af5ca5af75
Fix potential use after move in RtpVideoStreamReceiver
...
When a frame is assembled `packet_infos` is moved and must be
re-initialized before potentially being used in another iteration of the
loop. Clear `packet_infos` immediately instead of relying on it being
implicitly cleared in the next iteration of the loop.
Bug: None
Change-Id: I954aaa0c6df296cc2a27b3ab496e49fac200f135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238981
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35441}
2021-11-30 11:20:21 +00:00
d58ac5adf8
sdp: reject large number of channels
...
the maximum used in practice is multiopus with
6 or 8 channels. 24 is the maximum number of channels
supported in the audio decoder.
BUG=chromium:1265806
Change-Id: Iba8e3185a1f235b846fed9c154e66fb3983664ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238980
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Philipp Hancke <phancke@nvidia.com >
Cr-Commit-Position: refs/heads/main@{#35440}
2021-11-30 11:01:14 +00:00
6c167d8278
Remove NetEq::Create.
...
This method is no longer useful after a previous refactoring, but it was
not removed from the interface.
Bug: webrtc:13444
Change-Id: I9c4761e8503acdec06c16cc37c2a804d4913eac8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239366
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35439}
2021-11-30 10:31:16 +00:00
0680c4871c
Roll chromium_revision 22cbef27a6..a3e750c9b0 (946167:946315)
...
Change log: 22cbef27a6..a3e750c9b022cbef27a6..a3e750c9b0f8009dc41e..e6e0ee5bae2f9e5b5867..e0d5dbffaad11bb31289..af6e3bb63e42caab826d..69383604c017c7be145e..92767b0febhttps://chromium.googlesource.com/catapult.git/+log/75423c310e..8cf0922154
* src/third_party/depot_tools: 51d99b2f3a..804165b44525c7e11129..0e7a87dbf522cbef27a6..a3e750c9b0https://webrtc-review.googlesource.com/c/src/+/239425
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#35438}
2021-11-30 04:30:11 +00:00
c56645c0cd
Roll chromium_revision 0e21474081..22cbef27a6 (946029:946167)
...
Change log: 0e21474081..22cbef27a60e21474081..22cbef27a61d1c400730..f8009dc41e245517235e..2f9e5b58675bea6e0b34..d11bb31289bebfd47d77..42caab826d51f3bc37e1..17c7be145e02f4003fdb..51d99b2f3a64e26ad3a2..3b45f564e9889d9a924c..b9dad7362169300210d4..25c7e111290e21474081..22cbef27a6https://webrtc-review.googlesource.com/c/src/+/239421
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#35437}
2021-11-29 22:39:01 +00:00
6dd4997950
Fix TestTurnServer to not depend on rtc::Thread::socketserver() accessor
...
Updated all calls to constructor to pass appropriate
rtc::SocketFactory explicitly.
Bug: webrtc:13145
Change-Id: Ibaa764c7eaffbdb8dd6825fd518c54b37316ecb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238807
Auto-Submit: Niels Moller <nisse@webrtc.org >
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35436}
2021-11-29 21:10:31 +00:00
bf7532116b
Create socket server explicitly in PeerConnectionEndToEndBaseTest
...
Bug: webrtc:13145
Change-Id: I674382b3a784d371681426848461502b711dfe6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238430
Auto-Submit: Niels Moller <nisse@webrtc.org >
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35435}
2021-11-29 20:13:01 +00:00
7d6e34ae0b
Roll chromium_revision fe555f2e69..0e21474081 (945923:946029)
...
Change log: fe555f2e69..0e21474081fe555f2e69..0e21474081c8a98d6969..1d1c400730bc9340854d..5bea6e0b346410d12020..bebfd47d7751b895633d..51f3bc37e12ffa1bde79..02f4003fdbcff026d415..64e26ad3a27c7f2ccfb1..69300210d4fe555f2e69..0e21474081https://webrtc-review.googlesource.com/c/src/+/239400
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#35434}
2021-11-29 18:46:00 +00:00
9937c32371
Roll chromium_revision 1ce7d592c3..fe555f2e69 (944520:945923)
...
