external/webrtc.git
24 min agoUpdate all .isolate files for the new format. master
kjellander@webrtc.org [Fri, 31 Oct 2014 18:08:09 +0000 (18:08 +0000)]
Update all .isolate files for the new format.

R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27809004

Patch from Marc-Antoine Ruel <maruel@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d

79 min agoUpdate Android projects to API level 20.
kjellander@webrtc.org [Fri, 31 Oct 2014 17:13:37 +0000 (17:13 +0000)]
Update Android projects to API level 20.

This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 19.

Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-20 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-20 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-20 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.

BUG=
R=glaznev@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7582 4adac7df-926f-26a2-2b94-8c16560cd09d

94 min agoFix N7 camera aspect ratio.
glaznev@webrtc.org [Fri, 31 Oct 2014 16:58:58 +0000 (16:58 +0000)]
Fix N7 camera aspect ratio.

N7 video preview generates stretched output:
https://code.google.com/p/android/issues/detail?id=70830.
To workaround the problem set camera picture size in
addition to video preview size with the same resolution.

BUG=3971
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7581 4adac7df-926f-26a2-2b94-8c16560cd09d

2 hours agoBuild fix for MIPS32R6.
andrew@webrtc.org [Fri, 31 Oct 2014 16:26:17 +0000 (16:26 +0000)]
Build fix for MIPS32R6.

Exclude MIPS optimizations for MIPS32R6 build since some of the instructions
are not supported. This is temporary fix, until the MIPS32R6 code is added.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25989004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7580 4adac7df-926f-26a2-2b94-8c16560cd09d

2 hours agoFix a name collision with Android libc++
andrew@webrtc.org [Fri, 31 Oct 2014 16:01:25 +0000 (16:01 +0000)]
Fix a name collision with Android libc++

The Android libc++ has a symbol called '_P'
This CL renames a property called _P in webrtc.

BUG=chromium:427718
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30009004

Patch from Fabrice de Gans-Riberi <fdegans@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7579 4adac7df-926f-26a2-2b94-8c16560cd09d

5 hours agoImplement conference-mode temporal-layer screencast.
pbos@webrtc.org [Fri, 31 Oct 2014 13:08:10 +0000 (13:08 +0000)]
Implement conference-mode temporal-layer screencast.

Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788,1667

Review URL: https://webrtc-codereview.appspot.com/23269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d

5 hours agoConfigure A/V sync in WebRtcVideoEngine2.
pbos@webrtc.org [Fri, 31 Oct 2014 12:59:34 +0000 (12:59 +0000)]
Configure A/V sync in WebRtcVideoEngine2.

Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/23249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d

7 hours agoSimplify bwe tests.
stefan@webrtc.org [Fri, 31 Oct 2014 10:47:12 +0000 (10:47 +0000)]
Simplify bwe tests.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7576 4adac7df-926f-26a2-2b94-8c16560cd09d

12 hours agoAdapting bitrate according to maxplaybackrate for Opus.
minyue@webrtc.org [Fri, 31 Oct 2014 05:33:10 +0000 (05:33 +0000)]
Adapting bitrate according to maxplaybackrate for Opus.

BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d

13 hours agoRevert "Revert part of r7561, "Refactor audio conversion functions.""
andrew@webrtc.org [Fri, 31 Oct 2014 04:58:14 +0000 (04:58 +0000)]
Revert "Revert part of r7561, "Refactor audio conversion functions.""

This restores the conversion changes to AudioProcessing originally
added in r7561, with minor alterations to ensure it passes all tests.

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/28899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7574 4adac7df-926f-26a2-2b94-8c16560cd09d

18 hours agoarm64 iOS build.
tkchin@webrtc.org [Fri, 31 Oct 2014 00:14:39 +0000 (00:14 +0000)]
arm64 iOS build.

Allows successful build of arm64 libraries using
GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64".
Note that not all libraries will be NEON optimized (eg common_audio),
however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be
defined so that libvpx doesn't post-process, which is significantly
detrimental to performance.

BUG=3898
R=kjellander@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d

18 hours agoImprove the logging when a TCP connection is deleted.
jiayl@webrtc.org [Thu, 30 Oct 2014 23:50:54 +0000 (23:50 +0000)]
Improve the logging when a TCP connection is deleted.

