2494 Commits

Author SHA1 Message Date
Sergey Silkin
6432970fe9 Make it possible to set spatial layer bitrates explicitly
Bug: webrtc:14852
Change-Id: Ie41d4223d0d5aef5a79f7e6067f0855f022ed428
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335361
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41698}
2024-02-08 14:42:08 +00:00
Harald Alvestrand
974044efca Remove code for supporting SDES
Rework transport_description_factory to only have non-DTLS mode for
testing, and rewrite tests accordingly.

Bug: webrtc:11066, chromium:804275
Change-Id: Ie7d477c4331c975e4e0a3034fbbb749ed9009446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41697}
2024-02-08 14:34:04 +00:00
Tomas Lundqvist
aaa123debb Reland "Remove post-decode VAD"
This is a reland of commit 89cf26f1e0532130745f648cf16b1fb8af2f6b4f

Original change's description:
> Remove post-decode VAD
>
> Bug: webrtc:15806
> Change-Id: I6acf8734a70703085cfc1ccf82a79ee0931f59a4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336460
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Tomas Lundqvist <tomasl@google.com>
> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41653}

Bug: webrtc:15806
Change-Id: I1c2c0ce568c3c1817ff5c65bee91b9f961d46559
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337442
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41688}
2024-02-07 16:33:51 +00:00
Harald Alvestrand
3bddaed569 rtc_p2p: Split turn port and basic port allocator
This completes the breakup of the rtc_p2p target.
Remaining cleanup is to delete the rtc_p2p target and make clients
depend on the base targets.

Bug: webrtc:15796
Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-07 10:30:59 +00:00
Jakob Ivarsson
9d9b3a3553 Add option for the audio encoder to allocate a bitrate range.
Bug: webrtc:15834
Change-Id: I3223162e747621983dbe2704661bb87f7890b3fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338221
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#41682}
2024-02-07 09:47:16 +00:00
Harald Alvestrand
8bb54c1c42 Penultimate split-up of rtc_p2p build target
This takes the rest of the .cc files out of the rtc_p2p build
target, leaving only one entangled target to clean up.

Bug: webrtc:15796
Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 17:52:39 +00:00
Philipp Hancke
bda5cc63ce Clean up use of WebRTC-UseStandardBytesStats trial in tests
BUG=webrtc:10525

Change-Id: Ia0ec88d5b561ec98af540f849182805d49a327e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337520
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41663}
2024-02-05 09:24:01 +00:00
Jeremy Leconte
687ef0a136 Revert "Remove post-decode VAD"
This reverts commit 89cf26f1e0532130745f648cf16b1fb8af2f6b4f.

Reason for revert: breaking upstream projects

Original change's description:
> Remove post-decode VAD
>
> Bug: webrtc:15806
> Change-Id: I6acf8734a70703085cfc1ccf82a79ee0931f59a4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336460
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Tomas Lundqvist <tomasl@google.com>
> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41653}

Bug: webrtc:15806
Change-Id: I20e383a6b6d625d86830ecec1be01b42b22e86a2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337420
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41657}
2024-02-01 15:16:26 +00:00
Tomas Lundqvist
89cf26f1e0 Remove post-decode VAD
Bug: webrtc:15806
Change-Id: I6acf8734a70703085cfc1ccf82a79ee0931f59a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336460
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41653}
2024-02-01 12:37:23 +00:00
Alfred E. Heggestad
765024e67b test: fix fuzzers line-endings
Bug: None
Change-Id: I95edb5482bfc9cfc7241963bbe43a3873aa814ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335143
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41633}
2024-01-30 08:40:40 +00:00
Danil Chapovalov
62cee88e4b Propagate Environment through QualityAnalyzingVideoDecoderFactory
Bug: webrtc:15791
Change-Id: I9eddf7bf9fb66ee70495e9bc3c810126e2015287
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336800
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41630}
2024-01-29 20:11:46 +00:00
qwu16
f43e8ebab9 Add RTP depacketizer for H265
1. Depacketize single nalu packet/AP/FU

2. Insert start code before each nalu

Bug: webrtc:13485
Change-Id: I8346f9c31e61e5d3c2c7e1bf5fdaae4018a1ff78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325660
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41628}
2024-01-29 12:00:19 +00:00
Tony Herre
9c6874607a Consolidate encoded transform mocks into api/test/
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/

Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
2024-01-26 12:46:34 +00:00
Per K
9c166e064f Remove VideoSendStream::StartPerRtpStream
Instead, always use VideoSendStream::Start.

VideoSendStream::StartPerRtpStream was used for controlling if
individual rtp stream for a RtpEncodingParameter should be able to send RTP packets. It was not used for controlling the actual encoder layers.

With this change RtpEncodingParameter.active still controls actual encoder layers but it does not control if RTP packets can be sent or not.

The cleanup is done to simplify code and in the future allow sending
probe packet on a RtpTransceiver that allows sending, regardless of the
RtpEncodingParameter.active flag.

