When a value is set in RtpEncodingParameters::codec, the corresponding
payload_type will be set in the SDP a=rid: line.
a=rtpmap:96 VP8/90000
...
a=rtpmap:97 VP9/90000
...
a=rid:r0 send pt=96
a=rid:r1 send pt=97
Bug: webrtc:362277533
Change-Id: Ia9688a5fc83c53cf46621d97e87f8dd363a4d7f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43049}
This adds payload types to the codecs at the time when offer
is being generated, if they are unassigned at that point.
Bug: webrtc:360058654
Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43033}
When it waits for only one frame, the test is flaky.
When it waits for two frames, it is not.
# Relying on triviality for confidence due to purple bots atm,
# see b/367211396
NOTRY=True
NOPRESUBMIT=True
Bug: webrtc:367205682, webrtc:42220900
Change-Id: I14963b7a86961f438fd511aba8f29525e1f19750
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362583
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43025}
which may show useful debug logging.
Also document that we need to forward-declare the internal srtp_ctx_
struct instead of srtp_t.
BUG=webrtc:361372443
Change-Id: I76b1a4fb385af0fc1532f0ce6d0692b804f003dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360182
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43022}
This API should not modify the aspect ratio of the frame, e.g. if the
frame is 1280x720 and requested_resolution is 1280x360, the result
should be 640x360, not a streched out 1280x360 frame. The spec version
of this API calls this "maxWidth" and "maxHeight" which is the right
way to think about it rather than a forced width and height.
VideoAdapter continues to be used to apply adaptation restrictions, but
we now make sure to calculate the correct frame size BEFORE applying
restrictions. Prior to this CL, the VideoAdapter was also used to apply
requested_resolution restrictions. This is actually wrong and would
cause strange scaling factors in some cases, e.g. f=1280x720 + r=720x405
would result in 640x360 instead of 720x405. Now we make f=720x405 first
and only adjust further if restrictions or alignments require us to.
Since this is a change in behavior a WebRtcVideoChannelTest is updated.
Encodings integration test is also added, both for aspect ratio (new
behavior) and orientation agnosticism (old behavior still passing).
Bug: webrtc:366067962
Change-Id: I4e8dc27da5a84d73238b8ab74ef197eb5ee8072a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362101
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43020}
This allows detecting if it has been set reliably.
0 is a valid payload type.
Bug: webrtc:360058654
Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43016}
This CL makes `requested_resolution`, which is the C++ name for what
the spec calls scaleResolutionDownTo, align with the latest PR[1].
The PR says to ignore scaleResolutionDownBy when scaleResolutionDownTo
is specified as to be backwards compatible with scaleResolutionDownBy's
default scaling factors (e.g. 4:2:1). Ignoring is different than what
the code does today which is to throw an InvalidModificationError.
We don't want to throw or else get+setParameters() would throw by
default due to 4:2:1 defaults so the app would have to remember to
delete these attributes every time even though it never specified them
(Chrome has a bug here but fixing that would expose this problem, see
https://crbug.com/344943229).
[1] https://github.com/w3c/webrtc-extensions/pull/221
Bug: none
Change-Id: I21165c9b9f9ee7259d88b89f9ae58b862ea4521e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362260
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43002}
* 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}
This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code.
Bug: webrtc:360058654
Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42989}
Disable the checks ensuring we reject mixed-codec simulcast
when the field trial is enabled.
The feature is however not yet implemented.
Bug: webrtc:362277533
Change-Id: Ib1601767c951d61aaa37a3d8767d0a81444d626c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361404
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42942}
Since media/ and pc/ both have to use this, and both
depend on call/, this seems to be the right place to put it.
Also factor out the interface that media will use in a separate
interface class.
Bug: webrtc:360058654
Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42933}
Will be later used to conditionally enable mixed codec simulcast
with a field trial.
Bug: webrtc:42220378
Change-Id: I527a488c04cd2b5a9f4ec703504b67943e966ab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361403
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42929}
which broke since libsrtp included openssl/srtp.h instead of
its own srtp.h due to the order of include directories
BUG=webrtc:42234521
Change-Id: Idc5cba2114febd1e0835d201b6c23424a88e62d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360705
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42913}
- Modify munger to take (mutable)
std::unique_ptr<SessionDescriptionInterface> rather than
cricket::SessionDescription (that latter is embedded in the former)
- For all pranswer test cases, do a final SetRemoteDescription(kAnswer) and
check that signaling_state == stable
Add new test cases:
1) A test case that only applies it as prAnswer on caller (callee is stable)
2) A test case that "scramble" sdb between prAnswer and Anser.
Bug: None
Change-Id: Ifedd92ade01ae781a2e59d0569133c486c7093fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360781
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42891}
When declaring a lambda with a value-capture default `[=, ...]`, the
this pointer is implicitly captured by value as well. This results
in potentially-unintuitive behavior and has been deprecated in C++20.
It produces a warning in newer versions of clang
(https://reviews.llvm.org/D142639).
Unfortunately, the preferred C++20 pattern `[=, this, ...]` is not compatible with previous C++ versions. To maintain compatibility with C++14, 17, and 20, this CL modifies all lambdas which capture `this` to explicitly capture all the necessary variables, with no capture-default.