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7572 4adac7df-926f-26a2-2b94-8c16560cd09d

23 hours agoAdd 15 fps support for Android devices with missing 15 fps
glaznev@webrtc.org [Thu, 30 Oct 2014 18:38:26 +0000 (18:38 +0000)]
Add 15 fps support for Android devices with missing 15 fps
camera mode.

Some latest Android devices support only 30 fps for front camera,
but HW VP8 encoder performance is not enough for 720p 30 fps
encoding. Add 15 fps support for these devices by allowing
frame drop in Android camera wrapper.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7571 4adac7df-926f-26a2-2b94-8c16560cd09d

29 hours agoCreating a C++ wrapper class for VAD
henrik.lundin@webrtc.org [Thu, 30 Oct 2014 13:23:25 +0000 (13:23 +0000)]
Creating a C++ wrapper class for VAD

Also adding a mock. This work is part of an ongoing effort to
encapsulate encoders in AudioEncoder classes. The CNG encoder will also
be implemented as an AudioEncoder class, and will also contain a VAD
C++ wrapper.

BUG=3926
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7570 4adac7df-926f-26a2-2b94-8c16560cd09d

31 hours agoRevert part of r7561, "Refactor audio conversion functions."
kwiberg@webrtc.org [Thu, 30 Oct 2014 11:16:06 +0000 (11:16 +0000)]
Revert part of r7561, "Refactor audio conversion functions."

Specifically, revert this part:

  "Remove hacks in AudioBuffer intended to maintain bit-exactness with
   the float path. The conversions etc. are now all natural, and
   instead we enforce close but not bit-exact output between the two
   paths."

But keep the conversion function rename, since that doesn't seem to be
causing problems.

R=tina.legrand@webrtc.org, bjornv@webrtc.org
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7569 4adac7df-926f-26a2-2b94-8c16560cd09d

34 hours agoCleaning up r7562-7567.
minyue@webrtc.org [Thu, 30 Oct 2014 08:23:54 +0000 (08:23 +0000)]
Cleaning up r7562-7567.

Wrongly used git svn dcommit for committing a CL.

Then two reverts were applied.

Still something needs to be cleaned.

BUG=

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7568 4adac7df-926f-26a2-2b94-8c16560cd09d

34 hours ago(Auto)update libjingle 78822708-> 78823675
buildbot@webrtc.org [Thu, 30 Oct 2014 07:50:13 +0000 (07:50 +0000)]
(Auto)update libjingle 78822708-> 78823675

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7567 4adac7df-926f-26a2-2b94-8c16560cd09d

34 hours agoRevert 7563 "before rebase" due to wrong submission
minyue@webrtc.org [Thu, 30 Oct 2014 07:49:58 +0000 (07:49 +0000)]
Revert 7563 "before rebase" due to wrong submission

> before rebase

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7566 4adac7df-926f-26a2-2b94-8c16560cd09d

34 hours agoRevert 7564 "to submit" due to wrong submission
minyue@webrtc.org [Thu, 30 Oct 2014 07:46:47 +0000 (07:46 +0000)]
Revert 7564 "to submit" due to wrong submission

> to submit

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7565 4adac7df-926f-26a2-2b94-8c16560cd09d

35 hours agoto submit
minyue@webrtc.org [Thu, 30 Oct 2014 07:20:09 +0000 (07:20 +0000)]
to submit

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7564 4adac7df-926f-26a2-2b94-8c16560cd09d

35 hours agobefore rebase
minyue@webrtc.org [Thu, 30 Oct 2014 07:19:57 +0000 (07:19 +0000)]
before rebase

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7563 4adac7df-926f-26a2-2b94-8c16560cd09d

35 hours agoadding default rates
minyue@webrtc.org [Thu, 30 Oct 2014 07:19:49 +0000 (07:19 +0000)]
adding default rates

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7562 4adac7df-926f-26a2-2b94-8c16560cd09d

38 hours agoRefactor audio conversion functions.
andrew@webrtc.org [Thu, 30 Oct 2014 03:40:10 +0000 (03:40 +0000)]
Refactor audio conversion functions.

Use a consistent naming scheme that can be understood at the callsite
without having to refer to documentation.

Remove hacks in AudioBuffer intended to maintain bit-exactness with the
float path. The conversions etc. are now all natural, and instead we
enforce close but not bit-exact output between the two paths.