This CL also adds COM_DECLSPEC_NOTHROW where needed since Windows
is compiled with -std=c++17.
Change log: 1ce7d592c3..fe555f2e691ce7d592c3..fe555f2e69a0e30222a6..c8a98d6969749ecdaeea..245517235ee504863f9e..665b74f7d1038090f742..c936d73ff7d9a982f504..bc9340854d4e888a310b..6410d120207d86ceacba..51b895633dhttps://chromium.googlesource.com/catapult.git/+log/4b9301e9c5..75423c310e
* src/third_party/depot_tools: a29f589a15..2ffa1bde793cabd142ce..cff026d4153e0e32ba30..e2f397893749836d72bd..02959c3ee1ec80f88c5d..13f984c216d5cb19a57d..889d9a924c0e9c2f5abf..7c7f2ccfb11ce7d592c3..fe555f2e69https://webrtc-review.googlesource.com/c/src/+/239355
Reviewed-by: Tommi <tommi@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35433}
2021-11-29 16:23:30 +00:00
6844e51f47
VideoSendStreamTest: Fix flaky UAF.
...
This test being flaky impeded the landing of
https://webrtc-review.googlesource.com/c/src/+/239126 . Fix by
ensuring the test's OnSendRtp guts don't execute past all streams
stopped.
Bug: None
Change-Id: Ie8aefb3bb03c09d2a9514acecd162e7c079c77c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239363
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Markus Handell <handellm@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35432}
2021-11-29 15:58:40 +00:00
ee22543829
Zero-hertz encoding mode: avoid encoder bitrate overshooting.
...
The encoders wrapped in VideoStreamEncoder grossly over-estimates
available bitrate when capture FPS falls close to zero, and frames
re-commence highly frequent delivery. Avoid this by moving the input
RateStatistics inside VSE into the frame cadence adapter, and changing
the reported framerate under zero-hertz encoding mode to always return
the configured max FPS.
Bug: chromium:1255737
Change-Id: Iaa71ef51c0755b12e24e435d86d9562122ed494e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239126
Commit-Queue: Markus Handell <handellm@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35431}
2021-11-29 13:48:00 +00:00
b55c299f4d
Avoid TBR in DEPS autoroller (use CC instead).
...
No-Presubmit: True
No-Try: True
Bug: None
Change-Id: I71c71de4ff9cfb1b9e59620b56a13ff4e94639eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239361
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35430}
2021-11-29 12:54:01 +00:00
e2044664bd
Add unit test of AudioRtpReceiver
...
Bug: chromium:1272566
Change-Id: Ic10c586062cc8fdded2dd6249b49fab527c99823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238987
Commit-Queue: Tony Herre <toprice@chromium.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35429}
2021-11-29 12:18:29 +00:00
2db59a6584
dcsctp: Use rtc::CopyOnWriteBuffer
...
This avoids copying the payload at all. Future CL will change the
transport.
In performance tests, memcpy was visible in the performance profiles
prior to this change.
Bug: webrtc:12943
Change-Id: I507a1a316165db748e73cf0d58c1be62cc76a2d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236346
Commit-Queue: Victor Boivie <boivie@webrtc.org >
Reviewed-by: Florent Castelli <orphis@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35428}
2021-11-29 11:53:19 +00:00
8695282243
Remove unnecessary copy of suspended_ssrcs.
...
Also removing pass-by-value in ctor.
Bug: none
Change-Id: I09e36fd955c8f306c4a347d8befc6eea38384cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239183
Auto-Submit: Tommi <tommi@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Tommi <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35427}
2021-11-29 10:54:30 +00:00
9345bee860
DEPS: use python3
...
Bug: chromium:1208028
Change-Id: I1117855556e4a9e7cbaf54764c8a23c5b8581ded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239160
Auto-Submit: Takuto Ikuta <tikuta@google.com >
Reviewed-by: Jeremy Leconte <jleconte@google.com >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35426}
2021-11-29 07:27:52 +00:00
ef5b21e637
Deprecate and remove usage for WARNING log level
...