Bug: chromium:351004963
Change-Id: I10c4a9669f340efba75a3e4016f0988a2d606d1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357322
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Devon Loehr <dloehr@google.com>
Cr-Commit-Position: refs/heads/main@{#42886}
Normally (scaleResolutionDownBy) restrictions are applied at the source
which changes the input frame size which triggers reconfiguration with
appropriate scaling factors.
But when requested_resolution is used, encoder settings are by
definition not relative to the input frame size. In order for
restrictions to have an effect, they are applied inside
ReconfigureEncoder(): you get the minimum between the requested
resolution and the restricted resolution.
ReconfigureEncoder() happens when you SetParameters(), but the bug
here is that we don't do it again once the restrictions are updated.
So if restrictions are 540p when you ask for 720p, you get 540p and
after restrictions change to unlimited you're still stuck in 540p.
The fix is to also trigger ReconfigureEncoder() inside
OnVideoSourceRestrictionsUpdated() when the restricted resolution is
changing and a requested_resolution is configured.
To ensure reconfiguring the encoder "on the fly" like this does not
reset initial frame dropping logic, InitialFrameDropper caring about
input frame size changing is made conditional on not using
requested_resolution.
# Slow purple bots failing but they are not affected by this change.
NOTRY=True
Bug: webrtc:361477261
Change-Id: I1389aa16cf408b0d14e0b5b6f68c2442db955be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360200
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42882}
Before this cl, ReadyToSend signaled false if sending a packet failed and transport->GetError() returns ECONN.
ECONN may be reported by the TCP connection (TcpConnection) if the remote closed the connection. TcpConnection will attempt to reconnect and should change the writable state if it fail.
Changing the state in the context of sending packets may cause recursive
calls and seems to cause problems with incorrect states.
It is simpler if RtpTransport::SendPacket ignore these failures and
upper layers treat these lost packets similar to if the packets had been
lost via UDP.
For safety, this change can be reverted by field trial WebRTC-SetReadyToSendFalseIfSendFail/Enabled/.
Bug: webrtc:361124449 b/359989715
Change-Id: I8e7016dfb4301862286215c4512aa8ac03a16685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42868}
According to spec, if you ask for three encodings you get three
encodings (duh). But according to legacy code, if you ask for three
encodings AND codec is VP9, then surely you meant a single encoding that
is kSVC where the other encodings influence the scalability mode of the
first encoding.
Standard simulcast support in VP9 was shipped as an opt-in feature where
you have to specify `scalability_mode` and `scale_resolution_down_by` in
order to let WebRTC know that you want to disable the legacy path.
But `scale_resolution_down_by` is not the only way to configure
resolution, there is also the `requested_resolution` code path. This CL
adds standard simulcast support for this code path as well.
Prior to this change, our parameterized test would have passed in VP8
but failed in VP9. With this change the test passes for all codecs.
Bug: webrtc:361124448
Change-Id: Ic5a7136de8abf430813fd01342862775fca145fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42822}
by encryption a packet with sequence number 65535 followed
by a packet with sequence number 1. The second packet is encrypted
with a SRTP ROC of 1 as described in
https://datatracker.ietf.org/doc/html/rfc3711#section-3.3.1
The packets are (received and) decrypted in a different order,
the packet with sequence number 1 (and ROC=1) is decrypted first.
Since the ROC is maintained locally the decrypting session assumes
it to be 0.
Why is that a problem? The RFC recommends estimating the ROC with +-1 which, as demonstrated by the test, libSRTP does not.
But this is a rare problem that requires a random in a high range combined with packet loss/reordering which turns into no-a-problem if you choose carefully as done by packet_sequencer.cc which restricts the initial sequence number in the range 0..32767 which means you do not run into this issue in production.
See also Q6 in libsrtp's historical documentation at
https://srtp.sourceforge.net/historical/faq.html
BUG=webrtc:353565743
Change-Id: I9bd72b198c946937aeb25c229005a0c682447f53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42798}
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}
This requires making CallConfig move-only so it can hold a unique_ptr to
the factory, but as discussed with Danil, that seems fine.
Bug: chromium:355610792
Change-Id: Ie52e33faaa4a2af748daeb25f5327b7a532936e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357862
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@{#42679}
All calls in code under test were migrated to AudioEncoderFactory::Create and thus there is no longer need to propagate older function.
Bug: webrtc:343086059
Change-Id: I9e0ea4024759deb22c0d284e0e4bac7322a08f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357181
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42638}
Some dependencies still exist but are a bit more complex to remove.
This CL removes either unused or easily replaced with ToString()
instances of ostream usage. In one case, moving the operator<<
implementation to the one test file that requires it.
Bug: webrtc:8982
Change-Id: Ia5c840b12a42893494af401317a3daf2fe50ba9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356240
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42582}
Libvpx works without this, so the existing tests pass. However, other
encoder implementations (like rtc_video_encoder in Chrome) look at
different fields and get confused about the configuration.
Test: Integration tests with Chrome and windows hardware encoders.
Bug: webrtc:348342168
Change-Id: Id0d96cff34eb34c7e019a24255623f3aeeca5772
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42555}