Output of ApmTest.Process:
https://paste.googleplex.com/5931055831842816

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

Review URL: https://webrtc-codereview.appspot.com/13049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7561 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoUse external VideoDecoders in VideoReceiveStream.
pbos@webrtc.org [Wed, 29 Oct 2014 15:28:39 +0000 (15:28 +0000)]
Use external VideoDecoders in VideoReceiveStream.

Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.

Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.

Additionally addresses a data race in VideoReceiver that was exposed with this change.

R=mflodman@webrtc.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667

Review URL: https://webrtc-codereview.appspot.com/27829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoAdd stats for duplicate sent and received NACK requests.
asapersson@webrtc.org [Wed, 29 Oct 2014 12:42:30 +0000 (12:42 +0000)]
Add stats for duplicate sent and received NACK requests.

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

Review URL: https://webrtc-codereview.appspot.com/27799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7559 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agocommon_audio: Removed macro WEBRTC_SPL_RSHIFT_W32
bjornv@webrtc.org [Wed, 29 Oct 2014 10:29:16 +0000 (10:29 +0000)]
common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32

Replaces the trivial macro WEBRTC_SPL_RSHIFT_W32 with >> at various places in common_audio and removes it.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7558 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRemove unused code in overuse detector.
asapersson@webrtc.org [Wed, 29 Oct 2014 10:05:21 +0000 (10:05 +0000)]
Remove unused code in overuse detector.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7557 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoAudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket
kwiberg@webrtc.org [Wed, 29 Oct 2014 08:38:50 +0000 (08:38 +0000)]
AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket

Rename this accessor function to reflect its new, slightly changed
meaning. The reason for the change is that some codecs (iSAC) vary the
number of 10 ms frames from packet to packet, and so can't return a
truly constant value.

BUG=3926
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoEnable G.722 for Chromium builds
henrik.lundin@webrtc.org [Wed, 29 Oct 2014 08:32:44 +0000 (08:32 +0000)]
Enable G.722 for Chromium builds

BUG=3909
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7555 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days ago(Auto)update libjingle 78738075-> 78738103
buildbot@webrtc.org [Wed, 29 Oct 2014 08:14:14 +0000 (08:14 +0000)]
(Auto)update libjingle 78738075-> 78738103

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7554 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoApprtDemo Android: Switch between front and back camera.
perkj@webrtc.org [Wed, 29 Oct 2014 08:10:03 +0000 (08:10 +0000)]
ApprtDemo Android: Switch between front and back camera.
This adds a UI icon for switching between the front and back camera.
This cl adds the possibility to change between the front and back camera while in a call
or before the other end have connected.

BUG=3786
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7553 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoMake an AudioEncoder subclass for Opus
kwiberg@webrtc.org [Wed, 29 Oct 2014 07:28:36 +0000 (07:28 +0000)]
Make an AudioEncoder subclass for Opus

BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoRenaming bandwidth to bitrate in webrtcvoiceengine.
minyue@webrtc.org [Wed, 29 Oct 2014 02:27:08 +0000 (02:27 +0000)]
Renaming bandwidth to bitrate in webrtcvoiceengine.

"bandwidth" is usually a misleading mentioning. It can mean network throughput, audio frequency contents, etc.

This is to remove the confusion inside webrtcvoiceengine

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

Review URL: https://webrtc-codereview.appspot.com/28799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7551 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoMake NSinst_t* const and rename to self in ns_core
aluebs@webrtc.org [Tue, 28 Oct 2014 22:52:09 +0000 (22:52 +0000)]
Make NSinst_t* const and rename to self in ns_core

This is only to make the code more readable and maintainable.
It generates a bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7550 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agomove xmpp and p2p to webrtc
henrike@webrtc.org [Tue, 28 Oct 2014 22:20:11 +0000 (22:20 +0000)]
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

2 days agoMake local functions static and dropWebRtcNs_ in ns_core
aluebs@webrtc.org [Tue, 28 Oct 2014 21:06:57 +0000 (21:06 +0000)]
Make local functions static and dropWebRtcNs_ in ns_core

This is only to make the code more readable and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7548 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoMake all comments whole sentences in ns_core
aluebs@webrtc.org [Tue, 28 Oct 2014 20:56:53 +0000 (20:56 +0000)]
Make all comments whole sentences in ns_core

This is done to make the code more readable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7547 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoscoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chrom...
henrike@webrtc.org [Tue, 28 Oct 2014 18:06:42 +0000 (18:06 +0000)]
scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots.

