Simplifies the test so that it only tests the padding-related parts.
Header extensions for padding already has a dedicated test, as does
packet stats from RtpSenderEgress.
Bug: webrtc:11340
Change-Id: I88829409aac15f0aad0d4d634114731e819574bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218844
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34019}
Use cicular buffer instead of ever growing dynamic vector
That limits used memory and speed up fuzzing
Bug: chromium:1207177, chromium:1202535
Change-Id: Ia69ee7423f720942301b6d0b1a9c16a0cf1b3d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218602
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34002}
When verbose logs are enabled, SCTP packets will be dumped to debug
logs, allowing text2pcap to be used to generate PCAP files.
First start Chrome with verbose logs, and write those to file:
/path/to/chrome --enable-logging=stderr --v=4 2> out.log
Then extract the SCTP_PACKET traces and run text2pcap:
grep SCTP_PACKET out.log > sctp.log
text2pcap -n -i 132 -D -t '%H:%M:%S.' sctp.log sctp.pcapng
You may have to cut away more from the beginning if the debug logs
contain additional timestamps and more, e.g. like:
grep SCTP_PACKET out.log | cut -d ' ' -f 2- > sctp.log
Note that if there are multiple RTCPeerConnection objects created, each
will print out their packets to log, so to filter for a specific one:
grep "SCTP_PACKET DcSctpTransport0" out.log > sctp.log
Bug: webrtc:12614
Change-Id: Ibbceaf33719d09e7606247cb0496ddd827ea58bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218200
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33999}
The socket fuzzer is build as a structure-aware fuzzer where the full
public API is exercised as well as receival of SCTP packets with random
sequences of valid chunks.
It begins by putting the socket in a defined starting state and then,
based on the fuzzing data, performs a sequence of operations on the
socket such as receiving packets, sending data, resetting streams or
expiring timers.
This is the first iteration, and when running it a while and analyzing
code coverage, it will be modified to perform better. It could probably
be a little more random.
Bug: webrtc:12614
Change-Id: I50d6ffaecef5722be5cf666fee2f0de7d15cc2e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218500
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33998}
Moves OnSendSideDelayUpdated and OnSendPacketUpdated out from
rtp_sender_unittest and into rtp_sender_egress_unittest and
rtp_rtcp_impl2_unittest. The former test now only tests the logic for
updating send-side-delay stats. The latter is now on a proper
RtpRtcp-level and also verifies that frame timestamps makes it to the
egress (as assumed by the first test).
Bug: webrtc:11340
Change-Id: I784042ad91eb66a4d1eebdbbc625f9522528bfb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218502
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33996}
This is one step in getting rid of cricket::MediaType.
Bug: webrtc:12754
Fixes: webrtc:12764
Change-Id: Idee832572bdc4c0e3bfdec6fb31ec0ba9db3e995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218346
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33994}
This removes PacketRouter inheritance from RemoteBitrateObserver and TransportFeedbackSenderInterface.
Call binds methods for sending REMB and transport feedback messages from RemoteCongestionController to PacketRouter.
This is needed until the RTCPTranseiver is used instead of the RTP modules.
Bug: webrtc:12693
Change-Id: I7088de497cd6d1e15c98788ff3e6b0a2c8897ea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215965
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33993}
to avoid conflicts between
createOffer({voiceActivityDetection: false})
and the transceiver setCodecPreferences API
BUG=webrtc:12365
Change-Id: I369227103ab543f593b27145a37d3e5c19a59cd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218343
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#33992}
This is one step in getting rid of cricket::DtlsTransportState..
Bug: webrtc:12762
Change-Id: I65a6e72b587fd3dd6cdc1ce7fe201a2a9cfe936d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218460
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33991}
Xcode 12.5 triggers some warnings for -Wdeprecated-copy, and I believe
it is better to fix this problem than to suppress this warning.
Bug: webrtc:12749
Change-Id: I5ca5fd8fdcae18fe7d3941f78b3366b5f03b8c00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33990}
- The FEC receiver tracks maximum of 48 media packets at a time, and packet reordering can delay the FEC packet from its protected media packets by more than 48 sequences.
