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105 Commits

Author SHA1 Message Date
sprang@webrtc.org
779c3d16b9 Use ByteReader/ByteWriter instead of rtputility and manual shift/add. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41289004

Cr-Commit-Position: refs/heads/master@{#8761}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 16:44:54 +00:00
sprang@webrtc.org
3093390479 Parsing of transport wide sequence number rtp extension header. 
Plus some refactoring to correctly handle padding.

BUG=4311
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45429004

Cr-Commit-Position: refs/heads/master@{#8757}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 14:33:46 +00:00
guoweis@webrtc.org
fdd1057949 Add CVO support to Vie layer. 
1. standard plumbing CVO through vie layer.
2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation.

WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420.

BUG=4145
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429007

Cr-Commit-Position: refs/heads/master@{#8703}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 20:51:50 +00:00
guoweis@webrtc.org
4536289353 Add CVO support to RTP sender side. 
According to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf,
CVO byte should only be added in the last packet of each key frame or when the rotation changes. Currently, we're adding this byte in each frame to start with.

BUG=4145
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42439004

Cr-Commit-Position: refs/heads/master@{#8606}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 22:55:43 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro 
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 13:04:54 +00:00
pbos@webrtc.org
49096de442 DCHECK send DataCountersUpdated for valid SSRCs. 
Also updates RTPSender to not update RTX stats when RTX is disabled.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42399004

Cr-Commit-Position: refs/heads/master@{#8489}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-24 22:38:22 +00:00
pkasting@chromium.org
d324546ced Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : 
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36179004

Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 21:29:45 +00:00
sprang@webrtc.org
0200f70792 Set webrtc_rtp category to be disabled by default. 
Should not affect webrtc standalone. For chromium, disabling helps
mitigate viewing performance problems.

BUG=chromium:441440
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41909004

Cr-Commit-Position: refs/heads/master@{#8375}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:06:48 +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
asapersson@webrtc.org
4414939954 Add method for incrementing RtpPacketCounter. Removes duplicate code. 
Correction to check if rtx is enabled on send-side (and not receive) when updating rtx send bitrate stat.

Remove unneeded guarded by annotations.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41729004

Cr-Commit-Position: refs/heads/master@{#8239}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-04 08:35:21 +00:00
sprang@webrtc.org
c957ffc6dc Fixed potential crash if rtp packet history is completely full. 
Also performance enhanecement in rtp_sender (don't lookup if kDontStore)

BUG=4171
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39759004

Cr-Commit-Position: refs/heads/master@{#8226}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-02 13:08:14 +00:00
sprang@webrtc.org
43c883954f Allow rtp packet history to dynamically expand in size. 
When using the paced sender, packets will be put into the rtp packet
history and then retreived from there again when it is time to send.

In some cases (low send bitrate and very large frames created) this
may overflow, causing packets to be overwritten in the packet history
before they have been sent.

Check this condition and expand history size if needed.

This is primarily triggered during screenshare, when
switching to a large picture with lots of high frequency
details in it.

BUG=4171
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34879004

Cr-Commit-Position: refs/heads/master@{#8195}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 09:09:41 +00:00
asapersson@webrtc.org
273fbbb921 Update StreamDataCounter with FEC bytes. 
Add histograms stats for send/receive FEC bitrate:
- "WebRTC.Video.FecBitrateReceivedInKbps"
- "WebRTC.Video.FecBitrateSentInKbps"

Correct media payload bytes in StreamDataCounter to not include FEC bytes.

Fix stats for rtcp packets sent/received per minute (regression from r7910).

BUG=crbug/419657
R=holmer@google.com, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 12:17:29 +00:00
asapersson@webrtc.org
cfd82dfc11 Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. 
Prepares for adding FEC bytes to the StreamDataCounter.

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

Review URL: https://webrtc-codereview.appspot.com/37579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 09:39:59 +00:00
pbos@webrtc.org
0b0c24177b Only return Rtx mode in RTXSendStatus(). 
There is no need to return 'ssrc' and 'payloadtype' inside this function
since they are never used.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38569004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 14:15:15 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values. 
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-12 21:51:21 +00:00
andrew@webrtc.org
8f27fcce79 Revert 8028 "Support associated payload type when registering Rt..." 
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.

> Support associated payload type when registering Rtx payload type.
> 
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
> 
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26259004
> 
> Patch from Changbin Shao <changbin.shao@intel.com>.

TBR=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 20:22:46 +00:00
pbos@webrtc.org
2a169640a3 Support associated payload type when registering Rtx payload type. 
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.

BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26259004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 15:16:10 +00:00
pbos@webrtc.org
d16e839c6d Rtp-Rtcp sender cleanup. 
Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions.

