As of
1e4d4fdf88
we no longer expect an InitEncode on deativation of a layer.
Bug: webrtc:12540
Change-Id: I10d447d90d1019258f662caf7f6e649d63d6927a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215076
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33709}
Since we want most users to use the PeerConnection API, this is the
part that we should document.
If we want people to use other APIs, we can add to the file.
Bug: webrtc:12674
Change-Id: Icf14f218cf51c640e6f846f10b49dff84106dc21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215066
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33707}
Adding fuzzers to the build made "gn gen --check" discover a lot
of dependency errors between various components of dcSCTP.
Bug: webrtc:12614
Change-Id: I0b2dd7321aec2624da417f413c727bd11b4743e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215003
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33705}
This prepares for ability to defer sequence number assignment to after
the pacing stage - a scenario where the RtpRtcp module rather than than
RTPSender class has responsibility for sequence numbering.
Bug: webrtc:11340
Change-Id: Ife88f60258b9b7cfd9dbd3326f02ac34da8f7603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214967
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33702}
No customers have been identified.
Bug: chromium:1197965
Change-Id: Ia3063d0909c718ffb8e824225c8c60180551115a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214963
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33700}
Following up on https://webrtc-review.googlesource.com/c/src/+/213000
This CL prevents scheduling work before TaskQueuePacedSender::EnsureStarted(),
making it necessary to function.
Bug: chromium:1152887
Change-Id: I848c9e6d6057a404626ad693b1f4dc7fba797a9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214320
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33695}
This causes build failures in the Chromium fuzzers, so let's disable it
for now.
Bug: none
Change-Id: I0a076c0cd5cfb7d62383d733f3934f8b58f8ad34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215040
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33693}
This is the result of compiling Chromium with
Wtautological-unsigned-zero-compare. For more details, see:
https://chromium-review.googlesource.com/c/chromium/src/+/2802412
Change-Id: I05cec6ae5738036a56beadeaa1dde5189edf0137
Bug: chromium:1195670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213783
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33689}
The rename ensures we don't confuse this implementation with
the new one based on the new dcSCTP library.
Bug: webrtc:12614
No-Presubmit: True
Change-Id: Ida08659bbea9c98aba8247d4368799ff7dd18729
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214482
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33688}
When migrating to use StrongAlias types, the PPID was incorrectly
modeled as an uint16_t instead of a uint32_t, as it was prior to using
StrongAlias. Most likely a copy-paste error from StreamID.
As the Data Channel PPIDs are in the range of 51-57, it was never caught
in tests.
Bug: webrtc:12614
Change-Id: I2b61ef7935df1222068e7f4e70fc2aaa532dcf7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214960
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33687}
simplifying the code and comparing against the value libsrtp expects
and increase verbosity of error logging related to key length mismatches.
BUG=None
Change-Id: Icc0d0121d2983e23c95b0f972a5f6cac1d158fd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213146
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33685}
This test flakes due to the expectation at
http://shortn/_XxN4cgzMLD.
Bug: webrtc:12590
Change-Id: Id75ecd4f12cd6f9af86aeb2213fd3cb39aecb6d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33684}
VideoStreamEncoderTest: Remove unneeded set_timestamp_rtp in CreateFrame methods (the timestamp is set based on ntp_time_ms in VideoStreamEncoder::OnFrame).
Bug: none
Change-Id: I6b5531a9ac21cde5dac54df6de9b9d43261e90c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214488
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33683}
The Reassembly Queue receives fragmented messages (DATA or I-DATA
chunks) and - with help of stream reassemblers - will reassemble these
fragments into messages, which will be delivered to the client.
It also handle partial reliability (FORWARD-TSN) and stream resetting.
To avoid a DoS attack vector, where a sender can send fragments in a way
that the reassembly queue will never succeed to reassemble a message and
use all available memory, the ReassemblyQueue has a maximum size.
Bug: webrtc:12614
Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33678}
This class handles the assembly of fragmented received messages (as DATA
chunks) and manage per-stream queues. This class only handles
non-interleaved messages as described in RFC4960, and is not used when
message interleaving is enabled on the association, as described in
RFC8260.
This is also only part of the reassembly - a follow-up change will add
the ReassemblyQueue that handle the other part as well. And an even
further follow-up change will add a "interleaved reassembly stream".