- Such FEC packets do not get purged until much later when newer FEC packets with much higher sequence mark them as old.
- Until that happens, they sit in the receiver queue, wasting CPU cycles.
- If the receiver maintains a larger queue size for the media packets, it increases possibility of having all media packets in the queue, thereby organically purging the FEC packet.
- More importantly, this also increases the efficacy of FEC decode for such packet, since media packets now remain relevant for longer and aid in lost packet recovery.
Bug: webrtc:12656
Change-Id: Id0058df9a23ea31839decf2c37e0670a54c947fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215882
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33989}
When VideoFrameType for svc upper layer is kVideoFrameDelta for key pic,
the svc unittest will fail due to the wrong frame type for the super
frame of first key picture.
Bug: None
Change-Id: Iff026aaecb73890d3c45d2c88c9654a12d6fe3bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216461
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/master@{#33986}
When a socket is shutting down, either explicitly by the ULP calling
Shutdown(), or when the socket receives a SHUTDOWN chunk, the socket
should send all outstanding data and when all is sent and acked,
_then_ it should continue the shutdown protocol.
As it currently doesn't calculate correctly when all data has been sent,
as NACKED chunks are not included in what it believes is remaining in
the retransmission queue, it will shut down prematurely and may go back
to a previous state (ShutdownPending) from ShutdownSent or
ShutdownAckSent.
This is a workaround that just avoids the illegal state transition as
that puts the socket in an inconsistent state. The bug is merely
theoretical as WebRTC doesn't currently gracefully shut down a SCTP
socket, but just terminates the DTLS transport.
As TODOs mention, this will be fixed correctly a bit later.
Bug: webrtc:12739
Change-Id: Ibde2acc3a6aca701ac178d6181028404d470a5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218340
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33982}
Historically the PacketBuffer used a callback for assembled frames, and because of that RtpPacketInfos were piped through it even though they didn't have anything to do with the PacketBuffer.
With this CL RtpPacketInfos are stored in the RtpVideoStreamReceiver(2) instead.
Bug: webrtc:12579
Change-Id: Ia6285b59e135910eee7234b89b23425bb0fc0d2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215320
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33980}
Before this CL, the RemoteEstimatorProxy used a std::map to track which
arrival time a packet with a certain sequence number was received at.
While this works, it's fairly slow as most manipulations and insertions
are O(log(N)) and there were quite many of them.
By taking advantage that sequence numbers generally are received in
sequence, recording a packet is now amortized O(1). Also other
operations such as creating the periodic feedback reports, are also
much faster as it previously was done by searching quite a few times
in that map.
In highly loaded Media Servers, RemoteEstimatorProxy's usage of
std::map attributes to around 0.52% CPU.
Bug: webrtc:12689
Change-Id: I3dd58105f9fbfb111f176833cd4aa6b040c0e01d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217388
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33979}
While it's not strictly defined, the expectation is that sending a
message with a lifetime parameter set to zero (0) ms should allow it to
be sent if it can be sent without being buffered. If it can't be
directly sent, it should be discarded.
This is initial support for it. Small messages can now be delivered fine
if they are not to be buffered, but fragmented messages could be partly
sent (if this fills up the congestion window), which means that the
message will then fail to be sent whenever the congestion window frees
up again. It would be better to - at a higher level - realize early that
the message can't be sent in full, and discard it without sending
anything. But that's an optimization that can be done later.
A few off-by-one errors were found when strictly defining that the
message is alive during its entire lifetime. It will expire just _after_
its lifetime.
Sending messages with a lifetime of zero may not supported in all
libraries, so a workaround would be to set a very small timeout instead,
which is tested as well.
Bug: webrtc:12614
Change-Id: I9a00bedb639ad7b3b565b750ef2a49c9020745f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217562
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33977}
Keeping just the header doesn't save memory because header is taken as slice
of the original packet (and thus keeps a reference to the buffer containing
full packet)
Keeping full packet is simpler and avoid extra unused buffer created during
RtpPacket default contruction
Bug: b/187593466
Change-Id: I78d7201d110092fc039203e1caa2fb9c3afbc079
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218161
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33974}
