and switch usages to ByteBufferWriter::Write
This is part of getting rid of "pointer + length" arguments.
Bug: webrtc:42225170
Change-Id: I65a9b9550868022c0eb1f63b547195dadfbea678
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377461
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43916}
With this CL, the decision if an RTP packet should be sent as ect(1) is made in RtpControllerSend depending on if RFC 8888 has been negotiated and if CCFB is received with ECN enabled.
Since webrtc does not yet adapt to ECN feedback, packets are sent as ECT(1) until the first feedback is received.
Change-Id: Iddf63849328afbe54a7c8f921f2e8db134aeff6a
Bug: webrtc:42225697
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367388
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43609}
which implements the handshaking logic of the DTLS-STUN piggybacking.
Not wired up yet, split from
https://webrtc-review.googlesource.com/c/src/+/362480
BUG=webrtc:367395350
Change-Id: I9ee8ff17af4ec96fb891d9852ac50825155735a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370679
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43606}
dtls_transport will when detecting a new fingerprint
(e.g by usage of pranswer) signal DtlsTransportState::kNew.
When this happen, the dtls crypto state is lost, and
sctp should reconnect, srtp does this automatically
in current code base.
The existing behavior in dcsctp is that it will detect
peer sending an init, and reconnect. But any messages sent
between the dtls restart and the message arriving from the
peer will be lost.
This patch changes so that this case is gracefully handled by
a) letting dcsctp_transport listen to dtls state
this is big part of patch and involves changing the type of
the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this
into a separate patch...
b) if a dtls restart happens, delete and restart socket.
Testcase that fails before patch and works after is attached.
Bonus: And include-what-you-use on patch
Bug: b/375327137
Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43546}
For now, old file forward include api/transport/ecn_marking.h
Done in preparation for more usage of this enum when handling received
RFC8888 feedback.
Bug: webrtc:42225697
Change-Id: I022c2b7f1e7f986b24aa32b8911ad67c6640a5c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366440
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43282}
Compiling webrtc with `-Werror=unused-parameters` is failling duo to
those parameters.
Also, it shouldn't harm us to put those in comment for code readability as
well.
NOTE: This time I made sure to iterate over the C files in the
audio_processing folder and compile them using gcc.
On the original CL that was reverted - that failed with the same error
Danil mentioned. This time it seems fine.
I'll make sure to run the same script on the rest of my CLs for sanity
Bug: webrtc:370878648
Change-Id: I83cea3a08777e21d26a95bcad503a2d1b74566eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364537
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43249}
Compiling webrtc with `-Werror=unused-parameters` is failling duo to
those parameters.
Also, it shouldn't harm us to put those in comment for code readability as
well.
Bug: webrtc:370878648
Change-Id: I0ab2eafd26e46312e4595f302b92006c9e23d5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364340
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43157}
These functions have been deprecated since October 2022.
Bug: None
Change-Id: I74f51c9d0e8ee340a2043bf43f7a1b0d8b79726e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43118}
* IWYU export <sys/socket.h> from rtc_base/net_helpers.h.
* Add a presubmit check to ensures that <sys/socket.h> is included through net_helpers.h (expect if there is a IWYU pragma or a no-presubmit-check).
* Clean up existing includes of <sys/socket.h>
Change-Id: I4bc6cce045c046287f8f74f89edfc9321293b274
Bug: b/236227627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362082
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42996}
and update some usage to use the "correct" stun attribute names
BUG=webrtc:42229250
Change-Id: If0c34d1d9b399766d7073661ea2a5515100256a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359440
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42810}
The new type PriorityValue is a strong 16-bit integer matching RFC 8831
requirements that can be built from a Priority enum.
The value is now propagated and used by the SCTP transport, but enabling
the feature still requires a field trial for now.
Bug: webrtc:42225365
Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42695}
since it contains helpers mostly related to cryptographically secure random numbers and strings.
BUG=webrtc:339300437
Change-Id: I10db939534b25dc792ac1600a4721d1b84521880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42441}
Repeated initial probes are sent every second until
ProbeController::OnMaxAllocatedBitrate is invoked (Media is beeing sent) or 5s has passed.
Each probe has a duration of 100ms, sent in packet bursts every 20ms.
ProbeController::waiting_for_initial_probe_result_ is no longer needed
and is removed.
Setting field trial for duration between probe packets bursts are moved
from BitrateProber to ProbeController.
Bug: webrtc:14928
Change-Id: I3170533f679fc2cc2aa5402e248fa1f6996d3ccd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350640
Reviewed-by: Diep Bui <diepbp@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42323}
This is a reland of commit 47bfe39ecfe45b2f94c616ace97949003d9e87b4
Original change's description:
> Split digest methods from ssl target into digest target
>
> in an attempt to break up the monolithic ssl target.
>
> BUG=None
>
> Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#42249}
Bug: webrtc:339300437
Change-Id: I31bb79bbc6cc55a2634176f95ec67de195974e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42304}
Environment guarantees field trials are provided, thus GoogCcNetworkController doesn't need to fallback to the global field trials.
