fe1ef935e7
Implement I420FrameCallbacks in Call.
...
BUG=2425
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2393004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5005 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 10:34:43 +00:00
3e00505e9a
Have padding decay to zero if no frames are being captured.
...
BUG=1837
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4998 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-18 15:05:29 +00:00
b400aa7cd4
Don't pad if only one stream is sent, except if auto muted.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2406004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4975 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 13:03:10 +00:00
7ea4f24ea5
Piping AutoMuter interface through to ViE API
...
This is a piece of the AutoMuter effort. A second CL will follow containing modifications to the new API, and tests.
BUG=2436
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2331004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4899 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 13:34:26 +00:00
b2c8a952a7
Improving padding rules and breaking out bw allocation to ViEEncoder.
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BUG=1837
TESTS=vie_auto_test --automated, trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2170004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:58:01 +00:00
822fbd8b68
Update talk to 50918584.
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Together with Stefan's http://review.webrtc.org/1960004/ .
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:38:54 +00:00
12dc1a38ca
Switch C++-style C headers with their C equivalents.
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The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.
BUG=1833
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1917004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 16:22:53 +00:00
6879c8adad
Hooking up first simple CPU adaptation version.
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BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1767004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4384 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 11:35:00 +00:00
aa4d96a134
Revert r4301
...
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
1a7b9b94be
Cleanup WebRTC tracing
...
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.
The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.
R=marpan@webrtc.org , pwestin@webrtc.org , turaj@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1761004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-08 21:31:18 +00:00
66b2e5c05a
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
...
rtp_rtcp implementation.
This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.
With this change the dead-or-alive and packet timeout APIs are removed.
TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1745004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05 14:30:48 +00:00
1c986e7c89
Removed ViE file API.
...
R=asapersson@webrtc.org , niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1723004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-26 09:12:49 +00:00
2e402ce873
Enqueue packet in pacer if sending fails
...
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.
BUG=1930
R=pwestin@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1693004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20 20:18:31 +00:00
508a84b255
Wire up pacer-based padding.
...
This connects the pacer-based padding with the RTP modules, which will
generate padding packets roughly according to what the pacer suggests.
It will only generate padding packets of maximum size to keep the number
off padding packets as small as possible. This also sets a limit of how much
padding + media bitrate which the pacer is allowed to "request" from the
RTP modules.
Padding will for now only be generated by the first sending RTP module.
BUG=1837
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1612005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 12:53:37 +00:00
c3cc375499
Add support for padding in pacer.
...
This improves pacer-based padding by making sure it limits padding according to:
- Never pad more than 800 kbps.
- Padding + media should not go above a given target bitrate.
Also adds appropriate unittests to make sure we reach the given targets.
BUG=1837
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1582005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4168 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 09:36:56 +00:00
cb9cff0c71
Add functions to ViE API to enable/disable the absolute send time header extension.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1487004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4065 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-20 12:00:23 +00:00
f5d4cb1958
Include files from webrtc/.. paths in video_engine/
...
BUG=1662
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1492004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 13:44:48 +00:00
b3e5acfb66
Cleanup traces in WebRTC
...
Remove some unused traces and add a trace counter for encoded video size.
R=holmer@google.com , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1476004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4050 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16 21:13:02 +00:00
7bfb3a3227
Add more tracing for key frames.
...
R=mallinath@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1428004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 22:59:00 +00:00
7707d060bb
Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1450008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4007 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13 10:50:50 +00:00
52b4e8871a
Adding trace and changing pacing constants
...
BUG=1721,1722
R=mikhal@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1380005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 19:02:17 +00:00
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
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
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
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
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
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
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
6648093911
Destroy VCM and VPM instead of delete.
...
Review URL: https://webrtc-codereview.appspot.com/1149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3586 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-01 14:51:23 +00:00
ef9f76a59d
Adding a receive side API for buffering mode.
...
At the same time, renaming the send side API.
Review URL: https://webrtc-codereview.appspot.com/1104004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 23:22:18 +00:00
3d305c64b4
Updates to send side streaming mode:
...
1. Disabling frame-droppers from the vie encoder and not the channel.
2. Accounting for qpMax in the VP8 wrapper.
Review URL: https://webrtc-codereview.appspot.com/1101007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3492 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-10 18:42:55 +00:00
fcd8585874
Enable external encoders with internal picture source.
...
CL enables registering of external encoder with internal picture
source on API by adding simple passthrough parameter that is already
supported within video engine.
Review URL: https://webrtc-codereview.appspot.com/1006006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3344 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 08:35:40 +00:00
d73527ccab
Changed assert to log.
...
Review URL: https://webrtc-codereview.appspot.com/1010004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3320 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-20 09:26:17 +00:00
bf41508807
Properly remove the bitrate observer when ViEEncoder is destructed.
...
BUG=1090
Review URL: https://webrtc-codereview.appspot.com/969013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3196 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 09:18:53 +00:00
f89fb9dbf0
Remove ViE lint warnings that should have been caught at upload time.
...
TEST=cpplint.py video_engine/*
Review URL: https://webrtc-codereview.appspot.com/964018
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3151 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-22 09:41:42 +00:00
571a1c035b
Enable paced sender.
...
Review URL: https://webrtc-codereview.appspot.com/965016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-13 21:12:39 +00:00
2f476ed760
Fix uninitialzed memory and cleanup.
...
BUG=1022
Review URL: https://webrtc-codereview.appspot.com/966006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3019 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-30 16:21:52 +00:00
d6ec386ff5
Revert the revert in r2988 since that wasn't the issue.
...
Review URL: https://webrtc-codereview.appspot.com/931005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2992 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-25 11:30:29 +00:00
8239ca5096
Reverse Merged r2884 & r2888 from trunk.
...
Review URL: https://webrtc-codereview.appspot.com/929005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2988 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-24 22:35:52 +00:00
9fedff7c17
Switching to I420VideoFrame
...
Review URL: https://webrtc-codereview.appspot.com/922004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2983 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-24 18:33:04 +00:00
14b43beb7c
Move src/ -> webrtc/
...
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
