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394 Commits

Author SHA1 Message Date
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
mikhal@webrtc.org
36b95b4753 Adding a recording tool to vie autottest 
Review URL: https://webrtc-codereview.appspot.com/746006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-27 21:39:50 +00:00
henrike@webrtc.org
f7884f9900 Revert 2660 - updating cl 
Adding debug options

BUG=

Review URL: https://webrtc-codereview.appspot.com/751005

TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/752007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-25 02:00:19 +00:00
henrike@webrtc.org
3387b88595 Makes it possible to disable frame dropping in the VP8 codec. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/757006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-24 23:01:38 +00:00
mikhal@webrtc.org
6a6121c0b1 updating cl 
Adding debug options

BUG=

Review URL: https://webrtc-codereview.appspot.com/751005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-24 22:08:25 +00:00
elham@webrtc.org
2a74de1e78 Bump version number to 3.11 
Review URL: https://webrtc-codereview.appspot.com/744005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-23 16:50:57 +00:00
leozwang@webrtc.org
d6fcf7f0da Add debug options to test app 
Description:
1. add apm debug option
2. add voice/video rtp dump option
3. front facing camera as the default, minor change

BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/728012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-21 18:07:14 +00:00
leozwang@webrtc.org
4ff8a9ad2f Print out more audio codec information in vie_auto_test 
It would be good to print more audio codec information to avoid
confusion

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/744004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-21 03:50:24 +00:00
stefan@webrtc.org
4e8eabaab1 Properly handle switching between simulcast and unicast streams. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/733010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2644 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-20 14:29:52 +00:00
henrike@webrtc.org
f7a58f868e Fixes VP8 issue with sending simulcast->non simulcast->simulcast. 
Review URL: https://webrtc-codereview.appspot.com/722013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-17 19:10:27 +00:00
mflodman@webrtc.org
6620c68b1a Changed test case for r2629. 
TBR=elham@webrtc.org

BUG=756

Review URL: https://webrtc-codereview.appspot.com/714011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2632 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-17 17:03:28 +00:00
mflodman@webrtc.org
42a4891699 Fixed issue for rtp extension. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/731012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-17 14:44:21 +00:00
andrew@webrtc.org
5dffebc4d1 Remove disabling of warning 4351 from non-interface files. 
This is handled in Chromium's build/common.gypi.

BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/724008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-16 04:24:05 +00:00
leozwang@webrtc.org
425e680808 Enable PLI as the default. 
Description:
Enable PLI as the default option.

BUG=webrtc issue 744
TEST=local
Review URL: https://webrtc-codereview.appspot.com/735008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-14 17:03:33 +00:00
mflodman@webrtc.org
90071dd647 Added API to set RTP timestamp offset extension. 
BUG=745

Review URL: https://webrtc-codereview.appspot.com/710011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-13 17:13:27 +00:00
mflodman@webrtc.org
1fb39ba422 REMB changes, cloned from issue 722011. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/708012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-13 17:05:14 +00:00
leozwang@webrtc.org
a11299648c Retrieve data from input 
Espeically on tablet, we have to read data dirtectly from input text edit rather than
track key input to let text edit get updated automatically

BUT=None
TEST=local test
Review URL: https://webrtc-codereview.appspot.com/705010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-11 04:33:02 +00:00
andrew@webrtc.org
cdfa63f94f Fix mismatched signature (due to const) error. 
TBR=mikhal@webrtc.org
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/717013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 22:56:17 +00:00
henrike@webrtc.org
7742479428 Fixes build bot breakage. Resizing was enabled which some tests assumed wouldn't be the case. Changed the default so that it is now disabled. 
Review URL: https://webrtc-codereview.appspot.com/731006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 19:31:24 +00:00
henrike@webrtc.org
268a24fa56 Reverts changes to auto test. 
Review URL: https://webrtc-codereview.appspot.com/724006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 15:15:51 +00:00
astor@webrtc.org
c0496e66f6 Expose a function for setting bandwidth estimation parameters in ViERTP_RTCP. 
Review URL: https://webrtc-codereview.appspot.com/678007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-10 10:14:43 +00:00