Bug: webrtc:42220378
Change-Id: Iff8e00504b43b074dc41b5ac9908fd0e2be18959
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350540
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42300}
This would allow network controllers, GoogCcNetworkController in particular to have access to Environment at construction and thus it can rely on propagated field trials and won't need to fallback to the global field trial string
Bug: webrtc:42220378
Change-Id: I36099522e3866a89a8c8d6303da03f7d5b1cad8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350201
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42260}
This adds a new mode to the ProbeController that sends probes every
second the first 5seconds. Intented usage is before normal media has
started flowing. If enabled, the max allocated bitrat is ignored during
these initial probes.
The purpose is to have a more accurate estimate at the beginning of a
call.
The cl also removes ProbeController::SetFirstProbeToMaxBitrate.
Bug: webrtc:14928
Change-Id: I04feefb2f1b82ff38b35ee2be810f9119c53536a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349924
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42252}
in an attempt to break up the monolithic ssl target.
BUG=None
Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42249}
This reverts commit 501c4f37bfee47b26999ee291c5355ad64554df7.
Patch set 1 contains pure reland.
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.
Bug: webrtc:14928
Change-Id: I6a8660da20ac54237f04a29461e03b31bd988bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347643
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@google.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42086}
This reverts commit 33cc83595ae7dd144c57c614fb62d286d9d7bf68.
Reason for revert: Perf bots showed that this cl cause a change in metrics. It looks like it is for the better, but we want this to be behind a field trial.
Original change's description:
> Ignore allocated bitrate during initial exponential BWE.
>
> The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
> That is the, initial probe will try to probe up to the max configured bitrate.
>
> ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
> continue up to the max configured bitrate, regardless of of the max
> allocated bitrate.
>
> Bug: webrtc:14928
> Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42049}
Bug: webrtc:14928
Change-Id: I56ba58560b6857b6069552c02df822691f7af64d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347622
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42081}
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.
ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
continue up to the max configured bitrate, regardless of of the max
allocated bitrate.
Bug: webrtc:14928
Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42049}
It is a public interface and must be visible to allow tests to include the header file.
Bug: none
Change-Id: I4e6322c622f62c018b274b751e2c395eed7816e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346520
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42027}
This code was extracted to make the next following CL easier to review.
This CL simply exposes the getters, setters and callbacks to set the
buffered amount low threshold on a specific SCTP stream. It will be
used in a follow-up CL, but is just boilerplate.
Bug: chromium:40072842
Change-Id: Iccd72208b369ddc252cc5886f6446b9c2ceeb0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343360
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41943}
Before this change, calling buffered_amount only included what was
buffered on top of what was already buffered in the SCTP socket. With
the defaults, the SCTP socket can buffer up to 2MB of data (that is not
put on the wire) before the additional external bufferering in
SctpDataChannel will be used. The buffering that I am working on
removing completely.
Until it's removed completely, to avoid the issue reported in
crbug.com/41221056, include the bytes buffered in the SCTP socket to
what is returned when calling RTCDataChannel::buffered_amount.
This means that when this value is zero, it can be safe to know that all
bytes have been sent, but not necessarily acknowledged. And calling
close will not discard any messages.
This is a stopgap solution, but as functional as the proper solution
that removes all additional buffering. Follow-up CLs will merely improve
this solution.
Bug: chromium:41221056
Change-Id: I06edd52188d3bf13a17827381a15a4730722685a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41898}
The DcSctpTransport will soon use field trials to conditionally enable
some options.
And overall, there is a migration project to start using the Environment
and this CL is in that direction, also setting the boundary; The dcSCTP
library should not depend on it. But the transport is allowed to.
Bug: webrtc:14997
Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41872}
Using the Api, BWE components are recreated and new settings can be
applied. Initially, the only configuration available is allowing BWE probes without media".
Note that BWE components are created when transport first becomes writable. So calling this method before a PeerConnection is connected is cheap and only changes configuration.
Integration test in https://webrtc-review.googlesource.com/c/src/+/337322
Bug: webrtc:14928
Change-Id: If2c848489bf94a1f7a5ebf90d2886d90c202c7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41687}
ReadStringView() is a simple alternative to ReadString() but doesn't
involve a heap allocation for a new std::string.
Using the new methods in one place to start with.
Bug: none
Change-Id: I1100c6d258ffb4c8a31a46ba88a7f8bff9cf35cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332120
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41533}
This completes the conversion of ByteBufferReader and ByteBufferWriter
to uint8_t.
No-Try: True
Bug: webrtc:15661
Change-Id: I4152a8a4fd2462282d4107b3c2eed19acc8b29b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331640
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41403}
and make follow-on changes.
Bug: webrtc:15665
Change-Id: Ice646f88ba5a09d6a8d9ce70415d8a14d7050d3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329781
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41393}
Also make it more convenient to use uint8_t array view for
interfacing to the class.
Bug: webrtc:15665
Change-Id: Ib671b5add79a48004133a6ecd99429534f7de1de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328140
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41212}
