9d453931c5
Change return value for number of discarded packets to be int.
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R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 07:07:44 +00:00
eec6ecdb1e
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
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---
Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition
This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional
This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).
BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org , pkasting@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 19:09:59 +00:00
b9f5453e29
Add boilerplate code for H.264.
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R=mflodman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 12:42:07 +00:00
f223746521
Upping start bitrate to min, if set to a lower value i SetSendCodec.
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BUG=3276
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21379005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6014 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 12:38:42 +00:00
46106f2a05
Casting char to int in logs.
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BUG=3153
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12369006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5979 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 07:02:52 +00:00
5574dacd1f
Log Fixit for parts of video_engine folder.
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BUG=3153
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5853 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-07 10:56:31 +00:00
6031001565
If the configured start bitrate is higher than the configures max
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bitrate, cap the star rate accordingly.
BUG=2720
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5327 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 15:07:12 +00:00
0a3c1471b8
Add API to query video engine for the send-side delay.
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R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5225 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:05:07 +00:00
331d4402fc
Connect pacer/padding to SuspendBelowMinBitrate
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The suspend function must not be engaged unless padding is also enabled.
This CL makes the connection so that the pacer and padding is enabled
when SuspendBelowMinBitrate is.
Had to change the unit test to make it aware of the padding packets.
BUG=2606
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5153 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 14:05:40 +00:00
ce8e0936d9
Rename AutoMute to SuspendBelowMinBitrate
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Changes all instances throughout the WebRTC stack.
BUG=2436
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 12:18:43 +00:00
1a3a6e5340
Removing the threshold from the auto-mute APIs
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The threshold is now set equal to the minimum bitrate of the
encoder. The test is also changed to have the REMB values
depend on the minimum bitrate from the encoder.
BUG=2436
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 10:16:14 +00:00
7ea4f24ea5
Piping AutoMuter interface through to ViE API
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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
814e28413d
Revert r4562
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R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2117004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4623 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 23:21:03 +00:00
b655985abd
Added choice of decode error mode to loopback test.
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R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1997004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4562 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-16 23:07:14 +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
735a7c8b93
Add back is.FrameProvider() call lost in r4194.
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BUG=2119
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1946004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4480 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 09:03:03 +00:00
aa4d96a134
Revert r4301
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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
66b2e5c05a
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
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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
65a1f2cb2b
Remove log of undefined input values in GetCodec.
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BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1755004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4286 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-02 13:02:14 +00:00
1c986e7c89
Removed ViE file API.
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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
3ba883f0fc
Removing functionality for inserting pre-encoded frames instead of raw
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video frames. The functionality hasn't been used for a long time and
should be done properly if used in the future.
This is a pre-step for implementing CPU overload control.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1630004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4194 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07 13:57:57 +00:00
f5d4cb1958
Include files from webrtc/.. paths in video_engine/
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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
d72262dc01
Fix compile errors in ViE with latest clang.
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Rolling to the latest Chromium picks up a new clang, which catches a fresh error:
error: 'reinterpret_cast' to class 'webrtc::VideoEngineImpl *' from its base at non-zero offset 'webrtc::VideoEngine *' behaves differently from 'static_cast' [-Werror,-Wreinterpret-base-class]
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../webrtc/video_engine/vie_codec_impl.cc:36:31: note: use 'static_cast' to adjust the pointer correctly while downcasting
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~
static_cast
This was triggered by André's change here:
https://code.google.com/p/webrtc/source/detail?r=3986
which made VideoEngineImpl a derived class of VideoEngine (good).
Picked up one other error as well:
error: implicit conversion from 'long' to 'int' changes value from 9223372036854775807 to -1 [-Werror,-Wconstant-conversion]
AutoTestSleep(std::numeric_limits<long>::max());
~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This fixes the errors and is required before stable can be rolled in Chromium.
TBR=mflodman,andresp
Review URL: https://webrtc-codereview.appspot.com/1450004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3989 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-09 02:12:07 +00:00
aeff4f3003
Fixing Coverity issues.
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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
b238d1210b
WebRtc_Word32 -> int32_t in video_engine/
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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
b5bf54c4e7
Permit arbitrary payload names for kVideoCodecGeneric.
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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
8911ce46a4
Generic video-codec support.
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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
e07c661a29
VP8: Making key frame interval a tunnable parameter
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Review URL: https://webrtc-codereview.appspot.com/1070006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3444 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 16:37:13 +00:00
49273ffa79
logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid()
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BUG = Issue1283
Review URL: https://webrtc-codereview.appspot.com/1013008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3371 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-14 01:52:26 +00:00
4aee6b637d
Added API to get receive side video delay.
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BUG=1222
Review URL: https://webrtc-codereview.appspot.com/997004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3294 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14 14:02:10 +00:00
d6ec386ff5
Revert the revert in r2988 since that wasn't the issue.
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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.
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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
14b43beb7c
Move src/ -> webrtc/
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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
