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Author SHA1 Message Date
pbos@webrtc.org
161f808500 Add test for VideoEncoder setup/teardown. 
Verifies that InitEncode and RegisterEncodeCompleteCallback gets
called before Encode is called. Also verifies that teardown is correctly
done during DestroyVideoSendStream().

BUG=2339
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 14:22:35 +00:00
pbos@webrtc.org
2bb1bdab8d Preserve RTP states for restarted VideoSendStreams. 
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.

Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.

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

Review URL: https://webrtc-codereview.appspot.com/20819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 13:06:48 +00:00
stefan@webrtc.org
73823cafa4 Add initial gn build files for video_coding and video_processing. 
BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 11:46:43 +00:00
pbos@webrtc.org
03c817e405 Fix pacer to accept duplicate sequence numbers on different SSRCs. 
BUG=3550
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 10:20:35 +00:00
andresp@webrtc.org
b941fe8098 Fix data races related with traces in bitrate estimator test. 
BUG=3549
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 08:50:48 +00:00
pbos@webrtc.org
bd249bc711 Remove GetDefaultConfigs() from Call. 
Defaults for configs are instead placed in the Config constructors.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 04:45:15 +00:00
stefan@webrtc.org
7832648824 Add missing break introduced in r6603. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 17:04:00 +00:00
stefan@webrtc.org
bee164a214 Fix test issues and a win compile error introduced with r6605. 
Also changes the name of a variable which has been hijacked by windef.h (included by windows.h), which forces #define near and #define far upon us. This issue was introduced via the following inclusion chain:
bwe_test_framework_unittest.cc includes
  paced_sender.h
    tick_util.h
      windows.h
        windef.h

And causes EXPECT_NEAR(foo, bar, near); to expand to EXPECT_NEAR(foo, bar,); generating a very confusing compile error.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 17:00:06 +00:00
stefan@webrtc.org
875ad49dee Revert conversion from TickTime to int64_t in paced sender. 
Introduced with r6600, causing flakes in SuspendBelowMinBitrate. The reason for this flake is currently unknown.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 15:27:55 +00:00
pbos@webrtc.org
8faa5db992 Add pbos@webrtc.org as owner for webrtc/test/. 
BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 13:49:17 +00:00
stefan@webrtc.org
b9f5453e29 Add boilerplate code for H.264. 
R=mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 12:42:07 +00:00
tina.legrand@webrtc.org
d8440f7c45 Have Opus follow Chromium revisions 
Before this change, a pinned version of Opus was used in WebRTC. This could lead to WebRTC running a different version of Opus compared to the version used with the corresponding Chromium revision.
This CL pulls in the Opus version the Chromium uses.

BUG=3546
R=kjellander@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15979005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 12:13:08 +00:00
pbos@webrtc.org
20c1f56992 Configure RTX send status on new modules. 
Fixes bug where newly-allocated modules wouldn't send payload-based
padding (or probably not send over RTX at all).

As the newly-added test exposed lock-inversions shown on tsan in
VideoReceiver, VideoReceiver was thread-annotated and locks taken less.
BUG=chromium:391085
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 10:58:12 +00:00
stefan@webrtc.org
88e0dda475 Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 09:20:42 +00:00
mflodman@webrtc.org
614000d638 Adding pbos as video/ owner and removing persons never working with this folder. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 08:50:14 +00:00
kjellander@webrtc.org
c5e53dde71 Revert 6597 "Roll chromium_revision 280876:281094" 
Breaks GN on Linux with errors like this:
[133/485 | 4.471] CC obj/third_party/libjpeg_turbo/simd/simd.jsimd_x86_64.o
FAILED: g++ -MMD -MF obj/out/Debug/gen/library_loaders/libspeechd.libspeechd.o.d -DCHROMIUM_BUILD -DENABLE_ONE_CLICK_SIGNIN -DENABLE_NOTIFICATIONS -DENABLE_EGLIMAGE=1 -DENABLE_BACKGROUND=1 -DUSE_MOJO=1 -DV8_DEPRECATION_WARNINGS -DBLINK_SCALE_FILTERS_AT_RECORD_TIME -DCLD_VERSION=2 -DENABLE_MDNS=1 -DENABLE_PEPPER_CDMS -DENABLE_PLUGINS=1 -DENABLE_PRINTING=1 -DENABLE_FULL_PRINTING=1 -DENABLE_SPELLCHECK=1 -DUSE_UDEV -DTOOLKIT_VIEWS=1 -DUI_COMPOSITOR_IMAGE_TRANSPORT -DUSE_AURA=1 -DUSE_CAIRO=1 -DUSE_CLIPBOARD_AURAX11=1 -DUSE_DEFAULT_RENDER_THEME=1 -DUSE_GLIB=1 -DUSE_NSS=1 -DUSE_X11=1 -DUSE_XI2_MT=2 -DENABLE_WEBRTC=1 -DENABLE_EXTENSIONS=1 -DENABLE_CONFIGURATION_POLICY -DENABLE_TASK_MANAGER=1 -DENABLE_THEMES=1 -DENABLE_CAPTIVE_PORTAL_DETECTION=1 -DENABLE_SESSION_SERVICE=1 -DENABLE_APP_LIST=1 -DENABLE_SETTINGS_APP=1 -DENABLE_MANAGED_USERS=1 -DENABLE_SERVICE_DISCOVERY=1 -DENABLE_AUTOFILL_DIALOG=1 -DENABLE_REMOTING=1 -DENABLE_GOOGLE_NOW=1 -D_FILE_OFFSET_BITS=64 -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -D_DEBUG -DDYNAMIC_ANNOTATIONS_ENABLED=1 -DWTF_USE_DYNAMIC_ANNOTATIONS=1 -D_GLIBCXX_DEBUG=1 -I../.. -Igen -fno-strict-aliasing -fstack-protector --param=ssp-buffer-size=4 -m64 -funwind-tables -fPIC -pipe -pthread -Wall -Werror -Wsign-compare -Wendif-labels -Wno-missing-field-initializers -Wno-unused-parameter -fvisibility=hidden -O0 -g2 -fno-threadsafe-statics -fvisibility-inlines-hidden -fno-rtti -fno-exceptions -c gen/library_loaders/libspeechd.cc -o obj/out/Debug/gen/library_loaders/libspeechd.libspeechd.o
In file included from gen/library_loaders/libspeechd.cc:4:0:
../../out/Debug/gen/library_loaders/libspeechd.h:7:54: fatalerror: third_party/speech-dispatcher/libspeechd.h: No such file or directory
compilation terminated.
ninja: build stopped: subcommand failed.

