e.g all files in the api/test folder not including subdirectories
Bug: webrtc:42226242
Change-Id: I18d74a18f8feec41eb252faa9acfffd1d6f45ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359420
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@{#42773}
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}
Old copy of the header and some previous usage is kept around
for compatibility with downstream projects for now.
Bug: chromium:345101934
Change-Id: Icbe42fb8450d3a4115799438d209da4eda127bab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357441
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42681}
First batch of applying iwyu to the repo.
Done with:
> ./tools_webrtc/iwyu/apply-iwyu api
> git add api/[a-s]*
> python3 gn_autodeps.py ~/local/webrtc/src out/Default
Last step is a custom script I wrote to automatically apply new required
dependencies for target in gn, which saved tons of time manually going
over the files and fixing.
If this is something that interest others, I can submit it as well.
Bug: webrtc:42226242
Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42512}
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}
and files that broke when I fixed the first set.
Bug: webrtc:42226242
Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42429}
Abseil deps are automatically filtered and replaced by the catch-all
Abseil dependency when built within Chromium. This removes the need to
have special handling for Abseil dependencies in our rules and facilitates
tooling.
Bug: webrtc:341803749
Change-Id: I07425e85d9eeccb66ad423fb1db8b59527a0956a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351003
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42365}
with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target.
BUG=webrtc:339300437
Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42362}
This makes the downcasts currently used in eg
modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
safer.
Bug: webrtc:339815768
Change-Id: Ie6c7ab84666d399dbca4c95846b850aac5327f1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42325}
When they are unset, internally same defaults will be created by Environment, i.e., creating them in CreatePeerConnectionFactory is redundant
(https://webrtc.googlesource.com/src/+/refs/heads/main/api/environment/environment_factory.cc#96)
This would simplify changing the default in the future, and demonstrates that by default it is better to leave these dependencies unset.
Bug: webrtc:42226001
Change-Id: I8bf8614b9e8ca4062939e71d45e67cf2da1558c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350742
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42322}
Origina description:
NetworkBehaviorInterface::RegisterDeliveryTimeChangedCallback
adds support for a network behaviour to reschedule next time DequeueDeliverablePackets should be invoked.
SimulatedNetwork::SetConfig(const BuiltInNetworkBehaviorConfig& config,
Timestamp config_update_time)
adds possibility to change the configuration at config_update_time. Delivery time of a packet currently in the narrow section, will depend on the link capacity before and after the update.
Bug: webrtc:14525
Change-Id: Idaf3a4200cfeae0683e1e1d1e98e154119ddf22e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350043
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42253}
This reverts commit 51a70c0d6f.
Reason for revert: Breaks downstream project test.
Original change's description:
> Add more accurate support for changing capacity in SimulatedNetwork
>
> NetworkBehaviorInterface::RegisterDeliveryTimeChangedCallback
> adds support for a network behaviour to reschedule next time DequeueDeliverablePackets should be invoked.
>
> SimulatedNetwork::SetConfig(const BuiltInNetworkBehaviorConfig& config,
> Timestamp config_update_time)
> adds possibility to change the configuration at config_update_time. Delivery time of a packet currently in the narrow section, will depend on the link capacity before and after the update.
>
> Bug: webrtc:14525
> Change-Id: I271251992d05c68f9160bb81811ea8f2efe9c921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349461
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42243}
Bug: webrtc:14525
Change-Id: Iace13b1b4ef21005c9668ff27f6d1ec8f3212baf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349923
Owners-Override: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42245}
NetworkBehaviorInterface::RegisterDeliveryTimeChangedCallback
adds support for a network behaviour to reschedule next time DequeueDeliverablePackets should be invoked.
SimulatedNetwork::SetConfig(const BuiltInNetworkBehaviorConfig& config,
Timestamp config_update_time)
adds possibility to change the configuration at config_update_time. Delivery time of a packet currently in the narrow section, will depend on the link capacity before and after the update.
Bug: webrtc:14525
Change-Id: I271251992d05c68f9160bb81811ea8f2efe9c921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349461
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42243}
Old target and call/simulated.h exist but refer to new target in test/network.
Bug: webrtc:14525
Change-Id: Ida04cef17913f2f829d7e925ae454dc40d5e8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349264
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42191}
Since iwyu is now deprecated, we need to enable use of include-cleaner.
This approach gives some error messages when running, but does the job.
Bug: webrtc:15874
Change-Id: I431deef0f2e5ce99eb256a4d82aa32769ae58b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347642
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42085}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
Left in target are just .cc files with .h files used externally.
Bug: webrtc:14775
Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41835}
Move code from P2PTransportChannel to Candidate, where we set the
foundation value for remote prflx candidates.
