Commit Graph

37788 Commits

Author SHA1 Message Date
b92d3e6ef9 [PCLF] Move FEC and bitrate mulitplier into per peer configs
Bug: b/213863770
Change-Id: Idcf37150e769db18d4a12baa1057840d521b8e1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251761
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36006}
2022-02-15 18:10:31 +00:00
ac341df436 Adding fuzzer for PCM16b decoder and fixing a fuzzer problem
Bug: chromium:1280852
Change-Id: I7f6c5de86ceee01156743c0389c59f875e53bb5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251580
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36005}
2022-02-15 15:59:01 +00:00
e47493b3c0 Add restrictive visibility to all targets in //pc
This CL sets all visibility to ":*" (this buildfile) where no users
outside this directory are known, and marks up publicly exported
targets and Chrome dependencies explicitly.

Bug: webrtc:13661
Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 14:40:52 +00:00
1b083a998b Encode data for compression + add initial tests
Bug: webrtc:13607
Change-Id: I3bbec5558e676ca45125fad3fdbd10cc47c84601
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251390
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#36003}
2022-02-15 14:24:11 +00:00
23bb9d75fc Allow designated initializers in WebRTC
to align with chromium and google style guides

Bug: None
Change-Id: I92e1bb6d187eac6b531d495aedb8176f66186a5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251689
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36002}
2022-02-15 13:23:12 +00:00
599002c905 Restrict frame id range in frame buffer 3 fuzzer
Bug: chromium:1293129
Change-Id: Icc9152447363e69b2be561bc90a23f411d64b11a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251385
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36001}
2022-02-15 09:18:51 +00:00
987b671017 Add ability to add peer to the stats poller during the test
Bug: b/213863770
Change-Id: I65e0338806f808329725fce50d778738724cf13d
Pair: mbonadei@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251693
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36000}
2022-02-15 09:09:02 +00:00
f7a1937e70 Add FrameBufferProxy test for low-latency renderer
Ensures that frames are decoded instantly when in low-latency render
mode. This also tests the max queue size behaviour. Adds a new test
suite for FrameBufferProxy that sets the appropriate field trials.

* Fixes FrameDecodeTiming to never use negative wait times for decode
timestamps.

R=kron@webrtc.org

Change-Id: I06cbec52e1e866e21aa964b24c4fd0163c26961b
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35999}
2022-02-15 08:30:51 +00:00
405ac4e840 Add objc_class_prefix to the Audio Network Adaptor proto.
WANA: WebRTC Audio Network Adaptor

No-Try: True
Bug: None
Change-Id: I291e02ab70323ecc45d87cea0ea8d7e8cb62db9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249784
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35998}
2022-02-14 21:04:20 +00:00
2f194e0325 PipeWire capturer: Import DMA-BUFs with correct render node
With more GPUs it might happen that server used different render
node from the one we pick from the list. This would cause DMA-BUF to
fail to import so we use Wayland client library to obtain wl_display in
order to initialize EGLDisplay using same render node and have previous
approach as a fallback. Also everyone else uses EGL_LINUX_DMA_BUF_EXT
target for importing EGLImages from DMA-BUF file descriptors so use it
as well to be sure we import buffers same way as they are produced.

Bug: chromium:1290566
Bug: webrtc:13429
Change-Id: I32bbb0bdb28c08b6e7fcb3f94009f82a2041b6ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250661
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35997}
2022-02-14 20:08:50 +00:00
5723d854c9 Integrate sync decoding in video_receive_stream
Wires up DecodeSynchronizer in Call if there is a Metronome injected
into the PeerConnectionFactoryDependencies.

Change-Id: I362cd12648bfa0c32e73111fcd0f3296fca2b275
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251341
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35996}
2022-02-14 16:59:20 +00:00
f2b987377b in RtcpTransceiver implement sending rtcp sender reports
Bug: webrtc:8239
Change-Id: Id3298bf4e0eb18a3fc8072fb19416e67a126705f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249788
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35995}
2022-02-14 15:58:40 +00:00
66bfd6f57d Revert "Refactor AnalyzerConfig to use Timestamps instead of microseconds."
This reverts commit 43fb16921b29ecd3a2d87876dda75c575e05f66a.