> Roll chromium_revision 280876:281094
> 
> No significant DEPS changes in this roll, only some changes
> in how clang_format is downloaded.
> 
> BUG=
> TEST=Local testing as trybots currently cannot handle DEPS changes properly.
> R=niklas.enbom@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/20829004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 07:53:46 +00:00
kjellander@webrtc.org
cb1df98093 Roll chromium_revision 280876:281094 
No significant DEPS changes in this roll, only some changes
in how clang_format is downloaded.

BUG=
TEST=Local testing as trybots currently cannot handle DEPS changes properly.
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 07:31:19 +00:00
marpan@webrtc.org
720964faac Fix memcheck error in r6594. 
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 21:14:07 +00:00
kjellander@webrtc.org
11bea8977e GN: Implement BUILD.gn for common_video. 
This adds copying of Chromium's third_party/BUILD.gn
to acommondate libyuv's BUILD.gn that imports the 'jpeg'
config from that file.

BUG=3441
TEST=trybots + local compile passing with:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false build_libyuv=false" && ninja -C out/Default

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 17:04:12 +00:00
marpan@webrtc.org
c8364539d3 Fix for FEC decoding with sequence number wrap-around. 
BUG=3507
R=stefan@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 16:49:30 +00:00
bjornv@webrtc.org
69ef9911e4 delay_estimator: Allows dynamically used history sizes 
Gives the user a possibility to dynamically change the history size. The main advantage is, for example, that you now can start with a wide delay range and over time decrease the search window to lower complexity.

Adds
- two new APIs.
- and updates unit tests.
- a history_size member variable to BinaryDelayEstimator.
- two help function re-allocating buffer memory.

One thing that makes this a little complicated is that you are allowed to have multiple delay estimators with the same reference, so changing the buffer sizes at one place will automatically give you a mismatch at other places.

BUG=3532, 3504
TESTED=trybots and manually
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 14:59:03 +00:00
aluebs@webrtc.org
224a140339 Make experimental NS API not purely virtual 
Because not all subclasses will want to bother overriding these methods.

R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 13:41:39 +00:00
bjornv@webrtc.org
c0ba4392f1 common_audio: Removes macro WEBRTC_SPL_SHIFT_W16 
We should avoid macros in general (see style guide). This shift macro is not a severe one, since there is a check for negativity.

BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 13:38:53 +00:00
kwiberg@webrtc.org
38214d53db EchoCancellationImpl::ProcessRenderAudio: Use float samples directly 
This patch lets EchoCancellationImpl::ProcessRenderAudio ask the given
AudioBuffer for float sample data directly, instead of asking for
int16 samples and then converting manually.

Since EchoCancellationImpl::ProcessRenderAudio takes a const
AudioBuffer*, it was necessary to add some const accessors for float
data to AudioBuffer.