Bug: none
Change-Id: I7dbcb85bca35dca7297136b0706092dd8d2b153c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339902
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41774}
Candidate::type() is currently how the name of the type is fetched,
but that getter returns a non-standard type name.
Instead, I'm adding a new getter, type_name(), will follow up with
updating dependent code that needs the string, to use type_name (and
adapt to potential dependency on "local" or "stun") and then switch
type() to be enum based.
Also adding a test file for Candidate with a couple of basic tests to
start with.
Bug: webrtc:15846
Change-Id: I9b78b2405a9f962a3c07eaa8e72a79854c6f5ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41740}
Rework transport_description_factory to only have non-DTLS mode for
testing, and rewrite tests accordingly.
Bug: webrtc:11066, chromium:804275
Change-Id: Ie7d477c4331c975e4e0a3034fbbb749ed9009446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41697}
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}
This completes the breakup of the rtc_p2p target.
Remaining cleanup is to delete the rtc_p2p target and make clients
depend on the base targets.
Bug: webrtc:15796
Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41683}
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/
Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
This CL does:
- Run IWYU on the relevant elements
- Make connection depend on port_interface, not port
- Make port_allocator depend only on port
- Move some constants from port.h into p2p_constants
This allows a dependency graph without ugly groups.
Bug: webrtc:15796
Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41618}
This CL splits many of the source files in p2p:rtc_p2p into individual
compile targets.
One target - connection_and_port - was left with multiple source files
because it was too tangled to detangle at once.
Bug: webrtc:15796
Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41611}
This is a reland of commit 79ac694d9b
Original change's description:
> [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
>
> The moment we've all been waiting for.
>
> Step 1: Add type alias (this CL).
> Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
> Step 3: Delete type alias.
>
> Bug: webrtc:15164
> Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41593}
# Only unrelated bot failures
NOTRY=True
Bug: webrtc:15164
Change-Id: I0f1991409326679a260cb24907808eaa28385350
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41603}
This reverts commit 79ac694d9b.
Reason for revert: Breaks downstream (found another use of
ValueOrDefault instead of value_or)...
Original change's description:
> [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
>
> The moment we've all been waiting for.
>
> Step 1: Add type alias (this CL).
> Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
> Step 3: Delete type alias.
>
> Bug: webrtc:15164
> Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41593}
Bug: webrtc:15164
Change-Id: Ice3f44057b82a7ba9be000d9a0b714152fd07d2f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335701
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41595}
The moment we've all been waiting for.
Step 1: Add type alias (this CL).
Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
Step 3: Delete type alias.
Bug: webrtc:15164
Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41593}
The plan is to replace Members() with Attributes() instead.
For backwards-compatability during the transition, Attribute implements
RTCStatsMemberInterface but the two classes serve the same purpose
which is to allow iterating all metrics of a stats object.
The reason for moving away from "members" is that we already have a way
to express a variable that maybe has a value: absl::optional<T>. The
only information the member adds is the const char* name(), which we'll
move to Attribute in a future CL.
We don't need to maintain an RTCStatsMemberInterface::Type enum in the
future because absl::variant<T> has absl::holds_alternative<T>.
Step 1: Add Attributes().
Step 2: Migrate to Attributes() and delete Members().
Step 3: Replaces all uses of RTCStatsMember<T> with absl::optional<T>
and delete RTCStatsMember + RTCStatsMemberInterface.
Bug: webrtc:15164
Change-Id: I3fdd5b24214bb5cc340a54a0171df73b516e1803
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333840
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41507}
Add ulpfec and flexfec to list of resiliency mechanisms taken
into account and in general exclude Comfort Noise (CN) from media
codecs.
Also introduce RtpCodecCapability::IsMediaCodec & ::IsResiliencyCodec
behaving like the MediaCodec methods.
BUG=webrtc:15396
Change-Id: I79041898928190bfdd33a06d8f6975d7556c46b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330424
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41485}
To allow custom FecController use propagated rather than global field trials
note that there is one FecControllerFactory per peer connection factory,
but FecController is created per peer connection and may use per peer connection field trials.
Bug: webrtc:10335
Change-Id: Id25bfaf4b49d4f6d551730c8fd55596ddc49ab47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41478}
while cleaning up Call factory function,
- pick rtp_transport_controller_send_factory based on presence in the config instead of based on the call site thus removing one extra factory function.
- when Call is created through test helper TimeControllerBasedFactory use original media factory instead of direct factory, thus allow to configure degraded call through field trials in tests, and ensure difference with production code path stay minimal in the future.
Bug: webrtc:15656
Change-Id: If9c2a9fc871e139502db2bec0a241d8d64c53720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330061
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41329}
Replace CallFactory class with a factory function
Bug: webrtc:15574
Change-Id: Ib1d8cff8d7550da3af01693a7bc117a7bd342258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330000
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41321}
