Commit Graph

33755 Commits

Author SHA1 Message Date
752cbaba90 Enable quality scaling when allowed
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).

This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.

Bug: chromium:1179020, webrtc:12511
Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33373}
2021-03-03 13:57:22 +00:00
db0b4a8935 Do not crash if codec is not available
Check if codec was successfully created and exit from RunTest if not
before creating VideoProcessor.

Bug: none
Change-Id: Ia6d7171650dbc9824fb78f4a8e2851f755cfd63b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209362
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33372}
2021-03-03 13:56:17 +00:00
652ada5029 Enabling a safe fall-back functionality for overruns in the runtime settings
Bug: b/177830919
Change-Id: I9369f6fc004ceb2b626d33b36262bc8aeabdb1a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206988
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33371}
2021-03-03 12:06:54 +00:00
99bcf60a41 Check MID for illegal token characters.
Bug: webrtc:12516
Change-Id: I311dc984aa1dc8784d3ba3394676337b35cc92d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209360
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33370}
2021-03-03 08:55:39 +00:00
c67b77eee4 Add a fuzzer test that tries to connect a PeerConnection.
Bug: none
Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33369}
2021-03-02 19:28:23 +00:00
94f77ebbc5 Allow using test code in fuzzers
This makes certain fuzzers easier to write.

Bug: none
Change-Id: If1dad462ff71075c9cc81d35f19556b4c1b7108f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209221
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33368}
2021-03-02 15:30:21 +00:00
09932cded8 Manually update jdk DEPS.
This should hopefully unblock the WebRTC DEPS autoroller
https://ci.chromium.org/p/webrtc/builders/cron/Auto-roll%20-%20WebRTC%20DEPS

Bug: None
Change-Id: I0b6df2e4ca7df5a3ca1f58e86b3f95a493eddef1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209183
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33367}
2021-03-02 12:56:27 +00:00
07d83c8a9a Modified STUN verification functions
The new verification makes verification a function on a message.
It also stores the password used in the request message, so that
it is easily accessible when verifying the response.

Bug: chromium:1177125
Change-Id: I505df4b54214643a28a6b292c4e2262b9d97b097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209060
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33366}
2021-03-02 10:34:17 +00:00
d99da804d7 Parent CL.
Change-Id: I0221159dc334f8c358fcc73c8e2cbe41da89799c
No-Try: True
TBR: mbonadei@webrtc.org
Bug: webrtc:12512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209080
Reviewed-by: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33365}
2021-03-02 09:43:11 +00:00
a88fe7be14 Fix race between destroying SctpTransport and receiving notification on timer thread.
This gets rid of the SctpTransportMap::Retrieve method and forces
everything to go through PostToTransportThread, which behaves safely
with relation to the transport's destruction.

Bug: webrtc:12467
Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33364}
2021-03-01 22:53:13 +00:00
88a51b2902 Populate "total_round_trip_time" and "round_trip_time_measurements" for remote inbound RTP streams
Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*

Adding them into the stats definition as well.

Bug: webrtc:12507
Change-Id: Id467a33fe7bb256655b68819e3ce87ca9af5b25f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209000
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33363}
2021-03-01 20:49:22 +00:00
625e1d9b16 VP8 ref finder unittest cleanup
Change-Id: I627dda1229ceb4b2da3f37f7418da7b7653e4d04
Bug: webrtc:12221
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208482
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33362}
2021-03-01 17:23:38 +00:00
86f04ad135 Populate “fractionLost” stats for remote inbound rtp streams
Tests:
./out/Default/peerconnection_unittests

Manually tested with Chromium to see the data populated

Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
Bug: webrtc:12506
Change-Id: I60ef8061fb31deab06ca5f115246ceb5a8cdc5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33361}
2021-03-01 16:48:37 +00:00
6512af04ba Add root doc site definition for WebRTC documentation
Bug: None
Change-Id: I64f5a356e0358360bd8326c39c8b0a898b879ebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208641
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33360}
2021-03-01 16:40:58 +00:00
f57f2cd3d2 doc: add M88/M89 release notes
BUG=None

Change-Id: Ia5bb63a0825a4dd5aaeca305fb7796f36dd06e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33359}
2021-03-01 16:16:03 +00:00
64f7da0b16 Use input_state to get pixels for single active stream.
Bug: none
Change-Id: I103d2cc111ca08d1e5acde1f25c125c075502eca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208226
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33358}
2021-03-01 15:58:03 +00:00
31e06cb63d addIceCandidate: prefer ice candidate sdpMid over sdpMLineIndex
as described in JSEP
  https://tools.ietf.org/html/rfc8829#section-3.5.2.1
 If the MID field is present in a received IceCandidate, it
 MUST be used for identification; otherwise, the "m=" section
 index is used instead.