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7546 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days ago(Auto)update libjingle 78642371-> 78680406
buildbot@webrtc.org [Tue, 28 Oct 2014 17:37:17 +0000 (17:37 +0000)]
(Auto)update libjingle 78642371-> 78680406

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7545 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoaudio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>"
bjornv@webrtc.org [Tue, 28 Oct 2014 13:05:43 +0000 (13:05 +0000)]
audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>"

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7544 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agocommon_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16
bjornv@webrtc.org [Tue, 28 Oct 2014 13:03:10 +0000 (13:03 +0000)]
common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16

The macro made a cast to uint16_t before a plain multiplication. At the few places where it was used the variables were already uint16_t.

Affected components:
* isac/fix

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7543 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoUse neteq_unittest_tools in audio_decoder_unittests
henrik.lundin@webrtc.org [Tue, 28 Oct 2014 09:47:13 +0000 (09:47 +0000)]
Use neteq_unittest_tools in audio_decoder_unittests

With the recent move of RtpFileReader to the rtp_test_utils target
(in r7536), it is now possible to let audio_decoder_unittests depend
on neteq_unittest_tools without breaking the Android build.

BUG=2692
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7542 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoFix double backslashes in incoming_video_stream.cc
perkj@webrtc.org [Tue, 28 Oct 2014 08:47:16 +0000 (08:47 +0000)]
Fix double backslashes in incoming_video_stream.cc

Originally uploaded in https://codereview.appspot.com/149160043/.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7541 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days ago(Auto)update libjingle 78616359-> 78642371
buildbot@webrtc.org [Tue, 28 Oct 2014 05:35:35 +0000 (05:35 +0000)]
(Auto)update libjingle 78616359-> 78642371

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7540 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoCheck if a datachannel in the current local description is an sctp channel before...
tommi@webrtc.org [Mon, 27 Oct 2014 22:15:04 +0000 (22:15 +0000)]
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

4 days agoAdd a simple AudioConverter class.
andrew@webrtc.org [Mon, 27 Oct 2014 18:18:17 +0000 (18:18 +0000)]
Add a simple AudioConverter class.

This will be used to refactor AudioProcessing/AudioBuffer. We can
enable alternate downmixing schemes in AudioProcessing by pulling
the conversion logic out of AudioBuffer.

The unit test is largely stolen from voice_engine/utility_unittest.cc.
As commented, the voice_engine routines should be replaced with
AudioConverter.

BUG=chromium:405270
R=aluebs@webrtc.org, mgraczyk@chromium.org
TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/30779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7538 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoOnly configure the SSL library in one place.
henrike@webrtc.org [Mon, 27 Oct 2014 18:13:40 +0000 (18:13 +0000)]
Only configure the SSL library in one place.

Build settings now use use_openssl in both Chromium and standalone builds. It
moves all the platform-specific SSL-related build checks to be conditioned on
this flag as appropriate.

This is to avoid colliding with Chromium's transition away from NSS.

This is a fixup of https://webrtc-codereview.appspot.com/29559004 to avoid
breaking use_legacy_ssl_defaults.

BUG=chromium:413497
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7537 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoMove (test) RtpFileReader to a lightweight target.
pbos@webrtc.org [Mon, 27 Oct 2014 18:01:03 +0000 (18:01 +0000)]
Move (test) RtpFileReader to a lightweight target.

Moves RtpFileReader to rtp_packet_parser and renames it to
rtp_test_utils which is allowed to rely on rtp_rtcp.

R=andrew@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7536 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoMove scoped_ptr "free" functions into the webrtc namespace.
andrew@webrtc.org [Mon, 27 Oct 2014 17:42:22 +0000 (17:42 +0000)]
Move scoped_ptr "free" functions into the webrtc namespace.

Resolves a conflict with Chromium's scoped_ptr on the recently added
make_scoped_ptr().

TEST=local Chromium Linux build passes.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7535 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAdding setting screen to AppRTCDemo.
glaznev@webrtc.org [Mon, 27 Oct 2014 17:22:15 +0000 (17:22 +0000)]
Adding setting screen to AppRTCDemo.

