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}
also fix a few TODOs
Bug: None
Change-Id: I2d287ed1a58f71ef32d5dc5624879ae8c63348b5
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370122
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43485}
and include explanation of source access.
Bug: chromium:362397798
Change-Id: I7af673ffe060507b0e9dea95d650ffb0a681727c
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43484}
Rewrites some of the logic to better takine account RTX padding and
potential acking from transport cc. This should make it both more
robust and a bit faster.
Bug: webrtc:381216373
Change-Id: I1a395c6bd86445ba3e63d79bdc766c7ff582e2a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370060
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43483}
For an offer in sendrecv direction, if for example it can send H.265
level 5.2 while receiving 6.0, setCodecPreferences on offerer's transceiver will currently remove H.265 from the offer SDP, since currently we do a precise level match on send_recv_codecs with the codecs from setCodecPreferences.
Update the matching logic to ignore the level when matching.
Bug: chromium:41480904
Change-Id: Id0f89cbf117ce62249a99257dcce18b35f407cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369960
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43477}
Cases above 100 ms have been observed on mac; use 60 seconds as
an offset.
Bug: webrtc:380712819
Change-Id: I52a085cb196472188bb5493276a1b32524717c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369881
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43473}
Two fixes to deflake,
1. Increase the ramp up time for all layers - short time was flaky for
720p.
2. Wait for both the scalability mode AND implementation name to update.
Sometimes the implementation name would change before the scalability
mode did due to a race, so some OutboundRtpStats would have the wrong
values.
To achieve #2 (and #1 with some debugging) a new utility
WaitForCondition was added in order to apply matchers to a condition.
This is used instead of EXPECT_WAIT_EQ and similar because it gives
clear feedback on failure.
I have made 500 runs without a further flake.
Bug: webrtc:381216372
Change-Id: I0132377774e379857664e9a0c20f432bc9dc9fb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369742
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43472}
When an enum has AbslStringify, we log the text coming from stringifying it, not the numeric value.
Drive by changes,
1. Changed the tests to use string matchers rather than
std::string::find.
2. Fixed test includes.
3. Fix spelling.
Bug: webrtc:381502973
Change-Id: I6bab0afda1b20d72c02629e80ff2ac567650183a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369861
Auto-Submit: Evan Shrubsole <eshr@google.com>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43471}
A few tests in dcsctp_socket_test was named DcSctpSocketResendInitTest
instead of DcSctpSocketTest. There is no reason for that.
Bug: None
Change-Id: I845eb0ab6150c4e5d457307e12f056486f86e468
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43470}
The next expected MID to use (which applies to both ordered and
unordered streams, in contrast to SSNs) was properly handed over for
streams this socket sends on, but not for streams this socket receives
on.
Adding handover state first.
Bug: webrtc:41481008
Change-Id: Ib3941f0ee1a34c24605792d9f0b658bb6a9ade4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369821
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43469}
`VideoStreamEncoder` should not recreate the
`FrameInstrumentationGenerator` instace unless the encoder is actually
released. Otherwise it will restart and expect a keyframe the encoder
will likely not produce for a while.
Bug: webrtc:358039777
Change-Id: I111149d5e9b632df9eeb88bbbe8a07969c3e3f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369740
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43468}
Previous computation assumed that local clock is UTC. It isn't.
Adding integration test for abs-capture stats.
Bug: webrtc:380712819
Change-Id: I054d61984cbd017b7ad04ab13e5a687eab89db69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43465}
~BitstreamReader() DCHECKs that the last read has been verified, so all
paths where we may leave the slice_reader instance's scope early must be
guarded by an Ok().
Bug: None
Change-Id: Ic67f87c04d1f042392c1dd6a066fdccf26e19003
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369540
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43463}
Return default value T() if no values have been added to the filter.
Together with
https://webrtc-review.googlesource.com/c/src/+/369440, DataRate etc can be used by the filter.
Bug: None
Change-Id: I3d0e1a3e698a91a6197bf434ace2ff8246dc393e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43461}
0 is natural default value for types that can be accumulated
Having default constructor simplify usage of these types in templated code.
Bug: None
Change-Id: If005c69018a2a11011bc789502fdbc600cad3278
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43460}