Also removed const on non-pointer/reference types for related files.

BUG=
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34469004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 13:49:55 +00:00
stefan@webrtc.org
11d8176cb3 Move updating nack bitrate inside UpdateNACKBitRate. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 09:52:24 +00:00
pbos@webrtc.org
ce4e9a3562 Refactor some receive-side stats. 
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 13:50:16 +00:00
asapersson@webrtc.org
d08d389ce8 Add field to counters for when first rtp/rtcp packet is sent/received. 
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 12:03:11 +00:00
asapersson@webrtc.org
97d0489058 Add video send bitrates to histogram stats: 
- total bitrate ("WebRTC.Video.BitrateSentInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateSentInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps")
- retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")

Add retransmitted bytes to StreamDataCounters.

Change in UpdateRtpStats to also update counters for retransmitted packet.

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 09:47:53 +00:00
pbos@webrtc.org
9334ac2d78 Use vector of CSRCs for DeliverFrame & SetCSRCs. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28029004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 08:25:50 +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
stefan@webrtc.org
0bae1fab4a Wire up bandwidth stats to the new API and webrtcvideoengine2. 
Adds stats to verify bandwidth and pacer stats.

BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 14:05:29 +00:00
sprang@webrtc.org
dcebf2daa7 Reworked paced sender queue 
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.

Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.

Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 16:27:16 +00:00
stefan@webrtc.org
76960d5f74 For FIR packet, payload length is zero, so SendToNetwork function is failing. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 09:47:14 +00:00
andresp@webrtc.org
c3c29113d1 Expose setPayloadType on the rtp_sender. Thus allowing other users of this module 
to set the payload type to be used without having to call SendOutgoingData.

BUG=3694
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6988 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 09:39:43 +00:00
andresp@webrtc.org
817a034cf2 Fix TimeToSendPadding return to be 0 if no padding bytes are sent. 
BUG=3694
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15149005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6900 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-14 08:24:47 +00:00
stefan@webrtc.org
8b94e3da0f Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled. 
This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 16:10:14 +00:00
pbos@webrtc.org
63c60ed224 Remove old padding path in RTPSender. 
Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple
of arguments from SendPadData().

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 09:37:29 +00:00
pbos@webrtc.org
2f4b14e3f3 Make RTCP sender report send media bytes. 
r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 15:25:39 +00:00
stefan@webrtc.org
168f23faa5 Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. 
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.

R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 13:44:02 +00:00
pbos@webrtc.org
72491b9a90 Count total bytes sent in RTPSender::Bytes(). 
Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 16:24:54 +00:00
andresp@webrtc.org
8f1512140e Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 09:39:23 +00:00
andresp@webrtc.org
d11bec40b2 Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 14:32:58 +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
pbos@webrtc.org
2bb1bdab8d Preserve RTP states for restarted VideoSendStreams. 
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.

Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 13:06:48 +00:00
stefan@webrtc.org
88e0dda475 Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 09:20:42 +00:00
henrike@webrtc.org
fe526ff10f fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions. 
BUG=N/A
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-25 20:59:51 +00:00
stefan@webrtc.org
a15fbfdcde Add round-robin selection of send stream to pad on. 
BUG=1812
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 17:32:05 +00:00
stefan@webrtc.org
ef92755780 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. 
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.

BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15629005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-05 08:25:29 +00:00
stefan@webrtc.org
420b2567f3 Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. 
This caused only the first retransmission to be successful.
Introduced with https://code.google.com/p/webrtc/source/detail?r=5728.

BUG=1811
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 12:17:15 +00:00
andrew@webrtc.org
2c3f1abb69 Replace flooding logs in rtp_sender.cc with a comment. 
Started occurring after:
https://webrtc-codereview.appspot.com/11129004

BUG=3153
R=andresp@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 21:26:34 +00:00
stefan@webrtc.org
2f8d5f3302 Check if a header extension is registered before updating it and fail silently if it's not. 
BUG=
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 12:28:46 +00:00
andresp@webrtc.org
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
andresp@webrtc.org
d09d074827 Protect write of send_target_bitrate. 
This issue was catch by tsan bot.

BUG=3065
R=stefan@webrtc.org, andrew

Review URL: https://webrtc-codereview.appspot.com/10619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 14:27:34 +00:00
sprang@webrtc.org
efcad39f77 Fix race condition in RTPSEnder. 
In RTPSender::SendPayloadType(), payload_type_ should not be read
without owning send_critsect_.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 16:51:35 +00:00
stefan@webrtc.org
16395228f5 Properly account for retransmitted packets when not using the pacer. 
This regression was introduced in r5728.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-19 19:34:07 +00:00