Commit Graph

29393 Commits

Author SHA1 Message Date
fd76b5fe86 Introduce factory method for IVF frame generator
Bug: webrtc:10138
Change-Id: I9039aa289c935b7fcc2f3ab4ddec6413eb1302c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161324
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30011}
2019-12-05 09:28:56 +00:00
749f6604a1 Enable SSRC 0 in MediaChannel methods
Refactor voice engine and video engine to use default methods instead of
treating 0 as a special value.

Bug: webrtc:8694
Change-Id: I47c211c6e870cdec737d6b0d05df29a9b534a011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158600
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30010}
2019-12-04 23:49:04 +00:00
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
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
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
577b88dae7 Add new_request flag to SendFullIntraRequest
This allows one to request the same sequence number again
in the case of resending an FIR to the a sender before the
sender has time to send a key-frame.

Bug: webrtc:11171
Change-Id: Idd8e8120ccbcc194cefb8d0cf3f7cc64e7f76aa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161236
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30006}
2019-12-04 13:45:02 +00:00
17f82cfc68 Verifies trials are populated when creating a Call.
This check just makes it more clear what the expectations are.

Pululating trials was made mandatory in an earlier CL, but if you don't
populate this field it will trigger a DCHECK at lower layer where we're
actually trying to parse an experiment. That is confusing and
misleading.

Bug: None
Change-Id: I1f520841a5a3b911048c8ee6d309eb7bb179e037
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161301
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30005}
2019-12-04 13:36:02 +00:00
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
9c27ed23d2 VideoRtpReceiver: Enable encoded frame sink.
This change finally wires up VideoRtpReceiver::OnGenerateKeyFrame and
OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded
frames can flow to sinks installed in VideoTrackSourceInterface.

Bug: chromium:1013590
Change-Id: I76f8226752294aee8fe137d1a78ee66548900cc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161095
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30003}
2019-12-04 12:55:40 +00:00
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
32565f684b WebRtcVideoEngine: Enable encoded frame sink.
This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame
and OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded
frames can flow to sinks installed in VideoTrackSourceInterface.

Bug: chromium:1013590
Change-Id: I136132c210e5811547f2522ddc371d0acac90664
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161093
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30001}
2019-12-04 11:15:51 +00:00
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
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
b810646c54 AEC3: Ensure that the high-pass filter is on whenever the AEC is active
This CL ensures that the high-pass filter is on whenever the echo
controller is on. This is important as the echo controller code assumes
that the external high-pass filter is active.

The CL also corrects the ToggleAec unit test (which started failing
after this code change).

Bug: webrtc:11159, chromium:1030179
Change-Id: Ief86eda8f7c67df1c25ac1a06d2cc0778e01196d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161228
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29998}
2019-12-04 09:38:04 +00:00
246724b0fe Move messaging -> PostTask for freeing datachannels
I could find no reason for the extra complexity of doing messaging
in order to schedule a task to be done after the current cycle.
It also simplifies the peerconnection/datachannelcontroller coupling.

Bug: webrtc:11146
Change-Id: I68f45059b9f4a6869fb44b856e05a480f4652365
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161232
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29997}
2019-12-03 22:57:17 +00:00
e19a375f8c Propagate RtpPacketReceived::arival_time_ms() when demuxing RTX packets
Bug: webrtc:11163
Change-Id: I3bf4a662c84e9b31e0b0fc15660d360413a4aee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161224
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29996}
2019-12-03 21:10:53 +00:00
934afc6ba1 Deprecate RtpReceiver's SetParameters method
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.


Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
2019-12-03 19:50:42 +00:00
c5d7c52d44 Remove line number from rtc::Location
Concatenating __FILE__ with __LINE__ prevents the compiler from
aliasing strings within the same file, contributing ~30KB of .text
bloat. Chrome already omits from the file number from its Location
type so it doesn't seem to be a big loss.

Bug: b/145168048
Change-Id: I000bfdf43f4eb90f8b63ed017b08c1b5a7a84a6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160744
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29994}
2019-12-03 19:43:12 +00:00
b588353543 Implement handling for MS PowerPoint Presentation Mode.
* Introduce a FullScreenWindowDetector to manage routines for updating the list of sources being application agnostic, inspired by FullScreenChromeWindowDetector.
* Introduce a FullScreenApplicationHandler to make a decision about changing window to share in application specific way, inspired by FullScreenChromeWindowDetector.
* Remove FullScreenChromeWindowDetector as redundant.
* Add FullScreenApplicationHandler for MS PowerPoint and Apple Keynote on MacOS.
* Add FullScreenApplicationHandler for MS PowerPoint on Windows.

Bug: webrtc:3852
Change-Id: I06507d929308e85b882b2f8210a025afef7f26a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156020
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Wez <wez@google.com>
Cr-Commit-Position: refs/heads/master@{#29993}
2019-12-03 19:17:24 +00:00
831ce5f171 Export more symbols to fix Chromecast component build
When building certain Chromecast build flavors in component build mode,
there are some link errors due to symbols not being exported. This CL
fixes those issues.