- Move server URL from connection screen
to the setting screen.
- Add setting for local video resolution.
- Auto save last entered room number.
- Use full screen mode in video renderer and fix
texture offsets recalculation when rendering type is
dynamically changed.

BUG=3935,3953
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7534 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days ago(Auto)update libjingle 78583324-> 78583691
buildbot@webrtc.org [Mon, 27 Oct 2014 16:20:42 +0000 (16:20 +0000)]
(Auto)update libjingle 78583324-> 78583691

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7532 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoUpgrade our scoped_ptr copy to match Chromium's latest.
andrew@webrtc.org [Mon, 27 Oct 2014 16:12:38 +0000 (16:12 +0000)]
Upgrade our scoped_ptr copy to match Chromium's latest.

In particular add the move constructor and assignment operator.

Diff between our version and Chromium's:
https://paste.googleplex.com/4887047529562112

R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7531 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoFix the SrtpFilter crash caused by two local offers.
pthatcher@webrtc.org [Mon, 27 Oct 2014 16:10:29 +0000 (16:10 +0000)]
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

4 days agoImplement screencast settings for WebRtcVideoEngine2.
pbos@webrtc.org [Mon, 27 Oct 2014 13:58:00 +0000 (13:58 +0000)]
Implement screencast settings for WebRtcVideoEngine2.

Adds support for screencast_min_bitrate and sets content type
corresponding to the capture type.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7529 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoCleaning up audio_decoder_test.cc and adding ResampleInputAudioFile
henrik.lundin@webrtc.org [Mon, 27 Oct 2014 12:58:18 +0000 (12:58 +0000)]
Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile

This CL contains some cleaning up and refactoring of
audio_decoder_test.cc. A new class ResampleInputAudioFile is created
and used in the tests.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7528 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoisacfix: Refactor big-endian reading and writing
kwiberg@webrtc.org [Mon, 27 Oct 2014 11:25:37 +0000 (11:25 +0000)]
isacfix: Refactor big-endian reading and writing

Make subroutines for encoding and decoding arrays of 16-bit big-endian
integers, and in the process fix a bug: When decoding an odd number of
bytes from be16, the least significant byte of the last int16 in the
array was properly taken to be zero instead of actually being read
(since it's outside the array). However, when encoding an odd number
of bytes, the least significant byte of the last int16 in the array
was written to the output as-is instead of being taken to be zero;
thus, we encoded one byte more than we should. This was probably not
harmful, and the value was dropped at decoding anyway; nevertheless,
writing a constant zero is the safe thing to do, and this patch does
so.

R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7527 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoIncrease max trace message size to 1024 characters.
pbos@webrtc.org [Mon, 27 Oct 2014 09:31:05 +0000 (09:31 +0000)]
Increase max trace message size to 1024 characters.

A recent CL by pbos:
https://code.google.com/p/webrtc/source/detail?r=7518

added long log messages and triggered errors on the DrMemory bot due to
WEBRTC_TRACE. The trace mechanism _should_ truncate the log strings
but something appears to be going awry.

This sweeps the problem under the rug, but given that WEBRTC_TRACE
should die fairly soon, seems to be a reasonable tradeoff.

TEST=passing try on DrMemory.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27849004

Patch from Andrew MacDonald <andrew@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7526 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoFix ::~LogMessage to print as a string.
pbos@webrtc.org [Mon, 27 Oct 2014 09:22:03 +0000 (09:22 +0000)]
Fix ::~LogMessage to print as a string.

R=andrew@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7525 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoUse flags set by the port allocator.
braveyao@webrtc.org [Mon, 27 Oct 2014 03:01:37 +0000 (03:01 +0000)]
Use flags set by the port allocator.

Currently, port allocator flags are ignored. This is inconvenient if you
want to have your own PortAllocatorFactory subclass.

BUG=webrtc:3958
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7524 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoPRESUBMIT: Add linux_msan to default trybots.
kjellander@webrtc.org [Fri, 24 Oct 2014 21:41:24 +0000 (21:41 +0000)]
PRESUBMIT: Add linux_msan to default trybots.

Will commit as soon it's online.

