs/heads/master@{#11893}
2016-03-08 00:28:03 +00:00
kjellander
43942d1f1e Roll chromium_revision 508edd3..35d57a0 (379249:379535)
Change log: 508edd3..35d57a0
Full diff: 508edd3..35d57a0

Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/708db16..58218b6
DEPS diff: 508edd3..35d57a0/DEPS

No update to Clang.

TBR=torbjorng@webrtc.org
BUG=webrtc:5634
NOTRY=True

Review URL: https://codereview.webrtc.org/1773543002

Cr-Commit-Position: refs/heads/master@{#11890}
2016-03-07 21:59:15 +00:00
kjellander
f475277547 Rename constants files in webrtc/{media,p2p}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.

To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}

This CL will require coordinating landing a roll in Chromium.

BUG=webrtc:4256
NOTRY=True

Review URL: https://codereview.webrtc.org/1750593002

Cr-Commit-Position: refs/heads/master@{#11842}
2016-03-02 13:42:35 +00:00
kjellander@webrtc.org
9b8df25c73 Move talk/session/media -> webrtc/pc
The libjingle_p2p target is renamed to rtc_pc.
The libjingle_p2p_unittest test will be renamed in a
separate follow-up CL, to make it possible to run all
trybots successfully for this CL.

BUG=webrtc:5419
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1691463002 .

Cr-Commit-Position: refs/heads/master@{#11592}
2016-02-12 05:48:10 +00:00
kjellander@webrtc.org
5ad129741c Rename webrtc/media/webrtc -> webrtc/media/engine
BUG=webrtc:5420
NOTRY=True
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1684163002 .

Cr-Commit-Position: refs/heads/master@{#11591}
2016-02-12 05:39:50 +00:00
kjellander
b24317bfda Fix license headers in webrtc/api.
In addition to the code moved from talk/app/webrtc
there were some files in webrtc/api/objctests that still
had the libjingle license header.

BUG=webrtc:5418
TBR=tkchin@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1680293005

Cr-Commit-Position: refs/heads/master@{#11552}
2016-02-10 15:54:53 +00:00
Henrik Kjellander
15583c19d7 Move talk/app/webrtc to webrtc/api
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc

The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.

I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002

BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1610243002 .

Cr-Commit-Position: refs/heads/master@{#11545}
2016-02-10 09:53:26 +00:00