1633 Commits

Author SHA1 Message Date
Artem Titov
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-07 00:54:26 +00:00
Artem Titov
895069045f Fix: IvfFrameGenerator won't decode frame on release build
Bug: webrtc:10138
Change-Id: Id0a6328da20bbb841ed3cb013a0d96d8d88c0152
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161446
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30029}
2019-12-06 15:10:49 +00:00
Ivo Creusen
1518fd34d8 Add support for setting a custom NetEqFactory in PeerConnection level tests.
This allows running Peerconnection level tests with a custom NetEqFactory.

Bug: webrtc:11005
Change-Id: If3063cf61a6274a137e4ab74f9ec2665425f21ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161307
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30028}
2019-12-06 12:34:02 +00:00
Sebastian Jansson
cec2433c47 Exposing more features in the network emulation manager API.
Bug: webrtc:9883
Change-Id: I2a687b46e3374db0dd08b0c02dfea1482e6fb89f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161229
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30024}
2019-12-06 08:47:19 +00:00
Minyue Li
cae277959b Introduce InbandComfortNoise RTP header extension.
BUG: webrtc:11085
Change-Id: I9b556a0d67d3c239abc247787103af9e50af4e65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159710
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30014}
2019-12-05 13:35:01 +00:00
Artem Titov
0020226e63 Replace VideoSourceInterface with FrameGeneratorInterface in AddVideoConfig
Replace VideoSourceInterface with FrameGeneratorInterface in
AddVideoConfig in PC quality test fixture.

Bug: webrtc:10138
Change-Id: I6e5fe91d286e0360bfcad1785af1fb1d8f890563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161239
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30012}
2019-12-05 10:02:22 +00:00
Artem Titov
503d7237ce Introduce FrameGeneratorInterface
Introduce FrameGeneratorInterface to make FrameGenerator API available
for downstream projects.

Bug: webrtc:10138
Change-Id: I4216775e4b8b54c3f1c72d67ffbda31eb082fd7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161234
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30009}
2019-12-04 21:09:57 +00:00
Sebastian Jansson
340af975e9 Always enter yield policy scope using simulated TimeControllers.
This makes the class easier to use at a minor cost of making it slightly
more magic.

Bug: webrtc:9883
Change-Id: If807cfbf046615333c3bcd3b58a001813102a9f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161231
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30008}
2019-12-04 17:16:32 +00:00
Danil Chapovalov
242a9e0ffe Fuzz RtpPacketizerAv1
Bug: webrtc:11042
Change-Id: Id44699395f6dee9cb3bde84c936573b65ad0d848
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161009
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30007}
2019-12-04 13:52:51 +00:00
Eldar Rello
0095d37137 Replace hostCandidate with address and port in RTCPeerConnectionIceErrorEvent
Bug: chromium:1013564
Change-Id: Ie1bb86ed6a2a7d73fe6ee666f973d809ed05a7ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#30004}
2019-12-04 13:18:22 +00:00
Ilya Nikolaevskiy
648b9d77c7 Implement automatic animation detection in VideoStreamEncoder
If WebRTC-AutomaticAnimationDetectionScreenshare experiment is enabled,
content type is screenshare and degradation preference is BALANCED,
then input resolution is restricted if update_rect of the incoming frames
is the same for considerable amount of time and is big enough.

This entails treating BALANCED degradation preference for screenshare as
MAINTAIN_RESOLUTION in adaptation logic.

Bug: webrtc:11058
Change-Id: I903dddf53fcbd7c8eac6c5b1447225b15fd8fe5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161097
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30002}
2019-12-04 11:24:31 +00:00
Mirko Bonadei
a9ad36f322 Fix aec3_fuzzer chromium build config.
Dependencies need to use relative paths in order to work in Chromium,
see [1].

[1] - https://ci.chromium.org/p/chromium/builders/try/linux-libfuzzer-asan-rel/334174

TBR: saza@webrtc.org
Bug: None
Change-Id: I50c401e5983fbb501d1da2ad909198261a8cb940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161300
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30000}
2019-12-04 10:45:52 +00:00
Sebastian Jansson
41462d58b2 Always keep abs send time extension.
This makes the WebRTC-KeepAbsSendTimeExtension field trial
always enabled. This means that we no longer avoid sending the
abs-send-time extension if we have negotiated sending of transport
wide sequence numbers.