Reason for revert: New type breaks downstream projects.

Original change's description:
> Refactor AnalyzerConfig to use Timestamps instead of microseconds.
>
> Add optional offset-to-UTC parameter to output. This allows aligning
> the x-axis in the generated charts to other UTC-based logs.
>
> Bug: b/215140373
> Change-Id: I65bcd295718acbb8c94e363907c1abc458067bfd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250203
> Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35992}

TBR=terelius@webrtc.org,kerl@google.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: If4f2330b9731f26a0e55c9ce9a500322a111b783
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/215140373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251691
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35994}
2022-02-14 15:01:51 +00:00
e7b4c137cc Revert "Reland "Use non-proxied source object in VideoTrack.""
This reverts commit 1158bff15f33c467543928dd6a49cb6ad04da1ba.

Reason for revert: Downstream issues unresolved (2nd of two reverts)

Original change's description:
> Reland "Use non-proxied source object in VideoTrack."
>
> This is a reland of 3eb29c12358930a60134f185cd849e0d12aa9166
>
> This reland doesn't contain the AudioTrack changes (see original
> description) that got triggered in some cases and needs to be
> addressed separately.
>
> Another change in this re-land is that instead of the `state` property
> of the VideoTrack be marshalled to the signaling thread, it's readable
> from the calling thread. Previously this was marshalled to the worker
> and the original changed that to the signaling thread (same as for
> AudioTrack) - but in case that's causing downstream problems this reland
> uses BYPASS_PROXY_CONSTMETHOD0 for the `state()` accessor of the
> VideoTrack proxy.
>
> Original change's description:
> > Use non-proxied source object in VideoTrack.
> >
> > Use the internal representation of the video source object from the
> > track. Before there were implicit thread hops due to use of the proxy.
> >
> > Also, override AudioTrack's enabled methods to enforce thread
> > expectations.
> >
> > Bug: webrtc:13540
> > Change-Id: I4bc7aca96d6fc24f31ade45e47f52599f1cc2f97
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250180
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35911}
>
> Bug: webrtc:13540
> Change-Id: Icb3e165f07240ae10730a316d3a8a3b2b9167d82
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251387
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35979}

# Not skipping CQ checks because original CL landed > 1 day ago.

Using "No-Try" to not have to wait for the win chromium bot to unblock
(currently takes hours).

No-Try: true
Bug: webrtc:13540
Change-Id: I8f34536bf472a6d069344e84d889864f195c93f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251686
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35993}
2022-02-14 13:54:20 +00:00
43fb16921b Refactor AnalyzerConfig to use Timestamps instead of microseconds.
Add optional offset-to-UTC parameter to output. This allows aligning
the x-axis in the generated charts to other UTC-based logs.

Bug: b/215140373
Change-Id: I65bcd295718acbb8c94e363907c1abc458067bfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250203
Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35992}
2022-02-14 13:48:40 +00:00
97f8a5f7dd Revert partial revert, in order to do a full revert.
This reverts commit f342d6054ad984b7b80df2afe349c3bbb5f1d5b8.
  or "Reland "Use non-proxied source object in VideoTrack.""

Reason for revert: Didn't resolve the downstream issues.