R=aluebs@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 09:47:33 +00:00
andresp@webrtc.org
a82f9a243d Add Tsan2 to .gitignore 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 08:23:58 +00:00
asapersson@webrtc.org
dfdaeb92d8 Removed old code and default implementations. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 07:35:21 +00:00
braveyao@webrtc.org
9c89e932c9 WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. 
(Including another fix here, https://review.webrtc.org/16779004/, to make the test run)

BUG=3500
TEST=Manual Test
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-03 05:59:22 +00:00
buildbot@webrtc.org
3ffa1f917e (Auto)update libjingle 70422491-> 70424781 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6586 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 19:51:26 +00:00
kjellander@webrtc.org
b25b08b302 Remove tools/resources 
The script to update the resources before we used the .sha1 files
was moved out in r4277 and later deleted in r5099.
This dir serves no purpose, so let's remove it.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 15:48:29 +00:00
jiayl@webrtc.org
93426cd2ff Implement BUILD.gn for desktop_capture. 
BUG=3441
R=brettw@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 15:47:12 +00:00
andresp@webrtc.org
33586c83b1 Make deadlock suppressions less generic. 
Previously they were enabled on all webrtc and talk primitives directly when TSAN config changed to enable deadlock detections.

BUG=3509
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 14:19:05 +00:00
andresp@webrtc.org
1295dc6a23 Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators. 
I think this does not introduces any contention or new deadlocks. But that is hard to verify at the moment.

BUG=chromium:388191
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 13:23:19 +00:00
buildbot@webrtc.org
0bb9fac98c (Auto)update libjingle 70343444-> 70394475 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 11:54:09 +00:00
marpan@webrtc.org
895698067c Roll chromium 280149:280876. 
Pick up the libvpx roll:
https://codereview.chromium.org/367733002/

R=andrew@webrtc.org
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-02 02:50:05 +00:00
buildbot@webrtc.org
d8a9069080 (Auto)update libjingle 70340027-> 70343444 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 19:26:43 +00:00
tkchin@webrtc.org
74bf7a6523 Add tkchin@ to OWNERS. 
Adding myself to OWNERS of subdirectories containing iOS bits.  Added niklas.enbom@ for audio_device and wu@ for everything else.

R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 18:38:28 +00:00
jiayl@webrtc.org
974bbbb352 Fix uninitialized value in DtlsTransport and TransportDescription. 
BUG=crbug/390304
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 18:33:07 +00:00
fbarchard@google.com
08564546cb roll libyuv to r1025 for mips n32 support, arm nacl port, psnr tool jpeg support. 
BUG=none
TESTED=untested
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 18:17:21 +00:00
buildbot@webrtc.org
6335645400 (Auto)update libjingle 70329914-> 70330023 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 16:46:01 +00:00
henrike@webrtc.org
37b4e1bbcb webrtc/base: add dependent setting for gtest include directory that was missed when creating base_tests.gyp. Same as https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle_tests.gyp?r=6484#39 
BUG=N/A
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 16:39:17 +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
stefan@webrtc.org
9138eb649b Fix compile error introduced with r6571. 
TBR=mflodman@webrtc.org

BUG=3527

Review URL: https://webrtc-codereview.appspot.com/20799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 12:44:05 +00:00
stefan@webrtc.org
5779ca478d Fixes a potential BWE clock mismatch bug. 
Since libjingle provides a packet arrival timestamp to webrtc, and the clock in remote bitrate estimator and the clock used for packet arrival timestamp can be different. This can cause the bandwidth estimator to malfunction.

This CL changes the remote bitrate estimator so that packet arrival timestamps never are compared to the time taken from the internal clock.

BUG=3527
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 12:14:49 +00:00
bjornv@webrtc.org
6d21ddca5f audio_processing/aec: Refactors NonLinearProcessing to prepare for NEON optimizations 
Puts functionality necessary to calculate sub-band coherences into a function.

BUG=3131
TESTED=trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-01 10:03:42 +00:00
henrike@webrtc.org
d5a0506e84 Use X509_NAME, not struct X509_name_st. 
Also include openssl/x509.h explicitly since we're using functions and types
from it.

BUG=none
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6569 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 20:38:56 +00:00
bjornv@webrtc.org
59adb1dcd7 Neon version of cftmdl_128() 
The performance gain on a Nexus 7 reported by audioproc is ~2.3

The output is bit exact.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19829004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 19:34:33 +00:00
aluebs@webrtc.org
9825afc3bd Add ExperimentalNs support in Config 
R=andrew@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 17:39:53 +00:00
pbos@webrtc.org
2be53a3b00 Disable CanSwitchToUseAllSsrcs on DrMemory. 
Test times out.

R=mflodman@webrtc.org
BUG=3159

Review URL: https://webrtc-codereview.appspot.com/13799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 15:53:07 +00:00
pbos@webrtc.org
be9d2a4549 Reserve RTP/RTCP modules in SetSSRC. 
Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.

Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.

To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.

BUG=3078
R=kjellander@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 13:19:09 +00:00
bjornv@webrtc.org
cd9b90ab53 Neon version of cft1st_128() 
The performance gain on a Nexus 7 reported by audioproc is ~2%

See comments regarding the output.

R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21679004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 12:05:18 +00:00