414 Commits

Author SHA1 Message Date
mflodman@webrtc.org
aeb37d34aa Changed CriticalSectionScoped so the style correct constructor is used everywhere.
BUG=187

Review URL: https://webrtc-codereview.appspot.com/873009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2913 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-11 16:31:00 +00:00
leozwang@webrtc.org
71c13765ad Add a few functions to android test application
1. Add playing ringtone
2. Add receiving headset plug intent
3. Add a runnable to simulate cpu load

BUG=
TEST=trybot
Review URL: https://webrtc-codereview.appspot.com/858007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2892 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-09 20:20:37 +00:00
mflodman@webrtc.org
656477b930 Fixing Windows build.
Review URL: https://webrtc-codereview.appspot.com/864010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2888 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-08 20:58:37 +00:00
elham@webrtc.org
d63b18e67c Updated version number to 3.14
Review URL: https://webrtc-codereview.appspot.com/864009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2887 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-08 20:36:53 +00:00
mflodman@webrtc.org
15e4e34872 Wire up ssrc check in ViEEncoder for intra requests.
Review URL: https://webrtc-codereview.appspot.com/872004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2884 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-08 18:58:14 +00:00
stefan@webrtc.org
c530043684 Add per stream intra requests.
BUG=

Review URL: https://webrtc-codereview.appspot.com/829006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2883 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-08 07:06:53 +00:00
mflodman@webrtc.org
aca26292ae Hooking up EncoderStateFeedback to handle intra requests instead of trigger
ViEEncoder directly. This is one step towards adding send- and receive only
channels and getting rid of the default module.

Patch set 1 contains the reverted CL occasionally dead-locking:
http://review.webrtc.org/824004

BUG=769

Review URL: https://webrtc-codereview.appspot.com/859007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2880 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-05 16:17:41 +00:00
phoglund@webrtc.org
ad6612b1f8 Continuing to rewrite custom calls.
Added new helper for getting regular input. Rewrote remaining input handling for custom calls. Added a --choose_defaults flag which makes it possible to default on everything (e.g. with the flag, choosing custom call will accept all defaults and go directly to the call).

The next patch will add support for overriding arbitrary choices using flags. That is the point I want to arrive at and this patch paves the way for that. Fortunately it gets rid of some repetitive and bug-prone code on the way.

BUG=

Review URL: https://webrtc-codereview.appspot.com/858005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2878 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-05 12:34:09 +00:00
leozwang@webrtc.org
f9a0713866 Make Android.mk to be able to inclucde subfolder makefiles
BUG=None
TEST=trybot
Review URL: https://webrtc-codereview.appspot.com/868006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2870 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-04 15:59:11 +00:00
leozwang@webrtc.org
aee9120409 Move video android test to test folder
BUG=N/A
TEST=local test
Review URL: https://webrtc-codereview.appspot.com/863005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2862 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-02 20:13:56 +00:00
mikhal@webrtc.org
1e033e1594 Updating ConvertFromI420 to use VideoFrame - Related calls to DeliverFrame were also updated, and some refactoring (style-wise) was done on the way.
Review URL: https://webrtc-codereview.appspot.com/838006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2858 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-01 20:09:32 +00:00
phoglund@webrtc.org
8ff3ff1a8b Made ViE standard tests runnable under valgrind.
Ensured there are bugs for all open valgrind issues in the standard tests and suppressed the known issues. This way, we can get it running in continuous integration and keep new issues from entering.

Removed bad check in codec test, added suppressions.

Fixed simple memory leaks in tests.

BUG=Related to bug 329
TEST=Ran the vie_auto_test standard suite many times under valgrind to root out flakiness. Ran the standard suite without valgrind to ensure I didn't break anything.

Review URL: https://webrtc-codereview.appspot.com/843005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2854 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-01 10:04:26 +00:00
mflodman@webrtc.org
953368bf73 Remove unused video tests.
Review URL: https://webrtc-codereview.appspot.com/841010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2843 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-28 12:34:25 +00:00
mflodman@webrtc.org
f4f2145c6e Added API to set expected render delay.
BUG=905
TEST=API test added and manual delay tests.

Review URL: https://webrtc-codereview.appspot.com/810005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2841 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-28 11:27:35 +00:00
stefan@webrtc.org
a30eb31729 Make sure FEC packets aren't passed to the VCM with non-zero length.
BUG=902

Review URL: https://webrtc-codereview.appspot.com/843007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2840 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-28 11:13:08 +00:00
stefan@webrtc.org
64d9decc8d Move RtpToNtp functionality to its own file.
Removes the dependency on VideoEngine from RemoteBitrateEstimator.

