and fix other tests to actually include CN in supported codecs.
BUG=webrtc:12365
Change-Id: I7154dd0046497604a6fc825ce81e85c1b9cadcef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34033}
The number of invokes will vary based on number of receivers/senders,
so we can't have a fixed number there.
Bug: none
Change-Id: Iab3d529a5935c4b6cc95e9da6637acd880614972
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219101
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34027}
The minimum RTO time shouldn't be lower than the delayed ack timeout
of the peer to avoid sending retransmissions before the peer has
actually intended to reply.
In usrsctp, the default delayed ack timeout is 200ms and configurable
using the `sctp_delayed_sack_time_default` option. In dcsctp, it's
min(RTO/2, 200ms), to avoid this issue.
Bug: webrtc:12614
Change-Id: Ie84c331334af660d66b1a7d90d20f5cf7e2a5103
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219100
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34026}
Update `AudioProcessing::Config::ToString()` to also dump the config
from `AnalogGainController` which is missing.
Bug: webrtc:7494
Change-Id: Iea5dab1f6abb9ec8581ce690a2a119f202b4d1e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219082
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34025}
Reduce to testing what RTPSender is actually interested in: that
packets are actually forwarded to the pacer.
Partially the old test was verifying TransmissionOffset header extension,
add an explicit test for that at RtpRtcp-level instead.
Bug: webrtc:11340
Change-Id: I62be39e1d9d8c214c3277f4f1326db05b937674a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218845
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34023}
For implementations where the signaling and worker threads are not
the same thread, this significantly cuts down on Thread::Invoke()s that
would block the signaling thread while waiting for the worker thread.
For Audio and Video Rtp receivers, the following methods now do not
block the signaling thread:
* GetParameters
* SetJitterBufferMinimumDelay
* GetSources
* SetFrameDecryptor / GetFrameDecryptor
* SetDepacketizerToDecoderFrameTransformer
Importantly this change also makes the track() accessor accessible
directly from the application thread (bypassing the proxy) since
for receiver objects, the track object is const.
Other changes:
* Remove RefCountedObject inheritance, use make_ref_counted instead.
* Every member variable in the rtp receiver classes is now RTC_GUARDED
* Stop() now fully clears up worker thread state, and Stop() is
consistently called before destruction. This means that there's one
thread hop instead of at least 4 before (sometimes more), per receiver.
* OnChanged triggered volume for audio tracks is done asynchronously.
* Deleted most of the JitterBufferDelay implementation. Turns out that
it was largely unnecessary overhead and complexity.
It seems that these two classes are copy/pasted to a large extent
so further refactoring would be good in the future, as to not have to
fix each issue twice.
Bug: chromium:1184611
Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34022}
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}
