6955870806
Convert channel counts to size_t.
...
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org , henrika@webrtc.org , kjellander@webrtc.org , minyue@webrtc.org , perkj@webrtc.org , solenberg@webrtc.org , stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
1227e8b345
[rtp_rtcp] time helper functions
...
RTP timestams helper functions moved from rtp_utility
added functions to deal with CompactNtp timestamps
R=åsapersson
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1535113002
Cr-Commit-Position: refs/heads/master@{#11106}
2015-12-21 19:06:56 +00:00
6db6cdc604
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
...
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1513303003
Cr-Commit-Position: refs/heads/master@{#11025}
2015-12-15 10:54:50 +00:00
5c1def8892
modules/rtp_rtcp/include folder cleared of lint warnings
...
Functions that do not follow lint are marked deprecated, including function in the interface.
BUG=webrtc:5308
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1493403003
Cr-Commit-Position: refs/heads/master@{#10975}
2015-12-10 17:52:01 +00:00
ff761fba82
modules: more interface -> include renames
...
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
ebc0b4e993
Use webrtc/base/logging.h for rtp_rtcp.
...
BUG=webrtc:5118
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1422023002 .
Cr-Commit-Position: refs/heads/master@{#10437}
2015-10-28 15:39:43 +00:00
ac547a6538
Remove channel ids from various interfaces.
...
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org , mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
2015-09-17 21:06:02 +00:00
d436298332
Remove ResetStatistics from RTP feedback.
...
BUG=
R=asapersson@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1213603002
Cr-Commit-Position: refs/heads/master@{#9548}
2015-07-07 15:32:56 +00:00
026b892e72
Using << on an int8_t or uint8_t will output a character rather than a number.
...
Places that do this need to cast to int to get the desired behavior.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40579004
Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 19:54:19 +00:00
4591fbd09f
Use size_t more consistently for packet/payload lengths.
...
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
5a098c51ea
Refactor VP8 de-packetizer.
...
It's duplicated to parse VP8 RTP packet at the moment. We firstly call
RTPPayloadParser functions to save parsed information in RTPPayload
structure, then copy them to RTP header.
This CL removes RTPPayloadParser class and directly saves parsed data in
RTP header.
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7211 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 11:58:20 +00:00
6071b0636d
Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.
...
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.
This also highlighted a number of unused functions which I've removed.
-- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/ , but
-- a new cl was needed to resolve a small conflict before committing.
BUG=none
TEST=none
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 07:42:33 +00:00
1972ff8a6e
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
...
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions). I've removed some of
these.
This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class. Removed "virtual" in those
cases.
BUG=none
TEST=none
R=andrew@webrtc.org , henrik.lundin@webrtc.org , mallinath@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 06:20:28 +00:00
62bafae661
Some refactoring inside rtp_rtcp/.
...
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:10:51 +00:00
dc80bae2a6
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
...
Clean some logs and add asserts in the way.
BUG=3153
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-08 11:06:12 +00:00
48df38114d
Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
...
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.
Also makes sure that only valid timestamps and receive times are used for audio/video sync.
BUG=2608
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 15:18:52 +00:00
7bb8f02274
Adds support for combining RTX and FEC/RED.
...
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.
Enables retransmissions over RTX by default in the loopback test.
BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org , pbos@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2154004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:40:11 +00:00
286fe0b04d
Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ..."""
...
...and fixes the RTCP bug.
BUG=2277
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4588 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 20:58:21 +00:00
a0218a84d1
Revert 4582 "Reverts a second set of reverts caused by a bug in ..."
...
> Reverts a second set of reverts caused by a bug in a dependency.
>
> Revert "Revert r4328"
>
> Revert "Revert r4322 "Support sending multiple report blocks and keeping track
> of statistics on"
>
> BUG=1811
> R=henrika@webrtc.org , pbos@webrtc.org , tina.legrand@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2072004
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2087004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4585 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 19:44:13 +00:00
1a65d6c36b
Reverts a second set of reverts caused by a bug in a dependency.
...
Revert "Revert r4328"
Revert "Revert r4322 "Support sending multiple report blocks and keeping track
of statistics on"
BUG=1811
R=henrika@webrtc.org , pbos@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2072004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4582 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 16:22:21 +00:00
822fbd8b68
Update talk to 50918584.
...
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
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
9b82dced8d
Make sure first RTP packet counts as in-order.
...
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1811004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4350 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 13:03:35 +00:00
4a44ea21d7
Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc"
...
TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1803004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4346 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 21:46:06 +00:00
b7eda43810
Revert r4322 "Support sending multiple report blocks and keeping track of statistics on
...
several SSRCs"
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1774006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 21:08:27 +00:00
96edd56170
Sorted headers under rtp_rtcp/.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1781005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4325 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 15:40:42 +00:00
717d147ebb
Support sending multiple report blocks and keeping track of statistics on several SSRCs.
...
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org , tommi@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1768004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 13:39:27 +00:00
452d853c43
Fix three uninitialized members in rtp_receiver_impl.cc.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1781004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4320 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 10:54:56 +00:00
08933a5dfb
Initialize payload-type frequency in channel.cc.
...
Uninitialized values triggered divide-by-zero crashes in voe_auto_test.
BUG=
R=stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1780004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4319 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 10:06:29 +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