Bug: webrtc:13362
Change-Id: Ida112158e4ac5f667e533a0ebfedb400c84df4d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239124
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35425}
2021-11-27 22:21:54 +00:00
150503566c
Delete unused classes Win32SocketServer, Win32Socket, Win32Thread.
...
Bug: webrtc:6424. webrtc:13065
Change-Id: I7f180831d3b01ad6d127b7eec51ecc9b2d8046a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239125
Reviewed-by: Tommi <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35424}
2021-11-26 16:51:56 +00:00
b2860b0033
Update peerconnection example to not use Win32Socket
...
Followup to https://webrtc-review.googlesource.com/c/src/+/223540
Bug: webrtc:6424. webrtc:13065
Change-Id: I85ccfa9e6997a24464d240b0e9b3f1ac90112e8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224541
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Reviewed-by: Tommi <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35423}
2021-11-26 12:24:25 +00:00
deb1b1bc70
Always call IsOk() to ensure audio codec configuration is valid when negotiating.
...
We should avoid creating codecs with invalid parameters, since this can
expose security issues. For many codecs the IsOk() method to check the
codec config is only called in DCHECKs. This CL ensures IsOk() is always
called, also in non-debug builds.
Bug: chromium:1265806
Change-Id: Ibd3c6c65d3bb547cd2603e11808ac40ac693a8b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238801
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35422}
2021-11-26 10:11:21 +00:00
789a0f361f
Delete deprecated RtpExtension::FindHeaderExtensionByUri variant
...
this variant was deprecated 6 month ago in
https://webrtc-review.googlesource.com/c/src/+/219081
with a trivial replacement.
Bug: None
Change-Id: Ib9cd686280edf36da5f39e8e22b6073530837147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238983
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35421}
2021-11-26 07:57:26 +00:00
a40e6de242
Allow extremely low resolution for simulcast path
...
Some screen capturers may occasionally send an extremely small frame,
e.g. 2x2. If a scale_resolution_down_by is specified, WebrtcVideoEngine
would enforce configured resolution to be at least 16x16, which would
then break VideoStreamEncoder and cause a crash.
This changes disables scaling and alignment for extremely low resolutions.
Bug: chromium:1265303, webrtc:13371
Change-Id: Icdb736043e1fdf91fdde5a8e4b3c6a89f6b90577
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236850
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35420}
2021-11-25 19:07:57 +00:00
e1bbef1e6b
Add options to only NACK if there is a valid RTT and if loss rate is below a configured value.
...
Bug: webrtc:10178
Change-Id: I16a74ed6fb380cecaf82a303bb14bf215c944a73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238988
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35419}
2021-11-25 16:30:17 +00:00
144e5bf87d
Use NONE if scalability mode is not specified
...
Bug: none
Change-Id: I8ffdb7fc41dec3c5b37483a6dcbb8fe7f03b59da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238984
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35418}
2021-11-25 13:50:14 +00:00
61a8d9caaa
Call: Deduplicate SentPacket notifications
...
When bundling is in effect, multiple senders may be sharing the same
transport. It means every |sent_packet| will be multiply notified from
different channels, WebRtcVoiceMediaChannel or WebRtcVideoChannel.
Record |last_sent_packet_| to deduplicate redundant notifications to
downstream objects.
This CL reduces 50% PostTask/Wakeup of Dynamic Mode Pacer.
[1] https://datatracker.ietf.org/doc/html/rfc8829#section-4.1.1
[2] https://datatracker.ietf.org/doc/html/rfc8843
Bug: webrtc:13417
Change-Id: Ib121d5af07abe208bd7d36715a234f48cdabb032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238720
Reviewed-by: Markus Handell <handellm@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Tommi <tommi@webrtc.org >
Commit-Queue: Markus Handell <handellm@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35417}
2021-11-25 10:55:04 +00:00
5f34130f26
Declare LERROR deprecated and remove all usage in webrtc
...
Bug: webrtc:13362
Change-Id: I1c6c6eccd950d73be616b34f96db7832ff94377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238804
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35416}
2021-11-24 14:34:24 +00:00
4e8e36c745
Try to fix the version updated bot.
...
