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27 Commits

Author SHA1 Message Date
solenberg
b19d288c94 Remove chain of methods in RtpRtcp module to get current payload frequency for RTCP SRs, which anyway was stuck to defaults for video/audio. 
BUG=webrtc:2795,webrtc:6458

Review-Url: https://codereview.webrtc.org/2362373002
Cr-Commit-Position: refs/heads/master@{#14476}
 2016-10-03 13:22:32 +00:00
solenberg
d3d230f788 - Make RtpSenderAudio not inherit from DtmfQueue. 
- Remove unused method DtmfQueue::ResetDTMF()

BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2365873002
Cr-Commit-Position: refs/heads/master@{#14376}
 2016-09-23 20:10:50 +00:00
kwiberg
963be23e62 RtpRtcp: Remove the SetSendREDPayloadType and SendREDPayloadType methods 
The last in-tree call site recently disappeared, so they were unused.

BUG=webrtc:5922

Review-Url: https://codereview.webrtc.org/2066473002
Cr-Commit-Position: refs/heads/master@{#13751}
 2016-08-15 14:08:39 +00:00
Sergey Ulanov
525df3ffd1 Add EncodedImageCallback::OnEncodedImage(). 
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Original-Commit-Position: refs/heads/master@{#13615}
Cr-Commit-Position: refs/heads/master@{#13617}
 2016-08-03 00:46:47 +00:00
sergeyu
51db4dd1bd Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #14 id:300001 of https://codereview.chromium.org/2089773002/ ) 
Reason for revert:
broke browser_tests

Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Original-Commit-Position: refs/heads/master@{#13613}
> Cr-Commit-Position: refs/heads/master@{#13615}

TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691

Review-Url: https://codereview.webrtc.org/2203233002
Cr-Commit-Position: refs/heads/master@{#13616}
 2016-08-03 00:33:47 +00:00
Sergey Ulanov
4c7f4cd2ef Add EncodedImageCallback::OnEncodedImage(). 
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Commit-Position: refs/heads/master@{#13615}
 2016-08-02 22:14:51 +00:00
sergeyu
ac4dc2cefe Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #13 id:280001 of https://codereview.webrtc.org/2089773002/ ) 
Reason for revert:
broke internal tests

Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Commit-Position: refs/heads/master@{#13613}

TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691

Review-Url: https://codereview.webrtc.org/2206743002
Cr-Commit-Position: refs/heads/master@{#13614}
 2016-08-02 21:33:21 +00:00
Sergey Ulanov
ad34dbe934 Add EncodedImageCallback::OnEncodedImage(). 
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Cr-Commit-Position: refs/heads/master@{#13613}
 2016-08-02 20:44:25 +00:00
Sergey Ulanov
ec4f068bcd Style cleanups in RtpSender. 
- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h

R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2067673004 .

Cr-Commit-Position: refs/heads/master@{#13565}
 2016-07-28 22:19:18 +00:00
danilchap
7c9426cf38 Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module 
Review URL: https://codereview.webrtc.org/1877253002

Cr-Commit-Position: refs/heads/master@{#12359}
 2016-04-14 10:05:37 +00:00
skvlad
98bb6640d2 Added log messages for some important call setup events: 
- First audio RTP packet sent / received
 - First RTP packet of the first video frame sent / received
 - Last RTP packet of the first video frame sent / received
These timestamps should make it easier to measure how fast the call
becomes established from the user's perspective.

Review URL: https://codereview.webrtc.org/1765443002

Cr-Commit-Position: refs/heads/master@{#12287}
 2016-04-07 22:36:49 +00:00
solenberg
6021fe2b1e Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code. 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1803923003

Cr-Commit-Position: refs/heads/master@{#12003}
 2016-03-15 18:41:58 +00:00
Peter Kasting
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
danilchap
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
danilchap
162abd3562 lint whitespace warning removed from most rtp_rtcp/source/ files 
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.

BUG=webrtc:5277
R=pbos@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1512493002

Cr-Commit-Position: refs/heads/master@{#10966}
 2015-12-10 10:39:45 +00:00
Peter Boström
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
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:43:50 +00:00
pbos@webrtc.org
7c4d20fd6c Remove potential deadlock in RTPSenderAudio. 
Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls
by doing a lot shorter locking which fetches a current state of
RTPSenderAudio variables before sending.

Thread annotates locked variables and removes one lock in
RTPSenderAudio, bonus fixes data races reported in voe_auto_test
--automated under TSan (DTMF data race).

Also includes some bonus cleanup of RTPSenderVideo which removes the
send critsect completely as all methods using it was always called
from RTPSender under its send_critsect.

R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
BUG=3001, chromium:454654

Review URL: https://webrtc-codereview.appspot.com/41869004

Cr-Commit-Position: refs/heads/master@{#8348}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 12:20:50 +00:00
pkasting@chromium.org
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
pbos@webrtc.org
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
wu@webrtc.org
ebdb0e3ad0 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. 
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.

This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.

BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-06 23:49:08 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +00:00
pbos@webrtc.org
a048d7cb0a Include files from webrtc/.. paths in rtp_rtcp/ 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1557004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 14:27:38 +00:00
pbos@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
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
andrew@webrtc.org
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