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
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
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
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
7bc465bd21
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
...
Introduces shared functions for timestamp and sequence number wrap checks.
BUG=1607
TESTS=trybots
Review URL: https://webrtc-codereview.appspot.com/1291005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3833 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-11 17:48:02 +00:00
523f93729b
Re-write the build of the nacklist.
...
Review URL: https://webrtc-codereview.appspot.com/1304008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3822 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-11 11:30:39 +00:00
ab9202b673
Removing remaining WebRtc_Word32 not in typedefs.h
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/1306006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3813 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-10 17:59:17 +00:00
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
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
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
2f44673d66
WebRtc_Word32 => int32_t for rtp_rtcp/
...
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1279007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-08 11:08:41 +00:00
19da719a5f
Resolves TSan v2 reports data races in voe_auto_test.
...
--- Note that I will add more fixes to this CL ---
BUG=1590
Review URL: https://webrtc-codereview.appspot.com/1286005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3770 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-05 14:34:57 +00:00
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
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
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
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
d8a6e72057
Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust.
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/1232005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3726 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-26 14:02:30 +00:00
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
41211466d8
Revert the deletion of test_api_nack.cc in r3674.
...
TBR=mflodman@webrtc.org , mikhal@webrtc.org
BUG=1513
Review URL: https://webrtc-codereview.appspot.com/1217004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3677 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-18 15:00:50 +00:00
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
b7edd06530
Remove DTMF detection. Talk team has been in the loop and there is no need for
...
DTMF detection at the receiver side.
test=voe_auto_test, VoE extended test DTMF
Review URL: https://webrtc-codereview.appspot.com/1168004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-12 22:27:27 +00:00
03e3117d87
Removed redundant VP8 width/height and made sure the generic width/height is set.
...
Review URL: https://webrtc-codereview.appspot.com/1158005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3656 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-12 09:59:27 +00:00
a27107004d
Split the NACK list into multiple RTCPs if it's too big.
...
TEST=rtp_rtcp_unittests
BUG=1434
Review URL: https://webrtc-codereview.appspot.com/1148006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3609 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-05 09:02:06 +00:00
44f85a49d8
Fixed coverity defects (CID 14657 and 14656).
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/1153006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3597 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-04 14:59:31 +00:00
603ae3ece2
Make RtpHeaderExtensionMap::Register and ::Deregister idempotent.
...
This CL changes the return code of these methods to indicate
success instead of failure when there is nothing to change.
This change appears to resolve an issue where enabling the
timestamp offset extension via SDP would result in a failure if
that extension had already been enabled.
Review URL: https://webrtc-codereview.appspot.com/1118008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3588 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-01 17:03:02 +00:00
e1c4ed958d
Fix to send a full NACK list at least roughly once every 1.5 x RTT.
...
BUG=1434
Review URL: https://webrtc-codereview.appspot.com/1111007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3576 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-27 16:23:06 +00:00
59b2d5fbce
Stop and restart fix.
...
BUG=1398
TEST=Local stop and start test.
Review URL: https://webrtc-codereview.appspot.com/1115004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3545 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-20 16:00:27 +00:00
0cb48a0a18
Set SingleStream BWE in unittests.
...
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1094004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3494 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-11 08:30:23 +00:00
a7303bdfb5
Lint-cleaned video and audio receivers.
...
BUG=
TESTED=trybots
Review URL: https://webrtc-codereview.appspot.com/1093004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3471 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-05 15:12:39 +00:00
244251a9cd
Moved almost all payload-related stuff to the payload registry.
...
The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video.
BUG=
TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/1078004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 13:23:07 +00:00
fa53d8717c
Fixing/disabling Windows x64 warnings
...
Disabled MSVC #4267 warnings in common.gypi to enable x64 builds
for Windows.
Fixed MSVC #4267 warnings in test/testsupport.
Added third_party/directxsdk to .gitignore.
With http://review.webrtc.org/1070008 landed, this should make it possible
to build for x64 on Windows.