BUG=
R=pbos@webrtc.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7523 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 78430441-> 78445452
buildbot@webrtc.org [Fri, 24 Oct 2014 17:26:28 +0000 (17:26 +0000)]
(Auto)update libjingle 78430441-> 78445452

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7522 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 78427027-> 78430441
buildbot@webrtc.org [Fri, 24 Oct 2014 12:59:08 +0000 (12:59 +0000)]
(Auto)update libjingle 78427027-> 78430441

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7521 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoAdd HD support to Android if we detect a hardware video encoder that can be used...
perkj@webrtc.org [Fri, 24 Oct 2014 11:38:19 +0000 (11:38 +0000)]
Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected.

BUG=3934
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7520 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoAdding the subtool rtcBot report visualizer
houssainy@google.com [Fri, 24 Oct 2014 09:26:16 +0000 (09:26 +0000)]
Adding the subtool rtcBot report visualizer

This tool for visualize the output reports of rtcBot by calculating
the average and max of a specific stats and plot the output.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7519 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoMove min transmit bitrate to VideoEncoderConfig.
pbos@webrtc.org [Fri, 24 Oct 2014 09:23:21 +0000 (09:23 +0000)]
Move min transmit bitrate to VideoEncoderConfig.

min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agopatch from issue 25469004
pthatcher@webrtc.org [Thu, 23 Oct 2014 23:37:22 +0000 (23:37 +0000)]
patch from issue 25469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7517 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 78381351-> 78389679
buildbot@webrtc.org [Thu, 23 Oct 2014 23:07:23 +0000 (23:07 +0000)]
(Auto)update libjingle 78381351-> 78389679

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7516 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 78344087-> 78381351
buildbot@webrtc.org [Thu, 23 Oct 2014 21:36:17 +0000 (21:36 +0000)]
(Auto)update libjingle 78344087-> 78381351

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7515 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoBreak out WebRtcNs_ComputeDdUpdate function in ns_core
aluebs@webrtc.org [Thu, 23 Oct 2014 19:54:33 +0000 (19:54 +0000)]
Break out WebRtcNs_ComputeDdUpdate function in ns_core

This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7514 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoBreak out WebRtcNs_UpdateNoise function in ns_core
aluebs@webrtc.org [Thu, 23 Oct 2014 19:49:42 +0000 (19:49 +0000)]
Break out WebRtcNs_UpdateNoise function in ns_core

This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7513 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoBreak out FFT function in ns_core
aluebs@webrtc.org [Thu, 23 Oct 2014 19:36:42 +0000 (19:36 +0000)]
Break out FFT function in ns_core

This is done in order to make the code more readible and maintainable.
This introduces an error of only +1 and -1.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7512 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoBreak out ComputeSnr function in ns_core
aluebs@webrtc.org [Thu, 23 Oct 2014 19:34:14 +0000 (19:34 +0000)]
Break out ComputeSnr function in ns_core

This is done in order to make the code more readible and maintainable.
The output is bit-exact.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7511 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdding three video conference bots test
houssainy@google.com [Thu, 23 Oct 2014 16:45:07 +0000 (16:45 +0000)]
Adding three video conference bots test

A video conference between three bots, each bot creating two
peerConnections, and each peer connected to one of the other bots.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7510 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdding file from test.webrtc.org domain to be downloaded
houssainy@google.com [Thu, 23 Oct 2014 15:41:30 +0000 (15:41 +0000)]
Adding file from test.webrtc.org domain to be downloaded

This has been configured to allow cross domain to access this generated
file:
https://test.webrtc.org/test-download-file/9000KB.data

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7509 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdd macros and APIs for webrtc histograms.
asapersson@webrtc.org [Thu, 23 Oct 2014 12:57:56 +0000 (12:57 +0000)]
Add macros and APIs for webrtc histograms.

BUG=crbug/419657

Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.

R=andresp@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days ago(Auto)update libjingle 78296920-> 78342456
buildbot@webrtc.org [Thu, 23 Oct 2014 12:22:06 +0000 (12:22 +0000)]
(Auto)update libjingle 78296920-> 78342456

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7507 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoDownload full Chromium checkouts by default
kjellander@webrtc.org [Thu, 23 Oct 2014 12:17:58 +0000 (12:17 +0000)]
Download full Chromium checkouts by default

This changes sync_chromium.py to download a full Chromium
checkout instead of one with no history. It has been noticed
that the download of the no-history checkout is very slow, even
when on high-speed internet connections, due to current limitations
in the Git backend serving these clones.
Switching to a full checkout is faster, but requires more bandwidth
and disk space.