BUG=webrtc:12479

Change-Id: I688a5e59024fe8cc6a170c216c6f14536084cfb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33357}
2021-03-01 14:01:39 +00:00
df6d4ca3f5 Prevent TURN from connecting to ports < 1024 (except 443)
This should prevent TURN from being used in NAT slipstream attacks
with native clients.

Bug: webrtc:12497
Change-Id: I3c33df92e97b8e6430b72a05790c137d89661093
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208582
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33356}
2021-03-01 13:43:39 +00:00
bfc3ef0fec Mention ninja all in webrtc development documantation.
Bug: b/177971218
Change-Id: Ia49ca147ad286ef1ad8ac8ddd6b7fd8b05e384ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208942
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33355}
2021-03-01 12:14:13 +00:00
4785402475 Replace RecursiveCriticalSection with Mutex in ProcessThreadImpl
Bug: webrtc:11567
Change-Id: I03961ddc55f29a01c3e466217222fd56ba51d895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208764
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33354}
2021-03-01 12:13:09 +00:00
1e2da374b3 Update WebRTC code version (2021-02-28T04:02:30).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: If3304144dbf58dec0babaf873a5858f40539b908
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209020
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#33353}
2021-02-28 04:56:14 +00:00
ac615b6766 Update WebRTC code version (2021-02-27T04:03:14).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: If05da5731bedf3310c0748de02ade5c8597e94d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208920
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#33352}
2021-02-27 06:02:56 +00:00
c96aa30c13 Minor refactoring of RtpVideoSender
Bug: None
Change-Id: I7cc16d4e435974090310bd3a7cea9eaf0586be8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208761
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33351}
2021-02-26 13:20:52 +00:00
484acf2723 Add ability to configure sampling rate for input/output video dumps in PC level framework
Bug: b/179986638
Change-Id: I9ab960840e4b8f912abe4fb79cfd9278f4d4562a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33350}
2021-02-26 11:24:52 +00:00
9d9b8defae Reland "Rename SIGNALING and WORKER to PRIMARY and SECONDARY"
This is a reland of a37f2bd9421868e222d591d3371486a6ab939fd6

Original change's description:
> Rename SIGNALING and WORKER to PRIMARY and SECONDARY
>
> This makes the proxy macros less confusing when the secondary thread
> is sometimes the worker thread, sometimes the networking thread.
>
> Bug: none
> Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33346}

Bug: none
Change-Id: If46a6679ac0fc947797dd7be87626ef7702faca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208845
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33349}
2021-02-26 10:35:22 +00:00
376cf384ac Replace RecursiveCriticalSection with Mutex in EmulatedEndpointImpl
Bug: webrtc:11567
Change-Id: Ie9a1f123e7d2858c03414336875d8c537be67702
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208403
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33348}
2021-02-26 09:58:49 +00:00
07a01d09e4 Revert "Rename SIGNALING and WORKER to PRIMARY and SECONDARY"
This reverts commit a37f2bd9421868e222d591d3371486a6ab939fd6.

Reason for revert: Breaks compile step (e.g. https://ci.chromium.org/ui/p/webrtc/builders/ci/Android64%20Builder%20x64%20(dbg)/19773/overview).

Original change's description:
> Rename SIGNALING and WORKER to PRIMARY and SECONDARY
>
> This makes the proxy macros less confusing when the secondary thread
> is sometimes the worker thread, sometimes the networking thread.
>
> Bug: none
> Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33346}

TBR=hta@webrtc.org

Bug: none
Change-Id: I2014faab3392f445f56edd9e833d00000ebc5ca3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208840
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33347}
2021-02-26 07:20:49 +00:00
a37f2bd942 Rename SIGNALING and WORKER to PRIMARY and SECONDARY
This makes the proxy macros less confusing when the secondary thread
is sometimes the worker thread, sometimes the networking thread.

Bug: none
Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33346}
2021-02-25 22:16:00 +00:00
2aeab5ed3f Make the PC proxy invoke LookupDtlsTransportByMid on the network thread
Bug: webrtc:12489
Change-Id: I786c968e4ee07c9bbce4a1c850a6f8f0c55810c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208580
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33345}
2021-02-25 21:51:35 +00:00
5cfcf2282a modules/desktop_capture: replace memcpy with libyuv::CopyPlane
According to our previous data from trace_event with using direct memcpy
and libyuv::CopyPlane on chromebook atlas, the average cpu duration is
0.624ms and 0.541ms, so using libyuv::CopyPlane is 13.3% faster than
direct memcpy.

Bug: webrtc:12496
Change-Id: I1c41424b402a7eec34052c67933f2e88eaf0a8f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196485
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33344}
2021-02-25 21:50:30 +00:00
258e9899f4 Use default ResolutionBitrateLimits for simulcast with one active stream if not configured
Bug: none
Change-Id: I049dd0924adc43ce249a8eda63cdcb13da42b030
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208541
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33343}
2021-02-25 12:14:45 +00:00
1124ed1ab2 Communicate encoder resolutions via rtc::VideoSinkWants.
This will allow us to optimize the internal buffers of
webrtc::VideoFrame for the resolution(s) that we actually want to
encode.

