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Author SHA1 Message Date
mikhal@webrtc.org
47128ab5ab Removing vie file related code from vie_custom_call 
Follow up on https://code.google.com/p/webrtc/source/detail?r=3900

Review URL: https://webrtc-codereview.appspot.com/1361004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 20:09:54 +00:00
pwestin@webrtc.org
4e545b33b3 Fixed remaining nits from Stefan 
Review URL: https://webrtc-codereview.appspot.com/1323007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 15:23:34 +00:00
pwestin@webrtc.org
91563e42da Fix the encoder pause logic. 
BUG=1691

Review URL: https://webrtc-codereview.appspot.com/1352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3904 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 22:20:08 +00:00
mikhal@webrtc.org
b84f13f185 Disabling avi file interface 
Review URL: https://webrtc-codereview.appspot.com/1351004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3900 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-25 18:07:32 +00:00
stefan@webrtc.org
8ca8a71de2 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." 
This reverts commit aae26db1da5803482b094357c546b8454ab1c26d.

BUG=1613
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1327008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 16:48:32 +00:00
stefan@webrtc.org
ccd4b2aec8 Add a default RTT to CallStats and use different values for buffered/real-time mode. 
BUG=1613

Review URL: https://webrtc-codereview.appspot.com/1326007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 15:58:23 +00:00
pwestin@webrtc.org
63117339dc Updated the sync module with a slow moving filter 
Review URL: https://webrtc-codereview.appspot.com/1326008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 18:57:14 +00:00
mflodman@webrtc.org
7c9e992d05 Removed unused variable. 
Review URL: https://webrtc-codereview.appspot.com/1320013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3881 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 13:05:41 +00:00
mflodman@webrtc.org
aeff4f3003 Fixing Coverity issues. 
BUG=C14457, C10611

Review URL: https://webrtc-codereview.appspot.com/1320012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-22 12:41:57 +00:00
kjellander@webrtc.org
c41478f7eb Ensure build_demo.py run subprocesses with bash shell. 
It turns out the default shell becomes /bin/sh on Lucid. By specifying the shell for subprocess.check_call we ensure bash is used.

Thanks to yujie.mao@intel.com for pointing this out.

BUG=1659
TEST=Successful build with build_demo.py both on Ubuntu Lucid and Precise.
TBR=leozwang

Review URL: https://webrtc-codereview.appspot.com/1343004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3875 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-19 11:50:42 +00:00
mflodman@webrtc.org
65f995a3df New ViE interface. 
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/1113004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-18 12:02:52 +00:00
pbos@webrtc.org
6e788df19e Remove vim/emacs modelines from .gypi files 
BUG=1655

Review URL: https://webrtc-codereview.appspot.com/1326005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 12:40:34 +00:00
mflodman@webrtc.org
9f5ebb5251 Adding a payload type for RTX. 
BUG=736
TEST=Modified RTP unittests.

Review URL: https://webrtc-codereview.appspot.com/1278004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 14:55:46 +00:00
stefan@webrtc.org
b8e7f4cc97 Change capture interface to use NTP capture time. 
Move NTP functionality to Clock.

BUG=1563
TEST=trybots and vie_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/1313005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 11:56:23 +00:00
pwestin@webrtc.org
1de01354e6 Adding playout buffer status to the voe video sync 
Review URL: https://webrtc-codereview.appspot.com/1311004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 20:23:35 +00:00
elham@webrtc.org
1b2a6e0be0 Updated WebRTC version number to 3.29 
TBR=mallinath1 
Review URL: https://webrtc-codereview.appspot.com/1305005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 21:28:33 +00:00
fischman@webrtc.org
6f41ca9fd2 WebRTCDemo: Enable making multiple calls. 
Previously after the first call subsequent attempts to bind the RTP/RTCP ports would fail, since r3754.

BUG=1618

Review URL: https://webrtc-codereview.appspot.com/1302007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-10 20:33:27 +00:00
hclam@chromium.org
806dc3b0e6 More trace events 
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 19:54:10 +00:00
stefan@webrtc.org
7da3459b2a Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." 
This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1308004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 14:56:29 +00:00
pbos@webrtc.org
b238d1210b WebRtc_Word32 -> int32_t in video_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:41:51 +00:00
stefan@webrtc.org
afcc6101d0 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. 
We should consider making the same change to the render timestamps generated at the receiver.

BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1283005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:37:40 +00:00
pbos@webrtc.org
29758de9b6 Always set render delay in ViEChannel::RegisterExternalDecoder. 
BUG=1523

Review URL: https://webrtc-codereview.appspot.com/1219007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 00:34:42 +00:00
pwestin@webrtc.org
6faf71d27b Remove the old unused udp_transport 
Review URL: https://webrtc-codereview.appspot.com/1272009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 23:25:25 +00:00
mflodman@webrtc.org
367804cce2 Clean packets on the network when closing + made loopback test actually run again. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1290006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 10:42:50 +00:00
kjellander@webrtc.org
10eb92039b Add GYP target for WebRTC Video demo for Android. 
Add a build target for the Video demo app for Android that only
exists when OS=='android' during build.

Note that this doesn't solve webrtc:1029, it's more like a workaround
waiting for the complete solution, which is to great a proper GYP target
that doesn't involve an action and an external script.

BUG=1029
TEST=Built successfully with:
source build/android/envsetup.sh
gclient runhooks
ninja -C out/Debug
Also verified the target is not present when OS is not 'android'.