TBR: kwiberg@webrtc.org
Bug: None
Change-Id: I408f0a84b8ac4610cc6b5aa6ff58248ea82c9c66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161148
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29992}
2019-12-03 17:41:04 +00:00
3c2679f1bb Disable unittest that takes too long to finish
Bug: webrtc:11161
Change-Id: Ic8de7ce07fabc251bc6269a0ecab33bdb1b22280
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161235
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29991}
2019-12-03 17:27:55 +00:00
e725fdbcc1 Encoding constructor should be public.
Bug: webrtc:11157
Change-Id: Ic921b39c0aa6b24deef8452ec28352c16c193a1e
TBR: sakal@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161151
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29990}
2019-12-03 16:54:20 +00:00
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
269ac81a86 VideoReceiveStream: Enable encoded frame sink.
This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame and
OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames
can flow to sinks installed in VideoTrackSourceInterface.

Bug: chromium:1013590
Change-Id: I0779932c251a2159880a39b2d42d5ce439cc88e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161090
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29988}
2019-12-03 15:55:04 +00:00
05e4d08e35 Refactoring DataChannelController from PeerConnection part 4
This CL:
- Moved HasDataChannel and data_channel_type_
- Moved rtp_data_channels_
- Moved sctp_data_channels_
- Moved data_channel_controller to its own .h file
- Various changes to reduce the coupling between the classes
- Removed friendship between DataChannelController and PeerConnection

Bug: webrtc:11146
Change-Id: Ib8c395e4c90ce34baf40812d1dade0ffa79f2438
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161094
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29987}
2019-12-03 15:35:09 +00:00
bcce4530d4 AEC3: Add flag for requiring a high pass filter effect before the AEC
Bug: webrtc:11159, chromium:1030179
Change-Id: Ic6b686a36efa433156a66252394fa551b8ed1837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161227
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29986}
2019-12-03 15:34:04 +00:00
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
5d3a418a26 Add explicit copy constructors and assign operators for some classes.
It should fix compilation errors that happen on some iOS bots saying
"definition of implicit copy assignment operator for 'Foo'
is deprecated because it has a user-declared copy constructor"

Bug: webrtc:11162
Change-Id: Ife3d1a800ed6a4cd08bdfd156cd0e320504ee8dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161221
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29984}
2019-12-03 14:27:45 +00:00
a8c2f5180f Remove unused non-standard RtpEncodingParameters members
Bug: webrtc:7580
Change-Id: Ic1a6e52f25eb35c797e669bffe8040ec84fec386
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160415
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29983}
2019-12-03 13:26:53 +00:00
6c0e94650e Fix VP8 encoder maxFramerate support
By using the top level VideoCodec maxFramerate, the FrameBufferController
would sometimes not use the intended value for each simulcast layer.
In the case of "conference mode", top level maxFramerate was set to 5,
which matches the lower layer but is different from the overall maximum
maxFramerate which would be 60.

Bug: webrtc:11117
Change-Id: I4e1e68184d32675b083cd8e4e73a5291dc8fa620
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161096
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29982}
2019-12-03 12:47:20 +00:00
4011de012f Revert "AEC3: Ensure that the high-pass filter effect is on when AEC3 is active"
This reverts commit 3a77f93589f77aabfdfa75cd7202f09b4315ff7c.

Reason for revert: The change is breaking downstream tests.

Original change's description:
> AEC3: Ensure that the high-pass filter effect is on when AEC3 is active
> 
> This CL ensures that the high-pass filter is on whenever the echo
> controller is on. This is important as the echo controller code assumes
> that the external high-pass filter is active.
> 
> The CL also corrects the ToggleAec unit test (which started failing
> after this code change).
> 
> Bug: webrtc:11159,chromium:1030179
> Change-Id: Ie29db74bf3de6279a08564398d32d67d5e1569db
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161222
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29979}

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

Change-Id: I78b4e397555f50898ca42c4b32fb39cf06a2541a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11159, chromium:1030179
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161226
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29981}
2019-12-03 11:49:00 +00:00
6a05bb1b12 AEC3: Add signal dependent mixing before alignment
This CL adds code for doing signal-dependent downmixing
before the delay estimation in the multichannel case.

As part of the CL, the unittests of the render delay
controller are corrected. However, as that caused some of
them to fail, the CL (for now) as well disables the failing
test.

Bug: webrtc:11153,chromium:1029740, webrtc:11161
Change-Id: I0b765c28fa5e547aabd6dfbd24b626ff9a16346f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161045
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29980}
2019-12-03 11:30:48 +00:00
3a77f93589 AEC3: Ensure that the high-pass filter effect is on when AEC3 is active
This CL ensures that the high-pass filter is on whenever the echo
controller is on. This is important as the echo controller code assumes
that the external high-pass filter is active.

The CL also corrects the ToggleAec unit test (which started failing
after this code change).