Bug: webrtc:12614
Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33677}
The Data Tracker's purpose is to keep track of all received DATA chunks
and to ACK/NACK that data, by generating SACK chunks reflecting its view
of what has been received and what has been lost.
It also contains logic for _when_ to send the SACKs, as that's different
depending on e.g. packet loss. Generally, SACKs are sent every second
packet on a connection with no packet loss, and can also be sent on a
delayed timer.
In case partial reliability is used, and the transmitter has decided
that some data shouldn't be retransmitted, it will send a FORWARD-TSN
chunk, which this class also handles, by "forgetting" about those
chunks.
Bug: webrtc:12614
Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33676}
Each file is a SCTP packet (without any additional headers), all
extracted from a few Wireshark dumps that have been manually recorded.
Bug: webrtc:12614
Change-Id: I64bef0c563f1d83ae22735d702c8abafec6429b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214701
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33675}
To maintain interoperability between different capturer implementations
this change updates WgcScreenSourceEnumerator to return a list of
device indices instead of a list of HMONITORs, and WgcScreenSource to
accept a device index as the input SourceId. WGC still requires an
HMONITOR to create the capture item, so this change also adds a utility
function GetHmonitorFromDeviceIndex to convert them, as well as new
tests to cover these changes.
Bug: webrtc:12663
Change-Id: Ic29faa0f023ebc26b4276cf29ef3d15d976e8615
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214600
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33673}
This CL fixes a buffer copying issue introduced in this CL:
https://webrtc-review.googlesource.com/c/src/+/196485
In the BasicDesktopFrame::CopyOf function, the src and dst params
were swapped. For me this manifested as a missing cursor when using
Chrome Remote Desktop. I don't know of any other bugs this caused
but I have to assume it affects all callers of the function given
that the copy will never occur.
Bug: chromium:1197210
Change-Id: I076bffbad1d658b1c6f4b0dffea17d339c867bef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214840
Commit-Queue: Joe Downing <joedow@google.com>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33672}
There's been reports of dropped frames that are not counted and
correctly reported by getStats().
If a HW decoder is used and the system is provoked by stressing
the system, I've been able to reproduce this problem. It turns out
that we've missed frames that are dropped because there is no
callback to the Decoded() function.
This CL restructures the code so that dropped frames are counted
even in cases where there's no corresponding callback for some frames.
Bug: webrtc:11229
Change-Id: I0216edba3733399c188649908d459ee86a9093d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214783
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33671}
This fuzzer explores the SCTP parsing, as well as the individual
chunks, as a successfully parsed packet will have its chunks iterated
over and formatted using ToString.
Bug: webrtc:12614
Change-Id: I88f703c5f79e4775a069b1d5439d413870f6a629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214490
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33670}
The SdpOfferAnswerHandler::ssrc_generator_ variable is used from
multiple threads.
Adding thread checks + tests for UniqueNumberGenerator along the way.
Bug: webrtc:12666
Change-Id: Id2973362a27fc1d2c7db60de2ea447d84d18ae3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214702
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33668}
The Data Generator is a testonly library for generating
Data with correct sequence numbers.
Bug: webrtc:12614
Change-Id: Ifc04dfd14d858d905312ffed13e8905c23d59923
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214041
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33667}
Timer is a high-level timer (in contrast to the low-level `Timeout`
class). Timers are started and can be stopped or restarted. When a timer
expires, the provided callback will be triggered.
Timers can be configured to do e.g. exponential backoff when they expire
and how many times they should be automatically restarted.
Bug: webrtc:12614
Change-Id: Id5eddd58dd0af62184b10dd1f98e3e886e3f1d50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213350
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33666}
This change adds support for emitting encoded frames
for recording when the decoder can't easily read out
encoded width and height as is the case for AV1 streams,
in which case the information is buried in OBUs. Downstream
project relies on resolution information being present for key
frames. With the change, VideoReceiveStream2 infers the
resolution from decoded frames, and supplies it in the
RecordableEncodedFrames.
Bug: chromium:1191972
Change-Id: I07beda6526206c80a732976e8e19d3581489b8fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214126
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33662}
It was recommened to me to move this define to the top level BUILD.gn
file to avoid potential issues with the define not being available
where we need it.
Bug: webrtc:9273
Change-Id: Id0e939a51d1e381f684a3ae970569a255f52a5bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214101
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33661}