BUG=

Review URL: https://webrtc-codereview.appspot.com/850004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2828 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-26 16:47:40 +00:00
elham@webrtc.org
db6eca446d updating version number to 3.13
Review URL: https://webrtc-codereview.appspot.com/842004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2824 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-25 21:53:25 +00:00
leozwang@webrtc.org
66ddf72a72 Correct filename which is missed in r2815
TBR=wu
BUG=None
TEST=try bot
Review URL: https://webrtc-codereview.appspot.com/833007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2817 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-24 19:16:15 +00:00
mikhal@webrtc.org
6e2e0b8ed2 Reverting r2812
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/829007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2816 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-24 19:14:40 +00:00
mflodman@webrtc.org
f2c750deee Hooking up EncoderStateFeedback to handle intra requests instead of trigger
ViEEncoder directly. This is one step towards adding send- and receive only
channels and getting rid of the default module.

BUG=769

Review URL: https://webrtc-codereview.appspot.com/824004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2812 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-24 16:20:47 +00:00
stefan@webrtc.org
2dcbcc147b Changing two asserts which should have returned errors instead.
BUG=

Review URL: https://webrtc-codereview.appspot.com/827007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2810 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-24 15:13:30 +00:00
asapersson@webrtc.org
ce42ace6ed Added initial fec configuration for rtp module.
Review URL: https://webrtc-codereview.appspot.com/833004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2808 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-24 11:33:49 +00:00
leozwang@webrtc.org
60c741281d Simplify SetLoudSpeaker calls and add a function to receive plug intent
Remove reduntant calls and add a function to receive plug intent.

BUG=None
TEST=local

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2806 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-24 07:06:40 +00:00
kjellander@webrtc.org
31b61b5fb6 Updating Android demo app src path for audio_device
Due to source files moved in r2804, the build.xml needed to be updated.

TBR=leozwang
TEST=AndroidNDK trybot
BUG=none

Review URL: https://webrtc-codereview.appspot.com/822005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2805 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-22 22:06:32 +00:00
kjellander@webrtc.org
63c002871a Fixing Android Demo build.xml for SDK 20.0.3
BUG=
TEST=Android NDK Trybot

Review URL: https://webrtc-codereview.appspot.com/826004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2799 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-21 14:22:22 +00:00
stefan@webrtc.org
976a7e61c1 Adding support for jointly estimating bandwidth using all streams from the same sending client.
- Broke out the bandwidth estimation from the RTP module.
- Added conversion between RTP and NTP time bases.
- Added unittests.

BUG=

Review URL: https://webrtc-codereview.appspot.com/784009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2798 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-21 13:20:21 +00:00
andrew@webrtc.org
9663686546 Make EncoderStateFeedbackObserver prototypes consistent.
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/824006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2797 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-20 23:33:17 +00:00
leozwang@webrtc.org
aaf6ba57e0 Fix crash in java code
The bug fix is to take global reference of context.

TBR=henrikg
BUG=issue 826
TEST=local test
Review URL: https://webrtc-codereview.appspot.com/798008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2789 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-19 16:39:07 +00:00
leozwang@webrtc.org
2db85bcba7 Make webrtc build with audio device java impl and add an option to enable it
BUG=
TEST=buildbots

This cl is to make audio device java implemenation build in webrtc, and add an
option in gyp so we can switch between opensl implementaiton and java
implementation.
Review URL: https://webrtc-codereview.appspot.com/801004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2783 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-18 20:19:00 +00:00
leozwang@webrtc.org
f851802bd7 Change prebuilt libraries
Because file struction was changed again

BUG=
TEST=local
Review URL: https://webrtc-codereview.appspot.com/785008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2778 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-17 00:03:41 +00:00
elham@webrtc.org
19f200edf3 Updating version number to 3.12
Review URL: https://webrtc-codereview.appspot.com/805004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2774 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-14 20:38:56 +00:00
mflodman@webrtc.org
0f27089e52 Refactored vie_autotest_simulcast.cc. This CL on changes the style and renames variables.
BUG=

Review URL: https://webrtc-codereview.appspot.com/787008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2768 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-13 08:12:32 +00:00
phoglund@webrtc.org
0df21d01f0 snprintf doesn't exist on windows.
TBR=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/792005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2762 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-12 17:02:10 +00:00
phoglund@webrtc.org
54d7faa5e3 Fixed release error.
TBR=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/785007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2760 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-12 16:36:25 +00:00
phoglund@webrtc.org
db81d5b8f6 Fixed errors from last patch.
BUG=
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/793007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2759 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-12 16:24:20 +00:00
phoglund@webrtc.org
f72943dadc Rewrote menu handling for vie custom call.
The intended trajectory of this patch is to abstract out all i/o for custom_call.
The reason is that kjellander@ will need to be able to configure custom calls using
flags, and using the same framework to gather all input gathering to a single place
will make this a lot easier.