No-Presubmit: True
Bug: webrtc:13373
Change-Id: I30418f760cda6be31c75ff7e59311cb59c1874c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238805
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35415}
2021-11-24 13:04:06 +00:00
a018e677f2
Optimize block_delay_buffer.
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Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag.
Bug: None
Change-Id: Ic126bd2d53969a7e9d15e1c1081d5278e27a816c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238664
Reviewed-by: Per Åhgren <peah@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Christian Schuldt <cschuldt@google.com >
Cr-Commit-Position: refs/heads/main@{#35414}
2021-11-24 12:20:28 +00:00
624fb67bbc
Revert "Fix out-of-bounds memory access due to large number of audio channels."
...
This reverts commit 4cbfe4192cd5b8289f7896ce14e0bd8c4ae41a97.
Reason for revert: The fix in this CL is ineffective. A better one has been created here: https://webrtc-review.googlesource.com/c/src/+/238666
Original change's description:
> Fix out-of-bounds memory access due to large number of audio channels.
>
> The number of audio channels can be configured in SDP, and can thus be
> set to arbitrary values by an attacker. This CL fixes an out-of-bounds
> memory access that could occur when the number of channels is set to a
> large number.
>
> Bug: chromium:1265806
> Change-Id: Ic88ff6d85b978b8eb99bf03cc52457a4552e8c24
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237808
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#35354}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1265806
Change-Id: If695ed92f831c2a9631efdf47f1568f5a15c1841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238803
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35413}
2021-11-24 11:45:55 +00:00
c3a1ea1b3f
dcsctp: assert that CreateTimeout returns a usable pointer
...
BUG=None
Change-Id: I97852d5222a4a722e61b450fbe446dd949b6840e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238802
Reviewed-by: Victor Boivie <boivie@webrtc.org >
Commit-Queue: Victor Boivie <boivie@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35412}
2021-11-24 11:44:15 +00:00
a18cad9c09
Add compile-time warning for RTC_LOG(INFO)
...
Bug: webrtc:13362
Change-Id: I3727891bfeb192941f249a74ead5ef38ac8acc2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238800
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35411}
2021-11-23 21:46:38 +00:00
6c3c3db184
Roll chromium_revision 866506397b..1ce7d592c3 (944390:944520)
...
Change log: 866506397b..1ce7d592c3866506397b..1ce7d592c3379395e10f..a0e30222a6368880db83..749ecdaeeaf5aa54daec..d9a982f50483d4511de4..4e888a310b9a1aa8fd33..7d86ceacbahttps://chromium.googlesource.com/catapult.git/+log/7ae3bc4766..4b9301e9c5
* src/tools: 91e4d58d2c..0e9c2f5abf866506397b..1ce7d592c3https://webrtc-review.googlesource.com/c/src/+/238820
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#35410}
2021-11-23 16:35:18 +00:00
b09d87232b
Reland "Add dav1d decoder to WebRTC."
...
This reverts commit 8498b7e7f6b90fa036de2a6887d34256f0565b4f.
Reason for revert: Updating CL to include conditional build flag.
Original change's description:
> Revert "Add dav1d decoder to WebRTC."
>
> This reverts commit 147858577d4db6d257d3cc248fe571a1bbf887e3.
>
> Reason for revert: High binary size increase
>
> Original change's description:
> > Add dav1d decoder to WebRTC.
> >
> > Bug: none
> > Change-Id: I7642f42e592dcf510679f881f118bc4dab93b31c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237504
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org >
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> > Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> > Cr-Commit-Position: refs/heads/main@{#35394}
>
> TBR=danilchap@webrtc.org ,mbonadei@webrtc.org ,ilnik@webrtc.org ,philipel@webrtc.org ,mflodman@webrtc.org ,ssilkin@webrtc.org ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I00a8acd6ea94ce523c2d5ba705333c9174678180
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: none
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238560
> Reviewed-by: Artem Titov <titovartem@webrtc.org >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org >
> Owners-Override: Artem Titov <titovartem@webrtc.org >
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#35395}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: none
Change-Id: Iff51848731646159e87e075c38af7cb6355f5b5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238661
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35409}
2021-11-23 15:59:58 +00:00
707e5a0cd7
Make test framework create portallocator with an explicit PacketSocketFactory.