Original change's description:
> Revert "Reland "Use non-proxied source object in VideoTrack.""
>
> This reverts commit 1158bff15f33c467543928dd6a49cb6ad04da1ba.
>
> Reason for revert: This is a partial revert as we're tracking down
> the source of the downstream issues. This CL reverts the use of
> `internal()` for methods that relate to the source sink.
>
> Original change's description:
> > Reland "Use non-proxied source object in VideoTrack."
> >
> > This is a reland of 3eb29c12358930a60134f185cd849e0d12aa9166
> >
> > This reland doesn't contain the AudioTrack changes (see original
> > description) that got triggered in some cases and needs to be
> > addressed separately.
> >
> > Another change in this re-land is that instead of the `state` property
> > of the VideoTrack be marshalled to the signaling thread, it's readable
> > from the calling thread. Previously this was marshalled to the worker
> > and the original changed that to the signaling thread (same as for
> > AudioTrack) - but in case that's causing downstream problems this reland
> > uses BYPASS_PROXY_CONSTMETHOD0 for the `state()` accessor of the
> > VideoTrack proxy.
> >
> > Original change's description:
> > > Use non-proxied source object in VideoTrack.
> > >
> > > Use the internal representation of the video source object from the
> > > track. Before there were implicit thread hops due to use of the proxy.
> > >
> > > Also, override AudioTrack's enabled methods to enforce thread
> > > expectations.
> > >
> > > Bug: webrtc:13540
> > > Change-Id: I4bc7aca96d6fc24f31ade45e47f52599f1cc2f97
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250180
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35911}
> >
> > Bug: webrtc:13540
> > Change-Id: Icb3e165f07240ae10730a316d3a8a3b2b9167d82
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251387
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35979}
>
> TBR=tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I4d8e3aced019215b97a6263cafa2a7488cd118be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:13540
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251661
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35983}

# Not skipping CQ checks because original CL landed > 1 day ago.

Using "no-try" since a follow-up revert is also needed to get the bots
to turn green.

No-try: true
Bug: webrtc:13540
Change-Id: I361fca6949c01200d9d706749e7e825eb5b4fc1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251685
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35991}
2022-02-14 12:35:10 +00:00
09aaf6f7bc Revert "Reland "Remove unused APM voice activity detection sub-module""
This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.

Reason for revert: Breaks chromium roll, see 
https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview

https://chromium-review.googlesource.com/c/chromium/src/+/3461512

Original change's description:
> Reland "Remove unused APM voice activity detection sub-module"
>
> This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.
>
> Reason for revert: dependency in a downstream project removed
>
> Original change's description:
> > Revert "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
> >
> > Reason for revert: breaking downstream projects
> >
> > Original change's description:
> > > Remove unused APM voice activity detection sub-module
> > >
> > > API changes:
> > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > - cricket::AudioOptions::typing_detection deprecated
> > > - webrtc::StatsReport::StatsValueName::
> > >   kStatsValueNameTypingNoiseState deprecated
> > >
> > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > >
> > > Bug: webrtc:11226,webrtc:11292
> > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35975}
> >
> > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> >
> > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:11226,webrtc:11292
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35977}
>
> # Not skipping CQ checks because this is a reland.
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35984}

TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35990}
2022-02-14 12:25:51 +00:00
eb6c6fcf27 Fix delta frame delay calculation
The issue was introduced in https://webrtc-review.googlesource.com/c/src/+/132460.

Bug: webrtc:10412
Change-Id: I92d1bd2be63ea34d150145cec63c282f7aa49ce8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251683
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35989}
2022-02-14 11:15:50 +00:00
6cd6d8ecfd Introduce Sync-Decoding based on Metronome
Adds new class DecodeSynchronizer that will coalesce the decoding
of received streams on the metronome. This feature is experimental and
is backed by a field trial WebRTC-FrameBuffer3.

This experiment now has 3 arms to it,

"WebRTC-FrameBuffer3/arm:FrameBuffer2/": Default, uses old frame buffer.
"WebRTC-FrameBuffer3/arm:FrameBuffer3/": Uses new frame buffer.
"WebRTC-FrameBuffer3/arm:SyncDecoding/": Uses new frame buffer with
frame scheduled on the metronome.

The SyncDecoding arm will not work until it is wired up in the follow-up
CL.

This change also makes the following modifications,
* Adds FakeMetronome utilities for tests using a metronome.
* Makes FrameDecodeScheduler an interface. The default implementation is
TaskQueueFrameDecodeScheduler.
* FrameDecodeScheduler now has a Stop() method, which must be called
before destruction.