This patch focuses on choices. The next will focus on field entries, like "enter
ip address" or "enter port number."

BUG=
TEST=Manually tested all menus in custom call, ran new unit tests.

Review URL: https://webrtc-codereview.appspot.com/757005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2758 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-12 15:59:22 +00:00
mflodman@webrtc.org
5a7507f26a Add API for transmission smotthening.
BUG=818
TEST=Only API tests added now.

Review URL: https://webrtc-codereview.appspot.com/787009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2756 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-12 13:47:06 +00:00
leozwang@webrtc.org
430e31c2c0 Change VQE settings
Change some VQE settings and make iSAC (item 0 in the list) to be the
default

BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/790005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2754 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-12 03:47:08 +00:00
stefan@webrtc.org
7c3523c1a4 Change audio/video sync to be based on mapping RTP timestamps to NTP.
Video Engine:
- Instead compensate for video capture delay by modifying RTP timestamps.
- Calculate the relative offset between audio and video by converting
  RTP timestamps to NTP and comparing receive time.

RTP/RTCP module:
- Removes the awkward video modification of NTP to compensate
  for video capture delay.
- Adjust RTCP RTP timestamp generation in rtcp_sender to have the same offset
  as packets being sent from rtp_sender.

BUG=
TEST=trybots,steam_synchronization_unittest

Review URL: https://webrtc-codereview.appspot.com/669010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2733 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-11 07:00:42 +00:00
andrew@webrtc.org
fa418ac0af Consolidate common_video targets to improve gyp run time.
Not sure if this change is measurable; perhaps a 1% savings.

BUG=webrtc:34

Review URL: https://webrtc-codereview.appspot.com/785005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2732 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-11 01:34:21 +00:00
stefan@webrtc.org
e37ecc6f81 Adding test for relaying all simulcast streams to different receive channels.
BUG=

Review URL: https://webrtc-codereview.appspot.com/776007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2726 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-10 13:27:47 +00:00
mflodman@webrtc.org
deaf685b66 Fix gcc 4.6 compilation for video_engine_unittest
TEST=Manually built using gcc 4.6.

Review URL: https://webrtc-codereview.appspot.com/787004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2725 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-10 13:19:08 +00:00
leozwang@webrtc.org
a96f8d9584 Change audio_processing libraries because of r2723
Buildbot will be ready soon, so such problem will hopefully not happen again.

BUG=None
TEST=local test
Review URL: https://webrtc-codereview.appspot.com/782004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2724 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-08 23:26:16 +00:00
andrew@webrtc.org
f3b65dbfe8 Remove WEBRTC_MAC_INTEL.
Review URL: https://webrtc-codereview.appspot.com/765008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2715 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-06 18:17:00 +00:00
andrew@webrtc.org
b3b158db2e Put output files in the output directory.
Review URL: https://webrtc-codereview.appspot.com/771006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2714 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-06 18:11:25 +00:00
mflodman@webrtc.org
3be5863405 Adding a class receiving key frame requests and relying to corresponding ViEEncoder. This CL adds the new class and unittest, but doesn't wire up th efunctionality. That will come in a follow soon after.
Also added include path in file_recorder.h to make video_engine_core_unittest compile.

BUG=769
TEST=New unittest added.

Review URL: https://webrtc-codereview.appspot.com/776004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2708 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-06 08:19:40 +00:00
leozwang@webrtc.org
be322d158e Correct wrong function name
Which is missed in last vie patch

TBR=ronghua

BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/762009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2705 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-05 17:11:34 +00:00
leozwang@webrtc.org
770d06bd01 Add libns which was added recently
BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/765007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2704 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-05 17:10:57 +00:00
mikhal@webrtc.org
954cf806d9 Adding the video debug api to vie test record
Review URL: https://webrtc-codereview.appspot.com/763004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2681 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-28 20:55:10 +00:00
mikhal@webrtc.org
e41bbdfecc Adding an API that allows recording of video data
removing vie_codec from cl

Moving debug call from Codec to File impl.

Updating cl following review

Updating file name

Updating cl following review.

Updating CL following review.

Adding an API that allows recording of video data

updating cl

Adding debug options

BUG=

Review URL: https://webrtc-codereview.appspot.com/751006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2678 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-28 16:15:16 +00:00