Based on the description, this dependency have no meaningful upstream,
and is maintained inside webrtc.
Marking this dependency's URL to indicate the webrtc's repo is the
canonical repo.
Fixed: chromium:362397270
Change-Id: If6e16a6e34e0083be31d4436fcdfa7c83cd9179a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Jiewei Qian <qjw@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43535}
As of 485f2be7c1, this no longer has any effect; instead, the ABSL_NULLABILITY_COMPATIBLE attribute which is already present on the class determines whether a class is compatible with nullability annotations.
Bug: None
Change-Id: I5aeca86c86c2b6eadb2644695ee3621e92f1f568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43532}
Add SSLProtocolVersion for TLS13 and DTLS13
Allow setting max version to 13 (for BoringSSL)
Don't change any defaults.
This is a NOP.
BUG=webrtc:383141571
Change-Id: I11303c14e8d79c09d9437d44e44003c67d2fc31b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370900
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43530}
There were a few typos in the README file.
Bug: chromium:362397579
Change-Id: Ib0aa84f57f3d83851f085e595ffa72a53ec8311d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43525}
srtpfilter was a SDES thing which is gone.
BUG=None
Change-Id: I060582b5ba9e72d1fdad3662e2b478042f0c780c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370640
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43517}
Fix todo to ensure TransportSequence numbers are generated if CCFB according to RFC 8888 is used. Transport sequence numbers are used in BWE algorithms regardless of feedback format.
Bug: webrtc:42225697
Change-Id: I6eab95c0241d590f6e7a90d19c82d13ab8692f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370341
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43515}
Further, add use of it in libvpx_vp8_encoder and with tuning for keyframes and lower bound of std_dev = 1.25 to work around some edge cases. Plus some minor cleanup.
Bug: webrtc:358039777
Change-Id: I6f624a6a8c7ccfe2fe656e4c089c225296f0264f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370061
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43513}
The purpose is to be able to add more tests that verify that BWE still work and verify ECN behaviour e2e.
Bug: webrtc:42225697
Change-Id: Ie178d29d7870bfa3211d10925d00c621617ddf48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370561
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43511}
This reverts commit 03f56d75d5a4bbbc6b6fe93e119f73c69ff98267.
Reason for revert: Breaks downstream project.
Original change's description:
> Remove stun_prober
>
> The STUN prober shows the old RFC 3489 way of determining the NAT type
> by pinging two different servers. This is known to be faulty as pointed
> out by
> https://datatracker.ietf.org/doc/html/rfc5389#section-2
>
> Chromium dependency removed in
> https://chromium-review.googlesource.com/c/chromium/src/+/6036622
>
> BUG=None
>
> Change-Id: I2b61dfe2ff899ce71ec9d2253dc836c5908cf8c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368182
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43503}
Bug: None
Change-Id: I08d01d4c9d882aca883e1c889aed8bddbca65b91
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370540
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43506}
Set use_default_launcher=false in rtc_test on android
Bug: webrtc:42223878
Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43505}
to be deleted when downstream consumers are upgraded
BUG=webrtc:367395350
Change-Id: I35f1fefdc6535ad443b86176ea600455c2361834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370284
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43502}
for (partially) parsing DTLS packets and extracting the msg_seqs
BUG=webrtc:367395350
Change-Id: Ieb0fc121c6dc82118ced5939c1a9ebe2d72e3cb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370181
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43501}
The video engine MapCodecs function returned an empty list of
codecs when errors occured, which caused crashes downstream.
This created issues with diagnosing errors caused by PT redesign.
Bug: webrtc:360058654
Change-Id: I0b5bdc9f95814ac4cfb99f749075990c3077e7a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370420
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43499}
and misc cleanup
BUG=webrtc:367395350
No-Iwyu: remaining IWYU failure is deep inside gtest which is unrelated to the changes and needs to be investigated separately
Change-Id: I5c2b7a6cc6b15fc5474c55eb98635cb9145b7373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370180
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43498}
Without this, Firefox wasn't passing WPT
webrtc/simulcast/setParameters-maxFramerate.https.html.
The main issue is the SetRates API's RateControlParameters doesn't have
a way to model maxFramerate for simulcast layers.
A long term fix would probably be to represent maxFramerate for all
simulcast layers in RateControlParameters. This change is a short term
fix, and resets the encoder iff a simulcast layer's maxFramerate has
changed, and also differs from the maxFramerate of the codec (passed to
SetRates), which matches the layer with the highest maxFramerate.
Bug: None
Change-Id: I088dda0fe88092fe5a5cc61114e10847f072a87b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370124
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43497}
This test was missing, which made me believe that it wasn't supported as
the handover state only included SSN and not MID. But when adding tests,
I saw that the current implementation used the SSN field to handover the
MID information for ordered streams which is sufficient given the 32 bit
type used for that (SSNs are only 16 bits).
For unordered streams, there is no need to handover any state there are
no expected next MID for unordered streams (they can be received in any
order).
So, adding tests and removing the handover state I just added.
Bug: webrtc:41481008
Change-Id: If1799cb1def5bd9f585a87cff6d835f4a9053b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370121
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43495}
This replaces the WaitUntilCondition function that was used in the
peer_connection_encodings_integrationtest previously. Along with that it
adds tests and improved error message printing.
As a drive-by, matchers were added for RTCError as these are the return
type of this utility function.
Bug: webrtc:381524905
Change-Id: If7ff18692396d3996b5b289f2d2c92520226003e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43494}
The next step of the migration is to use the generated java wrappers
which requires depending on the generated java targets.
Bug: webrtc:353174456
Change-Id: I834da78f9ab6050f3be148f6557252897aa68711
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369781
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mohamed Heikal <mheikal@google.com>
Commit-Queue: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43492}
This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
Reason for revert: Broke internal tests.
Original change's description:
> Use PayloadTypePicker for video PT assignment
>
> This includes changes that change the order of codecs.
> It is preparatory to doing late assignment of video PTs.
>
> Bug: webrtc:360058654
> Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43489}
Bug: webrtc:360058654
Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43490}
This includes changes that change the order of codecs.
It is preparatory to doing late assignment of video PTs.
Bug: webrtc:360058654
Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43489}
For-loop has a 'continue' statement that skips increment of the index.
Added such an increment before 'continue' for the index to keep up with
the for-loop.
I guess current implementation will bug on codecs sequence like:
'red, unknown, opus'
since the subsequent for-loop (the 'red_codec' one) will not be able to
find 'opus'.
Seems like adding second increment statement is the easiest way to fix it.
Bug: None
Change-Id: Iab9cc66cf569458af9fd9ba5b938d83186c78c73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43488}
In addition, avoid empty conversion when no message is present.
Bug: chromium:379326016
Change-Id: I855069fa89a157ba862b5162c56858825ebc1a40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370160
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43487}