To keep the old behavior, users must set the CHROMIUM_NO_HISTORY
environment variable to 1.

Using a full checkout also enables the use of the Chromium
infrastructure teams' Git cache functionality, that speeds up
the initial download and also heavily reduces the traffic when
setting up multiple checkouts on the same machine.
This is not enabled by default, but is supported if the user is
setting the cache_dir variable in his checkout's .gclient file to
point at a directory on local disk.

BUG=3882
TESTED=
* Ran gclient sync and verified chromium/src now contained a Git
repo with full history.
* Tested rolling chromium_revision in DEPS forward + sync.
* Tested rolling it back again + sync.
* Tested with an existing no-history checkout:
  CHROMIUM_NO_HISTORY=1 gclient sync
  No change was performed.
* Tested with a .gclient that had cache_dir configured.
* Verified error message is displayed when .gclient has cache_dir
  configured and CHROMIUM_NO_HISTORY=1.

R=iannucci@chromium.org

Review URL: https://webrtc-codereview.appspot.com/22869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7506 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdds support for sending first set of packets at increasingly higher bitrates to...
stefan@webrtc.org [Thu, 23 Oct 2014 11:57:05 +0000 (11:57 +0000)]
Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.

Also wires up a finch experiment to control this.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoUsing the Unused turn configuration in two way test
houssainy@google.com [Thu, 23 Oct 2014 08:40:53 +0000 (08:40 +0000)]
Using the Unused turn configuration in two way test

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7504 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoLet video_loopback use internal VCM capturers.
pbos@webrtc.org [Thu, 23 Oct 2014 08:24:02 +0000 (08:24 +0000)]
Let video_loopback use internal VCM capturers.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7503 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdd a memcheck exclusion for EndToEndTest.CanSwitchToUseAllSsrcs.
andrew@webrtc.org [Thu, 23 Oct 2014 05:37:37 +0000 (05:37 +0000)]
Add a memcheck exclusion for EndToEndTest.CanSwitchToUseAllSsrcs.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7502 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days ago(Auto)update libjingle 78273470-> 78296920
buildbot@webrtc.org [Wed, 22 Oct 2014 22:02:00 +0000 (22:02 +0000)]
(Auto)update libjingle 78273470-> 78296920

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7501 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMerging Henrik's and Peter's changes for AppRTCDemo
glaznev@webrtc.org [Wed, 22 Oct 2014 17:43:37 +0000 (17:43 +0000)]
Merging Henrik's and Peter's changes for AppRTCDemo
from https://github.com/hkjellander/AppRTCDemo.

Description of changes:
- Add connect screen with an option to enter room number or select loopback mode.
- Add 'hangup' and 'WebRTC statistics' buttons to AppRTCDemo activity.

BUG=3938
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7500 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoNOTE: This code review based on the running issue:
houssainy@google.com [Wed, 22 Oct 2014 17:24:20 +0000 (17:24 +0000)]
NOTE: This code review based on the running issue:
https://webrtc-codereview.appspot.com/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7499 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoAdding Two way video and audio streaming test to RtcBot
houssainy@google.com [Wed, 22 Oct 2014 17:17:15 +0000 (17:17 +0000)]
Adding Two way video and audio streaming test to RtcBot

NOTE: This code review based on this running issue:
https://review.webrtc.org/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7498 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoHTTPS Server used instead of HTTP for loading the bots to avoid the media permission...
houssainy@google.com [Wed, 22 Oct 2014 16:34:25 +0000 (16:34 +0000)]
HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test.

This code review based on the running issue:
https://webrtc-codereview.appspot.com/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7497 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days ago(Auto)update libjingle 78262388-> 78262615
buildbot@webrtc.org [Wed, 22 Oct 2014 15:45:17 +0000 (15:45 +0000)]
(Auto)update libjingle 78262388-> 78262615

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7496 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRemove some disabled tests in WebRtcVideoEngine2.
pbos@webrtc.org [Wed, 22 Oct 2014 15:36:54 +0000 (15:36 +0000)]
Remove some disabled tests in WebRtcVideoEngine2.