TBR=philipel@webrtc.org

Change-Id: I58a306bb883604b0be3eb2a04b3d07dbdf185c71
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250665
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <holmer@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35988}
2022-02-14 11:14:00 +00:00
93348d89bc Remove unused audio options and corresponding media constraints
- experimental AGC (aka googAutoGainControl2) removed in [1]
- experimental NS (aka googNoiseSuppression2) removed in [2]
- typing noise detection (aka googTypingNoiseDetection)
  removed in [3]
- cricket::AudioOptions::tx_agc_ are unused

[1] https://webrtc-review.googlesource.com/c/src/+/219463
[2] https://webrtc-review.googlesource.com/c/src/+/232128
[3] https://chromium-review.googlesource.com/c/chromium/src/+/1617352

Bug: webrtc:11226
Change-Id: Id1ecef3d3e193c210fc11832e16db4f84d866d14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35987}
2022-02-14 10:50:20 +00:00
63472e5aea Replace dump_json_test_results flag by isolated-script-test-output.
This is because it's the default flag used in the recipes to dump a json output.
This CL also fixes some python3 lint issues in mb.py.

Bug: webrtc:13594
Change-Id: I9275b5da0963f801d3191703c2eb72d90befb5d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248142
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35986}
2022-02-14 08:55:01 +00:00
ea8dfd5011 Handle null structure for invalid scalability mode case
The scalability mode could be something invalid set by user, in this
case, |num_spatial_layers| should not be updated.

Bug: chromium:1292923
Change-Id: I78e1a6f12cf6d165597205608e4c124117a3d01b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/main@{#35985}
2022-02-14 03:00:39 +00:00
54d1344d98 Reland "Remove unused APM voice activity detection sub-module"
This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.

Reason for revert: dependency in a downstream project removed

Original change's description:
> Revert "Remove unused APM voice activity detection sub-module"
>
> This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
>
> Reason for revert: breaking downstream projects
>
> Original change's description:
> > Remove unused APM voice activity detection sub-module
> >
> > API changes:
> > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > - webrtc::AudioProcessingStats::voice_detected deprecated
> > - cricket::AudioOptions::typing_detection deprecated
> > - webrtc::StatsReport::StatsValueName::
> >   kStatsValueNameTypingNoiseState deprecated
> >
> > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> >
> > Bug: webrtc:11226,webrtc:11292
> > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35975}
>
> TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11226,webrtc:11292
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35977}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:11226,webrtc:11292
Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35984}
2022-02-13 14:02:08 +00:00
f342d6054a Revert "Reland "Use non-proxied source object in VideoTrack.""
This reverts commit 1158bff15f33c467543928dd6a49cb6ad04da1ba.

Reason for revert: This is a partial revert as we're tracking down
the source of the downstream issues. This CL reverts the use of
`internal()` for methods that relate to the source sink.

Original change's description:
> Reland "Use non-proxied source object in VideoTrack."
>
> This is a reland of 3eb29c12358930a60134f185cd849e0d12aa9166
>
> This reland doesn't contain the AudioTrack changes (see original
> description) that got triggered in some cases and needs to be
> addressed separately.
>
> Another change in this re-land is that instead of the `state` property
> of the VideoTrack be marshalled to the signaling thread, it's readable
> from the calling thread. Previously this was marshalled to the worker
> and the original changed that to the signaling thread (same as for
> AudioTrack) - but in case that's causing downstream problems this reland
> uses BYPASS_PROXY_CONSTMETHOD0 for the `state()` accessor of the
> VideoTrack proxy.
>
> Original change's description:
> > Use non-proxied source object in VideoTrack.
> >
> > Use the internal representation of the video source object from the
> > track. Before there were implicit thread hops due to use of the proxy.
> >
> > Also, override AudioTrack's enabled methods to enforce thread
> > expectations.
> >
> > Bug: webrtc:13540
> > Change-Id: I4bc7aca96d6fc24f31ade45e47f52599f1cc2f97
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250180
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35911}
>
> Bug: webrtc:13540
> Change-Id: Icb3e165f07240ae10730a316d3a8a3b2b9167d82
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251387
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35979}

TBR=tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I4d8e3aced019215b97a6263cafa2a7488cd118be
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:13540
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251661
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35983}
2022-02-12 16:28:55 +00:00
884e8ae640 Escape windows specific paths generated by os.path.join so that it works with regex.
Note that this will work on all platforms but is critical for windows due to the use backslash in the filesystem.