Bug: webrtc:12469, chromium:1157072
Change-Id: If378b52b5e35aa9a9800c1f7dfe189437ce43253
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33342}
2021-02-25 11:10:55 +00:00
bb52bdf095 Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium"
This reverts commit cd5127b11e04b7f49031b2412625c36e4a86c3da.

Reason for revert: Fuzzer build problems fixed.

Original change's description:
> Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium"
>
> This reverts commit dfe19719e53abfd4d73722942445c5e1046b671b.
>
> Reason for revert: Breaks fuzzers in Chromium builds. See https://ci.chromium.org/ui/p/chromium/builders/try/linux-libfuzzer-asan-rel/685438/overview. I am reverting since this blocks the roll but I will be in touch for a fix.
>
> Original change's description:
> > Enable use of rtc::SystemTimeNanos() provided by Chromium
> >
> > This is the third CL out of three to enable overriding
> > of the function SystemTimeNanos() in rtc_base/system_time.cc
> >
> > When WebRTC is built as part of Chromium the rtc::SystemTimeNanos()
> > function provided by Chromium will be used. This is controlled
> > by the build argument rtc_exclude_system_time which directly
> > maps to the macro WEBRTC_EXCLUDE_SYSTEM_TIME.
> >
> > By doing this we are making sure that the WebRTC and Chromium
> > clocks are the same.
> >
> > Bug: chromium:516700
> > Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33337}
>
> TBR=kron@webrtc.org
>
> Bug: chromium:516700
> Change-Id: I9ecd1784a6c1cdac8bae07d34f7df20c62a21a95
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208740
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33340}

Bug: chromium:516700
Change-Id: I4cd68bac1cc4befdb46351f5d6fb2cf1ef5c3062
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208742
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33341}
2021-02-25 10:48:55 +00:00
cd5127b11e Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium"
This reverts commit dfe19719e53abfd4d73722942445c5e1046b671b.

Reason for revert: Breaks fuzzers in Chromium builds. See https://ci.chromium.org/ui/p/chromium/builders/try/linux-libfuzzer-asan-rel/685438/overview. I am reverting since this blocks the roll but I will be in touch for a fix.

Original change's description:
> Enable use of rtc::SystemTimeNanos() provided by Chromium
>
> This is the third CL out of three to enable overriding
> of the function SystemTimeNanos() in rtc_base/system_time.cc
>
> When WebRTC is built as part of Chromium the rtc::SystemTimeNanos()
> function provided by Chromium will be used. This is controlled
> by the build argument rtc_exclude_system_time which directly
> maps to the macro WEBRTC_EXCLUDE_SYSTEM_TIME.
>
> By doing this we are making sure that the WebRTC and Chromium
> clocks are the same.
>
> Bug: chromium:516700
> Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33337}

TBR=kron@webrtc.org

Bug: chromium:516700
Change-Id: I9ecd1784a6c1cdac8bae07d34f7df20c62a21a95
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208740
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33340}
2021-02-25 08:47:19 +00:00
49b2792b70 Update WebRTC code version (2021-02-25T04:03:13).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I8cd8299f20e4f16b5ba9ee540764a1a8720af5cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208680
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#33339}
2021-02-25 06:36:17 +00:00
c500977983 Change the safe SCTP MTU size to 1191
Bug: webrtc:12495
Change-Id: Ie149391a5a9f61095cf3f31db141c9bbc8be8bee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208642
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33338}
2021-02-25 00:05:35 +00:00
dfe19719e5 Enable use of rtc::SystemTimeNanos() provided by Chromium
This is the third CL out of three to enable overriding
of the function SystemTimeNanos() in rtc_base/system_time.cc

When WebRTC is built as part of Chromium the rtc::SystemTimeNanos()
function provided by Chromium will be used. This is controlled
by the build argument rtc_exclude_system_time which directly
maps to the macro WEBRTC_EXCLUDE_SYSTEM_TIME.

By doing this we are making sure that the WebRTC and Chromium
clocks are the same.

Bug: chromium:516700
Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33337}
2021-02-24 22:25:33 +00:00
dac39c5b1e Reland "Add test for odd sizes with spatial layers"
This is a reland of 6fe3fa14c6686ba9c51095b97ad2e6833a9b03e5

Original change's description:
> Add test for odd sizes with spatial layers
>
> Bug: webrtc:12398
> Change-Id: If28f22f8c08913315806d26ad0b355eabda67da6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203889
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Jerome Jiang <jianj@google.com>
> Cr-Commit-Position: refs/heads/master@{#33319}

TBR=philipel@webrtc.org

Bug: webrtc:12398
Change-Id: I0c52a5d2d503180793603c148b3211df3ca035e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33336}
2021-02-24 21:10:03 +00:00
61d1773c03 Remove deactivated RTP modules from PacketRouter map.
Apart from making the map smaller, a purpose of this is guaranteeing
that if a module has been deactived it will not receive new packets
from the pacer, which will be needed for deferred sequencing.