Bug: webrtc:11159,chromium:1030179
Change-Id: Ie29db74bf3de6279a08564398d32d67d5e1569db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161222
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29979}
2019-12-03 11:06:08 +00:00
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
67d3bc2b3d Changed parameter name to match the use of it in AEC3
Bug: webrtc:8671
Change-Id: I016c8ad81556aef5b671466abcc88c0c15234465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160783
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29977}
2019-12-03 10:18:14 +00:00
b5aa0a85f3 Add IceControllerEvent::ICE_CONTROLLER_RECHECK
This patch adds a new enum value in IceControllerEvent::Type,
that allows an IceController to request a recheck without
any of the predefined event occuring.

This patch is a NOP for BasicIceController.

BUG=webrtc:10647

Change-Id: Idf1d0946480437109ff272946679fef559ca7beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161047
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29976}
2019-12-03 10:17:09 +00:00
464741276f Make new method pure virtual in the EchoControl interface
Bug: webrtc:8671
Change-Id: I673e79fae1e763131b71fceb64d7a909e662f568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160785
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29975}
2019-12-03 10:16:08 +00:00
21021f022b NetEq: Fix bug in PLC for multi-channel audio
There is currently a bug in NetEq that causes audio to leak from the
first channel to all others during loss concealment. This CL fixes the
problem and also adds a unit test to verify.

Bug: webrtc:11145
Change-Id: Ia6c4a234ff7f78e9a6080f1cb17eb80af671c3dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161091
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29974}
2019-12-02 17:44:58 +00:00
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
b2b58d84e3 AEC3: Adding default AEC3 configurations that are setup specific
This adds functionality to AEC3 to produce setup-specific
default configurations that are tailored to work well for the
number of channels at hand.

The tunings are only used for the case when no echo control factory
has been provided.

Bug: webrtc:11151,chromium:1029717
Change-Id: I1bd2d10327300c7b0f3169a52bf66700b781fd6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161086
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29972}
2019-12-02 15:04:07 +00:00
cf20519262 AEC3: Correct the number of render channels in the echo audibility code
Bug: webrtc:11150,chromium:1029707
Change-Id: I4d43bfcd52871a45e7608158bf19c32523976f55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161085
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29971}
2019-12-02 13:13:11 +00:00
00cf34c5e8 Refactor DataChannel control out of PeerConnection
This is step 1-3 of the refactoring process outlined in comment #1 of bugs.webrtc.org/11146

Bug: webrtc:11146
Change-Id: Iccad009bc0585f99d207a6ddb42fd8e71312fc0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161003
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29970}
2019-12-02 10:00:34 +00:00
b877e71c41 Delete FunctorMessageHandler, in tests use alternative ways to post functors
Bug: None
Change-Id: I22c7623300c2370269651b67a919a3c4a2ff3926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161080
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29969}
2019-12-02 09:47:34 +00:00
f2c0818fa2 Minor fixes to ChannelSend.
Bug: None
Change-Id: Ic651174afa2d8b9b105d03adcf725549bcc144df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160782
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29968}
2019-12-02 09:30:51 +00:00
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
b144c58973 Remove deprecated setting for activating multichannel processing
Bug: webrtc:10859
Change-Id: I86cae6a9b765bc807c00632ec7d743b754941f81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160780
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29966}
2019-12-02 07:47:53 +00:00
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
865a74ee59 Roll chromium_revision a9c1e4afb9..3f97848513 (720171:720272)
Change log: a9c1e4afb9..3f97848513
Full diff: a9c1e4afb9..3f97848513

Changed dependencies
* src/ios: 0c6f3cc26c..a31907ccb8
* src/testing: 7fc8295e7d..c011aaeb88
* src/third_party: d4ae047bf6..245344e1cb
* src/third_party/depot_tools: 7c62ed63ac..5ae4817ada
* src/third_party/freetype/src: d4f9c24989..4270e9f324
* src/tools: 0f56390d56..cc179a4932
DEPS diff: a9c1e4afb9..3f97848513/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I316b80951a88d3ccb6d4e03417254c60acade01c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161083
Reviewed-by: 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/master@{#29964}
2019-11-29 20:38:51 +00:00
d003662b15 Move SendBindingResponse to Connection
This patch moves the SendBindingResponse from Port
to Connection. This is a behavioural NOP, and I don't
understand why it was in Port in the firs place!

Found when working on GOOG_PING.

BUG=webrtc:11100

Change-Id: I0466c5381f08ec4926ca3380e6914f0bc0dfcf63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161081
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29963}
2019-11-29 16:32:14 +00:00
89313451d8 Take FunctionView rather than any functor reference in the rtc::Thread::Invoke
to generate less versions of the function template and FunctorMessageHandler helper
thus producing less binary size

Bug: None
Change-Id: Idbd6fb1e1f23b9b2dc4e4306a74ef11e74ba94cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161044
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29962}
2019-11-29 14:10:38 +00:00