Bug: webrtc:13607
Change-Id: Ie9a9987f1382133792c85820d38b770fadc0fff5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251442
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35982}
2022-02-12 16:08:35 +00:00
075db39756 Check for valid transport before setting criteria.
Bug: chromium:1295469
Change-Id: Ib797cf1baf275e3ae1ce167cd4bfbc1161652000
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251620
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35981}
2022-02-11 18:18:27 +00:00
06c87a1664 Add tests for DC odd/even numbering in a few more cases.
Also expands integration_test_helpers to deal with multiple
datachannels.

The bug has not yet been triggered.

Bug: webrtc:13668
Change-Id: I82a0fdae0cc32815c250a691b56c614bfd6d606b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251602
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35980}
2022-02-11 17:37:34 +00:00
1158bff15f Reland "Use non-proxied source object in VideoTrack."
This is a reland of 3eb29c12358930a60134f185cd849e0d12aa9166

This reland doesn't contain the AudioTrack changes (see original
description) that got triggered in some cases and needs to be
addressed separately.

Another change in this re-land is that instead of the `state` property
of the VideoTrack be marshalled to the signaling thread, it's readable
from the calling thread. Previously this was marshalled to the worker
and the original changed that to the signaling thread (same as for
AudioTrack) - but in case that's causing downstream problems this reland
uses BYPASS_PROXY_CONSTMETHOD0 for the `state()` accessor of the
VideoTrack proxy.

Original change's description:
> Use non-proxied source object in VideoTrack.
>
> Use the internal representation of the video source object from the
> track. Before there were implicit thread hops due to use of the proxy.
>
> Also, override AudioTrack's enabled methods to enforce thread
> expectations.
>
> Bug: webrtc:13540
> Change-Id: I4bc7aca96d6fc24f31ade45e47f52599f1cc2f97
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250180
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35911}

Bug: webrtc:13540
Change-Id: Icb3e165f07240ae10730a316d3a8a3b2b9167d82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251387
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35979}
2022-02-11 16:03:14 +00:00
9a6097046e Add noop stubs for encoding/parsing all RTC event log events.
The actual event definitions will be added in upcoming CLs.

Bug: webrtc:11933
Change-Id: Ie10b08a71aeb12118612b7717a08b6acbc699c4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249361
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35978}
2022-02-11 15:00:14 +00:00
a751f167c6 Revert "Remove unused APM voice activity detection sub-module"
This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.

Reason for revert: breaking downstream projects

Original change's description:
> Remove unused APM voice activity detection sub-module
>
> API changes:
> - webrtc::AudioProcessing::Config::VoiceDetection removed
> - webrtc::AudioProcessingStats::voice_detected deprecated
> - cricket::AudioOptions::typing_detection deprecated
> - webrtc::StatsReport::StatsValueName::
>   kStatsValueNameTypingNoiseState deprecated
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35975}

TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35977}
2022-02-11 12:15:44 +00:00
9cc5fffee1 Convert a few more uses of rtc::split to use string_view
Bug: webrtc:13579
Change-Id: I84bdb908bf390924c6d67cd1c5aabcc9e62f33da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251581
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35976}
2022-02-11 11:31:54 +00:00
b4e06d032e Remove unused APM voice activity detection sub-module
API changes:
- webrtc::AudioProcessing::Config::VoiceDetection removed
- webrtc::AudioProcessingStats::voice_detected deprecated
- cricket::AudioOptions::typing_detection deprecated
- webrtc::StatsReport::StatsValueName::
  kStatsValueNameTypingNoiseState deprecated

PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0

Bug: webrtc:11226,webrtc:11292
Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35975}
2022-02-11 10:47:39 +00:00
cc5532f95a Use in-memory output instead of file in RTC event log test.
Bug: webrtc:13670
Change-Id: I17bf1dbc29b8e8ebfd019f213c4ae1e24a91d356
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251520
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35974}
2022-02-11 09:25:18 +00:00
321ec3ba99 Add tests for SSL role and datachannel ID assignment.
Document with a comment the suspected place that could cause a bug.
Also fix an error in previous role observation code.