Removes some tests that shouldn't have to be implemented or have already
been through other tests.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7495 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoSuppress libyuv uninitialized read in CopyRow_AVX
kjellander@webrtc.org [Wed, 22 Oct 2014 13:51:49 +0000 (13:51 +0000)]
Suppress libyuv uninitialized read in CopyRow_AVX

BUG=libyuv:377
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7494 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMake ReconfigureVideoEncoder use current bitrate.
pbos@webrtc.org [Wed, 22 Oct 2014 12:15:24 +0000 (12:15 +0000)]
Make ReconfigureVideoEncoder use current bitrate.

Prevents bitrate drops when changing resolution etc.

R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/24069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7493 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoTighten up MSan blacklist.txt owners.
kjellander@webrtc.org [Wed, 22 Oct 2014 11:20:07 +0000 (11:20 +0000)]
Tighten up MSan blacklist.txt owners.

To avoid people adding stuff to the blacklist unless
it's really valid to do so.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7492 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoDisable TestVp8Impl.BaseUnitTest on MSan.
pbos@webrtc.org [Wed, 22 Oct 2014 10:30:30 +0000 (10:30 +0000)]
Disable TestVp8Impl.BaseUnitTest on MSan.

MemorySanitizer uses generic (non-optimized) libvpx which is not bit
exact. This may be a bug in upstream libvpx, but we're forced to disable
it now as it blocks launching the MSan bot.

R=stefan@webrtc.org
TBR=marpan@webrtc.org
BUG=3904

Review URL: https://webrtc-codereview.appspot.com/24089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7491 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoFor FIR packet, payload length is zero, so SendToNetwork function is failing.
stefan@webrtc.org [Wed, 22 Oct 2014 09:47:14 +0000 (09:47 +0000)]
For FIR packet, payload length is zero, so SendToNetwork function is failing.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRoll chromium_revision de13cf4..28d1981 (299488:300483)
kjellander@webrtc.org [Wed, 22 Oct 2014 06:43:29 +0000 (06:43 +0000)]
Roll chromium_revision de13cf4..28d1981 (299488:300483)

Mainly to pick up https://codereview.chromium.org/656293004/
to fix some MSan issues.

Summary of changes (https://chromium.googlesource.com/chromium/src/+/de13cf4..28d1981/DEPS):
* third_party/android_tools d2b8620..36bf7ac
* third_party/libyuv 455c66b..5a09c3e (1038:1130)
* third_party/usrsctp/usrsctplib a11b3c5..7accb99
* tools/gyp 1977:1990
* tools/swarming_client c28b74f..a57d7db

Clang updated 217949:218707 (git diff de13cf4..28d1981 tools/clang/scripts/update.sh)

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7489 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoBreak out WebRtcNs_Windowing function in ns_core
aluebs@webrtc.org [Tue, 21 Oct 2014 22:35:40 +0000 (22:35 +0000)]
Break out WebRtcNs_Windowing function in ns_core

This is done in order to make the code more readible and maintainable.
This introduces only +1 and -1 errors.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7488 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoBreak out WebRtcNs_Energy function in ns_core
aluebs@webrtc.org [Tue, 21 Oct 2014 22:14:10 +0000 (22:14 +0000)]
Break out WebRtcNs_Energy function in ns_core

This is done in order to make the code more readible and maintainable.
This generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7487 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoBreak out WebRtcNs_IFFT function in ns_core
aluebs@webrtc.org [Tue, 21 Oct 2014 21:27:00 +0000 (21:27 +0000)]
Break out WebRtcNs_IFFT function in ns_core

This is done in order to make the code more readible and maintainable.
This creates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7486 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days ago(Auto)update libjingle 78193292-> 78199328
buildbot@webrtc.org [Tue, 21 Oct 2014 20:44:16 +0000 (20:44 +0000)]
(Auto)update libjingle 78193292-> 78199328

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7485 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoFix local address leakage when IceTransportsType is relay
guoweis@webrtc.org [Tue, 21 Oct 2014 20:40:21 +0000 (20:40 +0000)]
Fix local address leakage when IceTransportsType is relay

As part of implementing IceTransportsType constraint, we should hide the raddr which is the mapped address to prevent local address leakage.

BUG=1179
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7484 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoBreak out WebRtcNs_UpdateBuffer function in ns_core
aluebs@webrtc.org [Tue, 21 Oct 2014 20:33:09 +0000 (20:33 +0000)]
Break out WebRtcNs_UpdateBuffer function in ns_core

This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7483 4adac7df-926f-26a2-2b94-8c16560cd09d