...
Bug: webrtc:13145
Change-Id: I04575517b1e215a2204611415f728c358c8d64fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238660
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35408}
2021-11-23 15:00:33 +00:00
aad4ba102b
Fix windows bot command line issue.
...
When wrapping test_env.py command on windows bot, there is an error because the python command is missing.
Adding vpython in the command like it is done on chromium mb.py script fixes the problem.
Bug: b/197492097
Change-Id: I91dbad31549fb29058424ca3b2fb8539c3e8010e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Christoffer Jansson <jansson@google.com >
Commit-Queue: Jeremy Leconte <jleconte@google.com >
Cr-Commit-Position: refs/heads/main@{#35407}
2021-11-23 13:56:03 +00:00
45e15e3343
Prepare for migrating to new AddPeer method
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Bug: webrtc:13145
Change-Id: I089d518e55cb8df32ddf3c587f82376226c18e9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238761
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35406}
2021-11-23 13:04:44 +00:00
646a90784e
[DVQA] Remove old constructor from StatsKey
...
Bug: b/205824594
Change-Id: I081ca48978ce498c56f47022e28fa0a66846765f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238667
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35405}
2021-11-23 12:30:02 +00:00
6d057976e9
StatsEndToEndTests: Remove unneeded fake renderer in tests.
...
Bug: none
Change-Id: I1f14fa2e606dde38113560c32640fb6b2996ea14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238428
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Åsa Persson <asapersson@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35404}
2021-11-23 11:10:04 +00:00
f47a724168
New struct PeerNetworkDependencies
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Preparation to make landing of
https://webrtc-review.googlesource.com/c/src/+/238660
easier.
Bug: webrtc:13145
Change-Id: I314a53cc634f842e5df009d0802b214aa6f8728b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238663
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35403}
2021-11-23 08:37:36 +00:00
89f1aa59b6
Roll chromium_revision 0755db8caa..866506397b (944150:944390)
...
Change log: 0755db8caa..866506397b0755db8caa..866506397be704213601..379395e10f5867e704e3..368880db83ee53859d89..f5aa54daec7597ad2c88..83d4511de455fa5116e5..9a1aa8fd33https://boringssl.googlesource.com/boringssl.git/+log/69030a0cea..3a667d10e9
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/8a112e2bdf..7ae3bc4766
* src/third_party/depot_tools: b6c1ed40d5..a29f589a1581912a1385..3cabd142ce715528ea10..d5cb19a57d020ad36eaf..91e4d58d2c0755db8caa..866506397bhttps://webrtc-review.googlesource.com/c/src/+/238700
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#35402}
2021-11-23 06:40:06 +00:00
f05f2823b3
Allow setting volume on unstarted AudioRtpReceivers
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This was the behaviour before https://webrtc-review.googlesource.com/c/src/+/218605 ,
and is currently relied upon by Chrome to mute received audio tracks
by default, until they should be played out.
Bug: chromium:1272566
Change-Id: I8a288a287e7c01392f4af1db5b083e8d7ee7b2a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238665
Commit-Queue: Tony Herre <toprice@chromium.org >
Reviewed-by: Tommi <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35401}
2021-11-22 18:58:56 +00:00
a9730e5292
Roll chromium_revision 2b49f0a621..0755db8caa (944047:944150)
...
Change log: 2b49f0a621..0755db8caa2b49f0a621..0755db8caab53fedf715..e704213601b8e6a02717..5867e704e3ad4a582ec6..ee53859d894f31c40149..7597ad2c88a57f1dde5f..55fa5116e503cb3fd2f9..715528ea10ff95136344..020ad36eaf2b49f0a621..0755db8caahttps://webrtc-review.googlesource.com/c/src/+/238682
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#35400}
2021-11-22 18:31:17 +00:00
8ef7da78b3
[DVQA] Remove sender from DVQA StatsKey
...
Sender isn't actually require to identify the stream, so specifying it
every time is useless. This CL removes sender from StatsKey object and
introduces StreamsInfo object which contains all required metadata about
streams that are seen by DVQA.