Review URL: https://webrtc-codereview.appspot.com/1286004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 13:36:32 +00:00
pbos@webrtc.org
b5bf54c4e7 Permit arbitrary payload names for kVideoCodecGeneric. 
BUG=1575

Review URL: https://webrtc-codereview.appspot.com/1282005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-05 13:27:38 +00:00
pwestin@webrtc.org
b9e402d99f Remove WEBRTC_*_ENGINE_NETWORK_API use 
Review URL: https://webrtc-codereview.appspot.com/1203009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:51:42 +00:00
edjee@google.com
79b0289bfc Adds event traces and counters for WebRTC receive side. 
Review URL: https://webrtc-codereview.appspot.com/1279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:43:34 +00:00
pwestin@webrtc.org
82dcc9ff11 Remove UDP transport API from ViE 
Review URL: https://webrtc-codereview.appspot.com/1232004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-02 20:37:14 +00:00
elham@webrtc.org
65243bdb18 Updated Webrtc version to 3.28 
Review URL: https://webrtc-codereview.appspot.com/1272006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3745 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-01 16:17:26 +00:00
solenberg@webrtc.org
a442d4d983 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. 
Today I had to figure out this code was legacy. Now next person doesn't have to.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 09:14:36 +00:00
leozwang@webrtc.org
458194ba65 Fix broken audio. 
The problem was introduced in 3712, no need to external transport in
real test app, revert the change.

TBR=pwestin@webrtc.org
BUG=1539
Review URL: https://webrtc-codereview.appspot.com/1266005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 20:55:54 +00:00
stefan@webrtc.org
e1a7193869 Fix flakiness in network up/down event tests when running under memcheck. 
TBR=pwestin@webrtc.org

BUG=1524

Review URL: https://webrtc-codereview.appspot.com/1261005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 17:01:48 +00:00
fischman@webrtc.org
add50b94a5 WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14. 
(required bumping minSdkVersion to 14)

This fixes a RuntimeException thrown on GalaxyNexus (but not N7, N4, or NS)
during startPreview() after the sequence of Start(), Stop(), Start(); seemingly
GN's OMX stack can't deal with parallel startPreview() & setPreviewDisplay() in
this situation.

Also:
- Only set the surface in the camera when valid
- Remove duplicate assignment
- Fix error check on voiceChannel allocation to account for multiple channel creation due to orientation change causing onDestroy()/onCreate() on the app, and rampant use of process-static holders for VoE data.

BUG=1537

Review URL: https://webrtc-codereview.appspot.com/1259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:48:34 +00:00
stefan@webrtc.org
bfacda60be Add interface to signal a network down event. 
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 16:36:01 +00:00
elham@webrtc.org
14c9909ef6 Updated WebRTC version to 3.27 
Review URL: https://webrtc-codereview.appspot.com/1235004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 21:59:19 +00:00
pwestin@webrtc.org
a078d5cc38 Bugfix for extended RTP/RTCP test 
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1234004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 20:03:03 +00:00
pwestin@webrtc.org
26e35e1d06 Move the VIE tests to use external transport instead of the built in udp transport 
Review URL: https://webrtc-codereview.appspot.com/1216010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 19:21:27 +00:00
marpan@webrtc.org
94bc4cf905 Add min and target bitrate to VideoCodec. 
Review URL: https://webrtc-codereview.appspot.com/1214004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 17:13:08 +00:00
stefan@webrtc.org
3d0b0d6902 Follow-up fix for r3681. 
TESTS=trybots and vie_auto_test
BUG=1514

Review URL: https://webrtc-codereview.appspot.com/1216006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-19 10:04:57 +00:00
elham@webrtc.org
f1ea0df728 Updated WebRTC version number to 3.26 
Review URL: https://webrtc-codereview.appspot.com/1219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:45:04 +00:00
stefan@webrtc.org
abc9d5b6aa Change VCM interface to take target bitrate in bits per second. 
This also solves issue 1469.

TESTS=trybots
BUG=1469

Review URL: https://webrtc-codereview.appspot.com/1215004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 17:00:51 +00:00
pbos@webrtc.org
8911ce46a4 Generic video-codec support. 
Labels frames as key/delta, also marks the first RTP packet of a frame as such,
to allow proper reconstruction even if packets are received out of order.

BUG=1442
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1207004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-18 16:39:03 +00:00
mikhal@webrtc.org
bda7f305c5 Adding RTX on source 
Review URL: https://webrtc-codereview.appspot.com/1190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-15 23:21:52 +00:00
pwestin@webrtc.org
684f0577fb Revert r3667 and r3665 
Review URL: https://webrtc-codereview.appspot.com/1199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 23:20:57 +00:00
pwestin@webrtc.org
2dc0367406 Added destructors for tests to control destruct order 
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1197005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 21:36:10 +00:00
mikhal@webrtc.org
15960c2b67 Increasing size of nack list in buffered mode. 
Review URL: https://webrtc-codereview.appspot.com/1187007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 20:52:49 +00:00
pwestin@webrtc.org
361bac7a4f Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. 
Review URL: https://webrtc-codereview.appspot.com/1029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 17:52:42 +00:00
pbos@webrtc.org
927296fd1b Lazy capture_device_info acquisition. 
BUG=1484

Review URL: https://webrtc-codereview.appspot.com/1169005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-08 13:12:29 +00:00
mikhal@webrtc.org
efe4edb6da Enabling bufffering mode with no sync module or VoE 
BUG= 1454

Review URL: https://webrtc-codereview.appspot.com/1149006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-06 23:29:33 +00:00