Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.
Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.
BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36889004
Cr-Commit-Position: refs/heads/master@{#9040}
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8905}
This CL avoids changing the mentioned callbacks during a call, to avoid
a potential deadlock when acquiring _sendCritSect and calling
_mediaOpt.SetTargetRates.
Moving the critsect revealed a race for the FEC parameters in RtpVideoSender, so the CL grew a bit to avoid this. I also cleaned up some code here at the same time, but tried to keep it at a minimum since this CL had already increased a lot in size.
BUG=769
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42939004
Cr-Commit-Position: refs/heads/master@{#8899}
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
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
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
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
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
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
This was introduced in r6472 where the target bitrate was changed to be stored in bits/s instead of kbits/s, but the storage type was accidentally left as uint16_t. This caused the bitrate to be truncated, which at times causes NACKs to be dropped due to insufficient bitrate available.
BUG=3518
TEST=Tested in Chrome, trybots and verified that it fixes the bug in vie_auto_test loopback test.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6544 4adac7df-926f-26a2-2b94-8c16560cd09d
- An RTX packet with no payload should be dropped prior to parsing RTX header since it doesn't have an RTX header. This can for example happen when sending padding-only packets over the RTX stream.
- The retransmit code path when the pacer is disabled doesn't properly update the abs-send-time and ts-offset header extensions.
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5728 4adac7df-926f-26a2-2b94-8c16560cd09d