Bug: webrtc:11340
Change-Id: I171a13413c5b8d3fa569c2d56bd9a54bff7c7976
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208542
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33335}
2021-02-24 20:09:13 +00:00
451a8af691 Feed the clock skew to AbsoluteCaptureTimeReceiver in audio receiver.
Bug: webrtc:10739
Change-Id: Ie61582079fb1791954b1929b6a3bf4c9edb7d75e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207433
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33334}
2021-02-24 16:02:52 +00:00
cd0373f013 peerconnection: add was-ever-connected boolean flag
and report some metrics only on the first connection state
change to connected

BUG=webrtc:12383

Change-Id: I32908e23c51aa40730be8e534793829268d4e25e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208583
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33333}
2021-02-24 13:22:42 +00:00
2ee9415a8c AndroidVideoDecoder: Ignore format updates with zero dimensions
Sometimes c2.qti.vp8.decoder reports format updates with zero frame
width / height right after initialization, that leads to the
precondition check failure made by SurfaceTextureHelper.setTextureSize.
This patch makes AndroidVideoDecoder.reformat to ignore such format
updates so as to continue to use this HW decoder.
It seems to be safe because this decoder singals one more format update
with valid dimensions soon and continue to operate in normal mode.

Bug: webrtc:12492
Change-Id: I5155166637bd2d4247d31e608d714e687e0ad1df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208222
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33332}
2021-02-24 11:37:32 +00:00
eaedde7e16 Remove old workaround in PacingController
Bug: None
Change-Id: I23f3548f21b464fe5e211c9895927ee0d978e1f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208543
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33331}
2021-02-24 10:44:42 +00:00
0093a38f7c Fix low-latency renderer with unset playout delays
The low-latency renderer is activated by the RTP header extension
playout-delay if the min value is set to 0 and the max value is
set to something greater than 0.

According to the specification of the playout-delay header
extension it doesn't have to be set for every frame but only if
it is changed. The bug that this CL fixes occured if a playout
delay had been set previously but some frames without any specified
playout-delay were received. In this case max composition delay
would not be set and the low-latency renderer algorithm would be
disabled for the rest of the session.

Bug: chromium:1138888
Change-Id: I12d10715fd5ec29f6ee78296ddfe975d7edab8a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208581
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33330}
2021-02-24 09:35:49 +00:00
198299c161 Roll src/third_party/libjpeg_turbo/ fa0de0767..7b4981b65 (2 commits)
fa0de07678..7b4981b650

$ git log fa0de0767..7b4981b65 --date=short --no-merges --format='%ad %ae %s'
2021-02-23 enm10k Include jpeglibmangler.h
2021-02-02 ehmaldonado Move metadata in OWNERS files to DIR_METADATA files

Created with:
  roll-dep src/third_party/libjpeg_turbo

Bug: none
Change-Id: I034188911ccf8cee1e864cf156d0dd4c35759992
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208600
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33329}
2021-02-24 08:40:13 +00:00
77475ecfd5 Delete unused sigslot variables.
Bug: webrtc:11943
Change-Id: I55b9360de5188b1519aed184144f66d69763828c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33328}
2021-02-24 06:47:39 +00:00
c32f00ea9d Remove usage of sigslot and replace with a function pointer.
- Deleted unused sigslot.

Bug: webrtc:11943
Change-Id: I7863dd04e3e63fcba0eabd0dd752ab339614814e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207840
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33327}
2021-02-24 06:40:09 +00:00
5fec23cdb5 Update WebRTC code version (2021-02-24T04:02:50).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: Ifaf8855bd61278071f7f7044ff6fbbcb9642d382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208620
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#33326}
2021-02-24 05:36:09 +00:00
8af6b4928a Populate jitter stats for video RTP streams
Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!

Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33325}
2021-02-23 15:10:02 +00:00
373bb7bec4 Don't use SystemTimeNanos() for current wallclock time on WINUWP
SystemTimeNanos() will soon be replaced with another implementation
when built with Chromium. This will break the assumption of
SystemTimeNanos() being relative to the NTP epoch. To avoid breaking
any UWP apps, call the TimeHelper::Ticks() function directly, which
is synchronized with the NTP epoch during initialization.

Bug: chromium:516700
Change-Id: I4e50cb3f88d06e1385e73b1a9ded52956501dc1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208520
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33324}
2021-02-23 14:53:22 +00:00