The field trial WebRTC-FilterAbsSendTimeExtension is introduced to allow
reverting to the previous behavior.

Bug: webrtc:10234
Change-Id: Ifd9761d84dd1fe79af840f98ad0882a2e5adf0b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29999}
2019-12-04 09:49:04 +00:00
Markus Handell
486cc55a02 TimeController: Rename Sleep to AdvanceTime.
This change renames TimeController's Sleep method to AdvanceTime, unifying
the same name with the same semantic as for downstream projects.

Bug: webrtc:11154
Change-Id: Id79bcf0eafcd0b47a76407ba220479d84df5a736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161092
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29989}
2019-12-03 16:08:54 +00:00
Danil Chapovalov
e3c4884b76 in PacketBuffer::Packet pass payload using smart buffer
Together with RtpDepacketizer refactoring that would reduce
number of memcpy while handling an rtp packet

Bug: webrtc:11152
Change-Id: I6f4e09c93af5e2a9314967a15eac8ced57ec712e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161087
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29985}
2019-12-03 14:55:54 +00:00
Erik Språng
014dd3c9f7 Trials should always be populated in call config.
The trials are always set when a Call instead is created by a
CallFactory, but a lot of unit tests creates instances directly.
This CL updates those call site. There is still a fallback in place
in RtpTransportControllerSend, since there are down-stream usages that
need to be clean up. After that, we'll remove the fallback.

Bug: webrtc:10809
Change-Id: I0aacf0473317bcd64252dd43d93c42de730e2ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160408
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29978}
2019-12-03 10:34:55 +00:00
Artem Titov
5256d8bc4b Refactor FrameGenerator to return VideoFrameBuffer with VideoFrame::UpdateRect
Bug: webrtc:10138
Change-Id: I22079e2630bb1f3bb27472795fe923f9143b3401
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161010
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29973}
2019-12-02 17:11:37 +00:00
Sam Zackrisson
b0db98cf06 Fuzz AEC3
This fuzzer fuzzes AEC3 with the default configuration and variable sample rates and channel counts.

Bug: None
Change-Id: I0d178a320b75fc4cc389657fa2b99931f359b517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160646
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29967}
2019-12-02 09:16:51 +00:00
Sebastian Jansson
efa3f76b08 Moves SampleStats and EventRateCounter to rtc_base/numerics
Bug: webrtc:9883
Change-Id: I53934c86cad3b7cd60bba6c78e5db66c10e5d56a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159821
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29965}
2019-12-02 07:46:48 +00:00
Artem Titov
9dc209a23a Add ability to disable detailed error message in RTC_CHECKs
Bug: webrtc:11133
Change-Id: I989654f1fb97b476a17956d69ee374406439ea8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160653
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29952}
2019-11-28 17:51:00 +00:00
Danil Chapovalov
5314b13a8d Fix undefined-shift in RtpDepacketizerAv1::AssembleFrame
Bug: chromium:1028348
Change-Id: I824e84138acbf4e73fc21ee8248e29e5cc7a0ba0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160643
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29945}
2019-11-28 11:27:33 +00:00
Artem Titov
8861d02f89 Restore tests that were accidently deleted during refactoring
Bug: None
Change-Id: Ic68282d9879581c2d7d42d5d80c876e252e53c37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160920
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29934}
2019-11-27 13:42:58 +00:00
Danil Chapovalov
dc36829db0 Add VideoCodecType::kVideoCodecAV1 value
Bug: webrtc:11042
Change-Id: I3c5151c9e47679760f8f7d79270488fa8f4c7db5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159282
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29927}
2019-11-27 10:18:45 +00:00
Per Åhgren
e14cb99408 Correct/update the activation of the multi-channel processing in APM
This CL removes the experimental status of the multi-channel processing
in APM, and accordingly updates the variable naming.

It also splits the activation of multi-channel processing to be separate
for render and capture.


Bug: webrtc:10859
Change-Id: I0e5d04dcb94b6637c33d97146231b8ddddbaea39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29926}
2019-11-27 10:15:25 +00:00
Bjorn A Mellem
7a9a092708 Delete media transport integration.
MediaTransport is deprecated and the code is unused.

No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
2019-11-26 19:19:36 +00:00
Sam Zackrisson
2b4bd97517 Fix fuzzer-found bug in fuzzer
The fuzzer itself does not call the APM API correctly.