Bug: b/205824594
Change-Id: I5b6be3865a30fd5980ff6e7e50906abe70a632ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238562
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35399}
2021-11-22 14:42:13 +00:00
f002e2fc90
Roll chromium_revision 40bb827aa5..2b49f0a621 (943489:944047)
...
Change log: 40bb827aa5..2b49f0a62140bb827aa5..2b49f0a62154b32958be..b53fedf71509c3e212be..b8e6a02717a63bbc1389..e504863f9ed9a147a20b..ad4a582ec62f79556c57..4f31c401494732b817fb..a57f1dde5fhttps://chromium.googlesource.com/catapult.git/+log/153af62da5..8a112e2bdf
* src/third_party/freetype/src: 9079c5d91a..81912a1385514e743cbf..03cb3fd2f9299bc80685..ff9513634440bb827aa5..2b49f0a621https://webrtc-review.googlesource.com/c/src/+/238680
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#35398}
2021-11-22 14:31:16 +00:00
520b365163
Fix WebRTC version updater bot.
...
Bug: webrtc:13373
Change-Id: If6e1cf9d6e62284b3ca496199b3bba5b8248302d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238561
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35397}
2021-11-22 08:44:15 +00:00
984cf9b837
Explicitly set encoder and decoder format in codec tests.
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This allows to differentiate and test codecs of the same type but
different implementations/settings.
Bug: none
Change-Id: I74f799b36411e63387513133ffc19a7f0c45d550
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238165
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35396}
2021-11-22 08:18:25 +00:00
8498b7e7f6
Revert "Add dav1d decoder to WebRTC."
...
This reverts commit 147858577d4db6d257d3cc248fe571a1bbf887e3.
Reason for revert: High binary size increase
Original change's description:
> Add dav1d decoder to WebRTC.
>
> Bug: none
> Change-Id: I7642f42e592dcf510679f881f118bc4dab93b31c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237504
> Commit-Queue: Philip Eliasson <philipel@webrtc.org >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#35394}
TBR=danilchap@webrtc.org ,mbonadei@webrtc.org ,ilnik@webrtc.org ,philipel@webrtc.org ,mflodman@webrtc.org ,ssilkin@webrtc.org ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I00a8acd6ea94ce523c2d5ba705333c9174678180
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238560
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org >
Owners-Override: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35395}
2021-11-19 18:47:42 +00:00
147858577d
Add dav1d decoder to WebRTC.
...
Bug: none
Change-Id: I7642f42e592dcf510679f881f118bc4dab93b31c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237504
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35394}
2021-11-19 15:03:12 +00:00
ae47cf7dc6
Optimize suppression_filter.
...
Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag.
Bug: None
Change-Id: I7cde835161e2d3e85fc7c919556fa9a9e87ef6df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238169
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Per Åhgren <peah@webrtc.org >
Commit-Queue: Christian Schuldt <cschuldt@google.com >
Cr-Commit-Position: refs/heads/main@{#35393}
2021-11-19 14:17:31 +00:00
ad3551af7a
Roll chromium_revision 20e23fa31a..40bb827aa5 (943065:943489)
...
Change log: 20e23fa31a..40bb827aa520e23fa31a..40bb827aa54f53b8db94..54b32958be70b834d169..09c3e212be81212d29a2..a63bbc13897f068e554e..038090f742519eeb6520..d9a147a20b62352386ee..2f79556c575f7bdd99df..4732b817fbhttps://chromium.googlesource.com/catapult.git/+log/e6e1eb6895..153af62da5
* src/third_party/depot_tools: 1c7dec337d..b6c1ed40d5e4f7673e46..9079c5d91a1a3e2a265f..3e0e32ba30d3ba6ae0c0..bf66d373ae19fb767c32..514e743cbf215804b665..299bc8068520e23fa31a..40bb827aa5https://webrtc-review.googlesource.com/c/src/+/238429
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Christoffer Jansson <jansson@google.com >
Reviewed-by: Christoffer Jansson <jansson@google.com >
Cr-Commit-Position: refs/heads/main@{#35392}
2021-11-19 14:01:43 +00:00