BUG=1348
TEST=Compiling with http://review.webrtc.org/1070008 applied:
set GYP_DEFINES="target_arch=x64"
set GYP_GENERATORS=ninja
gclient sync
ninja -C out\Debug_x64
Review URL: https://webrtc-codereview.appspot.com/1060008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3464 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 10:07:17 +00:00
becf9c897c
Fix mismatch between different NACK list lengths and packet buffers.
...
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.
BUG=1289
Review URL: https://webrtc-codereview.appspot.com/1065007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 15:09:57 +00:00
b586507986
Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
...
Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.
BUG=1298
Review URL: https://webrtc-codereview.appspot.com/1060005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 14:33:42 +00:00
63e0964039
Fix webrtc compilation errors for Chrome Win64
...
Mostly disabling warnings in the gyp files.
BUG=1348
BUG=http://crbug.com/166496
BUG=http://crbug.com/167187
Review URL: https://webrtc-codereview.appspot.com/1063011
Patch from Justin Schuh <jschuh@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3423 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-29 06:45:22 +00:00
43da54a458
Reformatted rtp_sender: made lint clean.
...
TESTED=rtp_rtcp_unittests
BUG=
Review URL: https://webrtc-codereview.appspot.com/1062004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3412 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-25 10:53:38 +00:00
5accd370e7
RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies.
...
BUG=
TESTED=vie/voe_auto_test, rtp_rtcp_unittests
Review URL: https://webrtc-codereview.appspot.com/1058004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3397 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-22 12:31:01 +00:00
a678a3baee
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
...
TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1044004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-21 07:42:11 +00:00
a3c82bf667
Remove '<(library)' in gyp files.
...
This will remove all usage of '<(library)' in all webrtc gyp files.
Review URL: https://webrtc-codereview.appspot.com/1049005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-18 23:42:21 +00:00
efae5d5901
Extracted rtp receiver payload management to its own class, made video receiver depend on that instead.
...
Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine.
BUG=
TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/1022011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 16:10:45 +00:00
20ed36dada
Break out RtpClock to system_wrappers and make it more generic.
...
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.
Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.
TEST=vie_auto_test, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1041004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 14:01:20 +00:00
acfdd96ee3
Reformatted rtp_rtcp_impl*.
...
BUG=
TEST=Trybots.
Review URL: https://webrtc-codereview.appspot.com/1035004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3374 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-16 10:27:33 +00:00
a22a9bd9ca
Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.
...
The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.
BUG=
TEST=vie & voe_auto_test full runs
Review URL: https://webrtc-codereview.appspot.com/1014006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-14 10:01:55 +00:00
f908011eb4
Remove extra line.
...
TBR=elham
Review URL: https://webrtc-codereview.appspot.com/1024008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3365 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-11 22:39:55 +00:00
2f225cadde
Add logs when no RTCP RR has been received for three regular RTCP intervals.
...
BUG=1267
TEST=Unittest added.
Review URL: https://webrtc-codereview.appspot.com/1019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 13:54:43 +00:00
c38eef896a
Reformatted RTPReceiver.
...
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)
BUG=
TEST=Trybots, vie_ & voe_auto_test --automated
Review URL: https://webrtc-codereview.appspot.com/998007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 10:18:30 +00:00
1b6da28047
Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests.
...
Landing of 573005 On behalf of an1kumar@gmail.com
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1002008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3322 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-21 17:46:24 +00:00
ad0ed582b5
Fixed a missed initialization (found by valgrind FYI bot).
...
http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/327/steps/memory%20test%3A%20memcheck_voe_auto_test/logs/stdio
BUG=
TEST=Reproduced valgrind error, verified gone after fixing.
Review URL: https://webrtc-codereview.appspot.com/1008005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3318 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-20 09:14:36 +00:00
61f39a3174
Fixed bad header name.
...
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/1001008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3307 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-18 16:02:13 +00:00
07bf43c673
Replaced the _audio parameter with a strategy.
...
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.
In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.
BUG=
TEST=vie/voe_auto_test, trybots
Review URL: https://webrtc-codereview.appspot.com/1001006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-18 15:40:53 +00:00
7659d914bb
Decoupled video rtp receiver from rtp receiver.
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/995005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3292 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14 09:57:37 +00:00