Bug: chromium:1028614
Change-Id: If94808e6d3a1f550c1001b081321e87448e4be49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160703
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29922}
2019-11-26 18:34:06 +00:00
Per Åhgren
27bd76bcb2 DCHECKing for deprecated 8kHz support in AGC and changing fuzzer
This CL adds a DCHECK for the deprecated 8 kHz rate in APM.
It also updates the agc fuzzer code to properly do band-split on
the signals, and not send 8 kHz signals into the AGC.

Bug: chromium:1028092,chromium:1028172
Change-Id: I1e7c8d721834310e94b0e21efea07f75da837cab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160600
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29914}
2019-11-26 11:10:54 +00:00
Artem Titov
a101a4f186 Reland "Add IvfVideoFrameGenerator"
This is a reland of 712a26f3842b4eba1f38c3ba7371b1cf771fd232

Original change's description:
> Add IvfVideoFrameGenerator
> 
> Bug: webrtc:10138
> Change-Id: Iea590f334d22fb7d22077c9bdd3b5ba79691df2e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160185
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29902}

Bug: webrtc:10138
Change-Id: If522d079f0a1e30d6f2b330792aa1d1fc043b8b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160418
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29913}
2019-11-26 09:56:47 +00:00
Steve Anton
13ea34f305 Revert "Add IvfVideoFrameGenerator"
This reverts commit 712a26f3842b4eba1f38c3ba7371b1cf771fd232.

Reason for revert: consistently failing on iOS64 Debug: https://ci.chromium.org/p/webrtc-internal/builders/ci/iOS64%20Debug/20119

Original change's description:
> Add IvfVideoFrameGenerator
> 
> Bug: webrtc:10138
> Change-Id: Iea590f334d22fb7d22077c9bdd3b5ba79691df2e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160185
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29902}

TBR=ilnik@webrtc.org,titovartem@webrtc.org

Change-Id: Ie34e254a7a4ff5ff8fdab7c6b3212792b52b6f53
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160560
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29910}
2019-11-25 21:58:02 +00:00
Artem Titov
712a26f384 Add IvfVideoFrameGenerator
Bug: webrtc:10138
Change-Id: Iea590f334d22fb7d22077c9bdd3b5ba79691df2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160185
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29902}
2019-11-25 13:38:09 +00:00
Artem Titov
2aaf4afb09 Semiautomatic attempt to clean test/BUILD.gn deps
Bug: None
Change-Id: I79e9c2343000abe097e1029c0233a746e43601b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160300
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29895}
2019-11-25 11:00:57 +00:00
Danil Chapovalov
429d8fe28b Add fuzzer test for RtpDepacketizerAv1::AssembleFrame function
Bug: webrtc:11042
Change-Id: If5b7e0d81fd8c6590823ecab8f3909ed6c824f06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160016
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29893}
2019-11-25 10:45:38 +00:00
Sam Zackrisson
31b01c0534 Fuzz APM float interface with up to 8 channels
Bug: webrtc:10859
Change-Id: Ie50b5fc102296bd71917852674cd2289e690ad78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160305
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29891}
2019-11-25 08:50:44 +00:00
Mirko Bonadei
51868f52c6 Revert "Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium.""
This reverts commit 8994c8bab315fa34b75a8e79b78bb99c86f69966.

Reason for revert: While RTC_EXPORTS are needed, this is still not
enough, I will try another approach, similar to what we do for
rtc_base/logging.{cc,h}.

Original change's description:
> Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium."
> 
> This is a reland of 2148e9a931ea1a8a2ac0bfffd56e12370f8bf18c
> 
> Original change's description:
> > Make webrtc_fuzzer_main depend on webrtc_component in Chromium.
> >
> > This is needed in order to land [1] and restrict visibility of some
> > //third_party/webrtc_overrides targets.
> >
> > [1] - https://chromium-review.googlesource.com/c/chromium/src/+/1930801
> >
> > Bug: chromium:896154
> > Change-Id: Ie71c44ee9a0203a85d77a1199acdcb8581dfb71b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160308
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29875}
> 
> No-Try: True
> No-Tree-Checks: true
> TBR: kwiberg@webrtc.org
> Bug: chromium:896154
> Change-Id: I157bd4f90528a38ac16f17dd17af2f255dbd5ec9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160401
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29888}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: If969618e3f0a0cd70204128f1e8a2b06cf407b6e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:896154
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160402
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29889}
2019-11-23 15:10:47 +00:00
Mirko Bonadei
8994c8bab3 Reland "Make webrtc_fuzzer_main depend on webrtc_component in Chromium."
This is a reland of 2148e9a931ea1a8a2ac0bfffd56e12370f8bf18c