Bug: webrtc:13668
Change-Id: Id7f6af6905d90f7974b5570145c201c8339aaf72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251388
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35973}
2022-02-10 15:52:16 +00:00
cb03e385ad Fix LOGGING_INSIDE_WEBRTC propagation in Chromium builds.
This macro needs to be both present in all WebRTC targets (see its
definition in at [1] but also propagated to all the targets
depending on the Chromium component defined in
//third_party/webrtc_overrides:webrtc_component (to properly support
transitive header #includes), by using "public_config" GN propagates
the macro accordingly.

[1] - https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/BUILD.gn;l=315;drc=61dbc2db2b84eed9c9769c1b79070e6bd4030331

Bug: None
Change-Id: Idd51643da63be48324c86a5b89676c63c3998e14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251440
Reviewed-by: Björn Terelius <terelius@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35972}
2022-02-10 15:20:16 +00:00
316ab12821 Make DTLS role visible on DtlsTransport interface
This is important for writing tests that affect the DTLS role.

Bug: webrtc:13668
Change-Id: I5d9a93eca7996a8b74cdcfe412f59a85892e1ec1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251389
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35971}
2022-02-10 11:04:36 +00:00
e6aa6a8740 Set file mode in the argparse argument
Bug: webrtc:13607
Change-Id: I7943761933e0e110ff16b65284d8b36644b5c4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251381
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35970}
2022-02-10 09:02:26 +00:00
1a41178e33 scoped_glib: Fix ODR violation
Moving the template specialization into the header causes ODR
violation when the header file is included in other units. Making
the specialization inline to avoid this problem.

Bug: chromium:1291247
Change-Id: I090548c1c3dd07a8c46b87ae90ebdd45a60a5cde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251200
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35969}
2022-02-09 21:52:26 +00:00
9b3c792f67 screencast_portal.h: Remove unused typedef
Minor cleanup to remove unused typedef.

Bug: chromium:1291247
Change-Id: Idbbe8dba13d4d14888f843ae170a898ff604852b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249700
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#35968}
2022-02-09 18:52:55 +00:00
20d8d9150c Reland "Remove stopped_ from AudioRtpReceiver and VideoRtpReceiver."
This is a reland of 3ed36c0521546881656c73984456485dcab16205

Original change's description:
> Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver."
>
> This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f
>
> The difference from the original CL is that a check for
> `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
> This caused a side effect that registering the sink while the source
> was in an "initializing" state, failed. The last remaining state
> however, is `kEnded` - but since there's no logic in the class around
> the expected value of the states, the check inside of AddSink()
> doesn't provide an additional value - it's rather a surprise for
> developers if it doesn't succeed. So, now removed.
>
> Original change's description:
> > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
> >
> > This simplifies the logic in these classes a bit, which makes upcoming
> > change easier. The `stopped_` flag in these classes was essentially
> > the same thing as `media_channel_ == nullptr`, which is what's
> > consistently used now for the same checks.
> >
> > Bug: webrtc:13540
> > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35907}
>
> Bug: webrtc:13540
> Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35958}

Bug: webrtc:13540
Change-Id: I6d7d67fddb1ddfc69a302f0f69a9b815f2fd82f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251386
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35967}
2022-02-09 16:54:06 +00:00
981c572eab Updated apply-iwyu to autogenerate compile_commands.json
Also deleted iwyu script that was not maintained, and deleted
some options that made the script more complex.

Bug: none
Change-Id: I39d8eaa37f12c72ddc127ae145e6a3a80f328316
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251384
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35966}
2022-02-09 14:30:57 +00:00
bc32c56f83 Move pc.transport_controller_ to be network thread only
A pointer to the transport controller is now maintained on
both the network thread and the signaling thread. We use
thread specific accessors to make it explicit which copy we
are accessing at any given time.

We also move the initial offerer value to the SDP offer/answer
class; this is determined on the basis of SDP offer/answer, so
there is no need to hop to the network thread for that.

Work in progress.

Bug: webrtc:9987
Change-Id: Idbe5a7fbf44f667adcd119e486133cf6e43ab1f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251382
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35965}
2022-02-09 13:06:15 +00:00
ffdc6804bf Reland: Added support for H264 YUV444 (I444) decoding.
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/235340

Bug: chromium:1251096
Change-Id: Icd997c7f7732229954d5494343b4e7a70deb09d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251303
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35964}
2022-02-09 11:57:55 +00:00
2da85916ab Revert "Reland "Remove stopped_ from AudioRtpReceiver and VideoRtpReceiver.""
This reverts commit 3ed36c0521546881656c73984456485dcab16205.