Original change's description:
> Make webrtc_fuzzer_main depend on webrtc_component in Chromium.
>
> This is needed in order to land [1] and restrict visibility of some
> //third_party/webrtc_overrides targets.
>
> [1] - https://chromium-review.googlesource.com/c/chromium/src/+/1930801
>
> Bug: chromium:896154
> Change-Id: Ie71c44ee9a0203a85d77a1199acdcb8581dfb71b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160308
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29875}

No-Try: True
No-Tree-Checks: true
TBR: kwiberg@webrtc.org
Bug: chromium:896154
Change-Id: I157bd4f90528a38ac16f17dd17af2f255dbd5ec9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160401
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29888}
2019-11-23 11:22:30 +00:00
Mirko Bonadei
3af0cd8de2 Revert "Make webrtc_fuzzer_main depend on webrtc_component in Chromium."
This reverts commit 2148e9a931ea1a8a2ac0bfffd56e12370f8bf18c.

Reason for revert: Breaks linux-libfuzzer-asan-rel,
https://ci.chromium.org/p/chromium/builders/try/linux-libfuzzer-asan-rel/326226. I will export symbols in this CL when relanding.

Original change's description:
> Make webrtc_fuzzer_main depend on webrtc_component in Chromium.
> 
> This is needed in order to land [1] and restrict visibility of some
> //third_party/webrtc_overrides targets.
> 
> [1] - https://chromium-review.googlesource.com/c/chromium/src/+/1930801
> 
> Bug: chromium:896154
> Change-Id: Ie71c44ee9a0203a85d77a1199acdcb8581dfb71b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160308
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29875}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I1000e90e687d01c29a9ec4a3c8ded646b97fcaab
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:896154
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29885}
2019-11-23 08:41:57 +00:00
Per Åhgren
e35b32c29f AGC: Removing unnneccessary copying and changing to using const
The changes have been shown to be bitexact on a large dataset.

Bug: webrtc:10859
Change-Id: Iedc0e9e944ebfabb717dd7fb4d2682c695da883e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159694
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29883}
2019-11-22 20:14:16 +00:00
Bjorn A Mellem
c4f865413a Add TimeController to api/test/ and add a CreateTimeController API.
Creates an abstraction for an "alarm clock" which can schedule
time-controller callbacks and exposes a time controller driven by
an external alarm.

Bug: webrtc:9719
Change-Id: I08c2aa9dba25603043bfba48f55c925716a55bae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158969
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29879}
2019-11-22 17:07:23 +00:00
Mirko Bonadei
2148e9a931 Make webrtc_fuzzer_main depend on webrtc_component in Chromium.
This is needed in order to land [1] and restrict visibility of some
//third_party/webrtc_overrides targets.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1930801

Bug: chromium:896154
Change-Id: Ie71c44ee9a0203a85d77a1199acdcb8581dfb71b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160308
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29875}
2019-11-22 14:31:00 +00:00
Danil Chapovalov
ad020f5a5c Delete DEPRECATED_SingleThreadedTaskQueueForTesting
Bug: webrtc:10933
Change-Id: I4d04b15a5691e586595d59ecd1c8012cba5e3dc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160041
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29861}
2019-11-21 13:41:57 +00:00
Artem Titov
c4e506ebdf Fix writing into closed FrameWriter when dumping video in PC framework
Bug: webrtc:10138
Change-Id: I2b8e2cac2903c957dab1170ba098880a4f0252e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160049
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29853}
2019-11-21 10:01:57 +00:00
Rasmus Brandt
287e464705 Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants
This change makes it easier to propagate more information from the sink
 to the video adapter, for example alignment requirements.

Bug: None
Change-Id: I536248d59f871c103a18a48615b6c5e61f61697b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29827}
2019-11-19 10:14:31 +00:00
Patrik Höglund
56d945233d Move stun.h to api/.
We now have two downstream users of stun.h, so it appears to be
generally usable. I put this in a new dir networking/, but I'm open to
suggestions here (maybe some things in api/ should move in there).