Reason for revert: Breaks downstream project.

Original change's description:
> Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver."
>
> This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f
>
> The difference from the original CL is that a check for
> `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
> This caused a side effect that registering the sink while the source
> was in an "initializing" state, failed. The last remaining state
> however, is `kEnded` - but since there's no logic in the class around
> the expected value of the states, the check inside of AddSink()
> doesn't provide an additional value - it's rather a surprise for
> developers if it doesn't succeed. So, now removed.
>
> Original change's description:
> > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
> >
> > This simplifies the logic in these classes a bit, which makes upcoming
> > change easier. The `stopped_` flag in these classes was essentially
> > the same thing as `media_channel_ == nullptr`, which is what's
> > consistently used now for the same checks.
> >
> > Bug: webrtc:13540
> > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35907}
>
> Bug: webrtc:13540
> Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35958}

TBR=ilnik@webrtc.org,tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Ieb7235d88c808c78ad0847403be991d4dce1ace6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:13540
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251383
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35963}
2022-02-09 10:55:25 +00:00
b02220d1a0 Reland "Mark all bool conversion operators as explicit"
This is a reland of 325789c4576b60147ee1ef225d438cbb740f65ff

Original change's description:
> Mark all bool conversion operators as explicit
>
> An explicit bool conversion operator will still be used implicitly
> when an expression appears in "bool context", e.g., as the condition
> in an if statement, or as argument to logical operators. The
> `explicit` annotation prevents conversion in other contexts, e.g.,
> converting both a and b to bool in an expression like `a == b`.
>
> Bug: None
> Change-Id: I79ef35b1ea831e6011ae472900375ae8a3e617ab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250664
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35927}

Bug: None
Change-Id: Ie057dfc8c0b5c498e2c8daff7620172c89f0e011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35962}
2022-02-09 09:40:05 +00:00
15ad4ed676 Break out peer_connection_factory and peer_connection
The peerconnection target now has no files, which means that no
target that needs .h files depends on it. This is good.

Bug: webrtc:13634
Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 07:55:35 +00:00
c98fb2ca0f Use binary mode for proto ingestion
Bug: webrtc:13607
Change-Id: Id0385f74215360ff604641a50ce9f599c87abb5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251327
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35960}
2022-02-09 07:19:35 +00:00
d12a14e21f Add new RTC event log encoding for AudioPlayout and DelayBasedBwe events.
Bug: webrtc:11933
Change-Id: Ia54d973099916c8dba9fedf362f25e46fe5cc541
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246204
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35959}
2022-02-08 23:09:24 +00:00
3ed36c0521 Reland "Remove stopped_ from AudioRtpReceiver and VideoRtpReceiver."
This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f

The difference from the original CL is that a check for
`state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
This caused a side effect that registering the sink while the source
was in an "initializing" state, failed. The last remaining state
however, is `kEnded` - but since there's no logic in the class around
the expected value of the states, the check inside of AddSink()
doesn't provide an additional value - it's rather a surprise for
developers if it doesn't succeed. So, now removed.

Original change's description:
> Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
>
> This simplifies the logic in these classes a bit, which makes upcoming
> change easier. The `stopped_` flag in these classes was essentially
> the same thing as `media_channel_ == nullptr`, which is what's
> consistently used now for the same checks.
>
> Bug: webrtc:13540
> Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35907}

Bug: webrtc:13540
Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35958}
2022-02-08 21:41:14 +00:00
f4cad8ac51 PipeWire capturer: drop DMA-BUF modifier and renegotiate parameters on failure
In case we fail to import a DMA-BUF with given modifier, we can try to
drop the modifier we failed to use and renegotiate stream parameters
in order to use a different modifier or fallback to shared memory buffers.

Bug: chromium:1290566
Change-Id: I617513bdd67a43f62b647a172e0c166af138b3f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249798
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35957}
2022-02-08 20:38:54 +00:00