I checked what our downstream users are actually using, and it's

cricket::ComputeStunCredentialHash
cricket::<constants>
cricket::TurnMessage
cricket::GetStunErrorResponseType
cricket::StunAttribute::CreateAddress
cricket::StunErrorCodeAttribute
cricket::StunByteStringAttribute
StunAttribute::CreateUnknownAttributes
cricket::TurnErrorType
cricket::StunMessage

I reckoned that was pretty much everything in stun.h, so I didn't
bother splitting it up. They don't use every function and constant
in there, but all _types_ of functions and constants, so for the
sake of coherence I don't think it makes sense to split it.

There's some old stuff in there like GTURN which could arguably
be split out, but it should likely go away soon anyway, so I don't
think it's worth the effort.

Steps:
1) land this
2) update downstream to point to the new header and target
3) remove p2p/base:stun_types.

Bug: webrtc:11091
Change-Id: I1f05bf06055475d25601197ec6fefb8d3b55e8e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159923
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29822}
2019-11-18 16:11:27 +00:00
Danil Chapovalov
aa3f5da8dc Fork VCMPacket for PacketBuffer into own struct
it is easier to reduce and eliminate it when it is not bound to legacy video code

Bug: webrtc:10979
Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29821}
2019-11-18 15:48:07 +00:00
Danil Chapovalov
ccf12c6e97 Reland "Add AV1 RtpDepacketizer class"
This is a reland of 49470c2ac460ed8cce250942e8525c5f14e32778
Tentative reland to rule-out bot flakiness.

Original change's description:
> Add AV1 RtpDepacketizer class
>
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
>
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29814}

TBR=saza@webrtc.org,philipel@webrtc.org

Bug: webrtc:11042
Change-Id: Ibd672ce685bcab86960500740465539ed70fcdf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159941
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29819}
2019-11-18 15:23:08 +00:00
Yves Gerey
9f99175710 Revert "Add AV1 RtpDepacketizer class"
This reverts commit 49470c2ac460ed8cce250942e8525c5f14e32778.

Reason for revert: Seems to trigger linker error on iOS64. See:
https://ci.chromium.org/p/webrtc/builders/ci/iOS64%20Debug/17733

Original change's description:
> Add AV1 RtpDepacketizer class
> 
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
> 
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29814}

TBR=danilchap@webrtc.org,saza@webrtc.org,philipel@webrtc.org

Change-Id: I2eb5994d8e31e12d6cb6e9f792b691ed10d9df81
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11042
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159940
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29815}
2019-11-18 12:14:56 +00:00
Danil Chapovalov
49470c2ac4 Add AV1 RtpDepacketizer class
Implement Parse function that extracts is_first_packet_in_frame,
is_last_packet_in_frame, and frame_type fields.

Bug: webrtc:11042
Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29814}
2019-11-18 09:39:34 +00:00
Jerome Humbert
39bab5afb5 Add missing assert.h for win no-test build
Add some missing `#include <assert.h>` for Windows build when compiling
without RTC tests (rtc_include_tests = false) with the MSVC compiler
(is_clang=false, use_lld=false).

Bug: None
Change-Id: Ie9861100efeae87f4c4e29303d62293ad541125a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158533
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29811}
2019-11-17 07:52:32 +00:00
Danil Chapovalov
1b66890a45 Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in call tests
This is practiaclly a reland of the
https://webrtc-review.googlesource.com/c/src/+/157896
except that video multi stream tests are still using the deprecated TaskQueue
(see https://webrtc-review.googlesource.com/c/src/+/159280)

Bug: webrtc:10933
Change-Id: Ie715345924f9dd2d7dd52c99de3ea595b6fad5ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159699
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29794}
2019-11-13 16:21:40 +00:00
Sebastian Jansson
50f8686c6d Providing EmulatedRoute instances when creating TcpMessageRoute
This CL makes it so the caller explicitly has to provide
EmulatedRoute instances when creating TcpMessageRoute.
Previously those were automatically generated.

This means that the EmulatedRoute instances can be reused.

Bug: webrtc:9883
Change-Id: I7ad03ca6a5a3dbb91df76e3ca1190a1a75bf0cc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159703
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29791}
2019-11-13 14:55:39 +00:00