Bug: webrtc:14928
Change-Id: I896c055ed4de76db58d76f452147c29783f77ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335042
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41619}
2024-01-26 09:19:50 +00:00
Danil Chapovalov
d213dd5517 Pass Environment to VideoDecoders through VideoCodecTester
Bug: webrtc:15791
Change-Id: I002734a17ece1d11b77a261aa8160c4afa1702b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336241
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41617}
2024-01-26 08:11:19 +00:00
Harald Alvestrand
a310d78662 Refactor a lot of the p2p:rtc_p2p target
This CL splits many of the source files in p2p:rtc_p2p into individual
compile targets.

One target - connection_and_port - was left with multiple source files
because it was too tangled to detangle at once.

Bug: webrtc:15796
Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 18:28:27 +00:00
Sergey Silkin
37e9b378fd Use default H264 SDP parameters
We lost H264 [1] in https://webrtc-review.googlesource.com/c/src/+/327260 where we started using QueryCodecSupport which is sensetive to SDP parameters.

Use CBP3.1, packetization_mode=1 (singlecast NALU) as defaults.

[1] https://chromeperf.appspot.com/report?sid=1e12d661147889123ddeea4ef88a87bcdd38cf09cb23c13ee130770be695ac83&start_rev=41064&end_rev=41226

Bug: webrtc:14852, webrtc:15779
Change-Id: I69137ac847ae3a79238abcfe2a76dc2ba097a06d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335081
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41576}
2024-01-19 15:01:12 +00:00
Henrik Boström
7209548090 Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."
This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb

Previously reverted due to an importer issue (b/320646178) and later a
dependency on RTCStatsMember<T>::ValueToString().

In this reland, we add Attribute::ToString() but we don't delete the
RTCStatsMember<T> stringifier methods, allowing downstream to migrate
before they are deleted.

Original change's description:
> [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
>
> Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> Attribute::ToString().
>
> The difference between "ToString" and "ToJson" is that the "ToJson"
> version converts 64-bit integers and doubles to floating points with no
> more than ~15 digits of precision as to not exceed JSON's precision
> limitations. So only in edge cases of really large numbers or numbers
> with a silly number of digits will the two methods produce different
> results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> as opposed to "{foo:123}".
>
> Going forward we see no reason to maintain two different string
> converted paths that are this similar, so we only implement one
> Attribute::ToString() method which does what "ToJson" did.
>
> In the next CL we can delete RTCStatsMember<T>.
>
> Bug: webrtc:15164
> Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41544}

Bug: webrtc:15164
Change-Id: I281ccf5b23d8f194b5ce00186a32846c757b46fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41575}
2024-01-19 14:42:10 +00:00
Henrik Boström
ed1d084d0a [Stats] Replace all uses of is_defined() with has_value().
Same method, different name. Unblocks replacing RTCStatsMember<T> with
absl::optional<T>.

Bug: webrtc:15164
Change-Id: I251dd44d3b0f9576b3b68915fe0406d1b3381e5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41573}
2024-01-19 12:26:56 +00:00
Danil Chapovalov
f49d96d6e4 Remove usage of the rtc::TaskQueue in test/
Bug: webrtc:14169
Change-Id: Ie95973e5f58ee203c13243866782696ed14de908
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335144
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41569}
2024-01-19 10:04:54 +00:00
Mirko Bonadei
4d706a9fd1 Revert "Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson.""
This reverts commit 55cdc29b9d7259d17ccc281855dd21adc51ca957.

Reason for revert: Breaks downstream project.

Original change's description:
> Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."
>
> This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb
>
> Original change's description:
> > [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
> >
> > Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> > Attribute::ToString().
> >
> > The difference between "ToString" and "ToJson" is that the "ToJson"
> > version converts 64-bit integers and doubles to floating points with no
> > more than ~15 digits of precision as to not exceed JSON's precision
> > limitations. So only in edge cases of really large numbers or numbers
> > with a silly number of digits will the two methods produce different
> > results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> > as opposed to "{foo:123}".
> >
> > Going forward we see no reason to maintain two different string
> > converted paths that are this similar, so we only implement one
> > Attribute::ToString() method which does what "ToJson" did.
> >
> > In the next CL we can delete RTCStatsMember<T>.
> >
> > Bug: webrtc:15164
> > Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> > Commit-Queue: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Evan Shrubsole <eshr@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41544}
>
> Bug: webrtc:15164
> Change-Id: If34509ebf3d7c0291442ae11596e7c2d3978fb64
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335240
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41566}

Bug: webrtc:15164
Change-Id: I5819811237a6dbd85a8c738ca0180039fc705909
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335280
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41567}
2024-01-19 08:03:48 +00:00
Henrik Boström
55cdc29b9d Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."
This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb

Original change's description:
> [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
>
> Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> Attribute::ToString().
>
> The difference between "ToString" and "ToJson" is that the "ToJson"
> version converts 64-bit integers and doubles to floating points with no
> more than ~15 digits of precision as to not exceed JSON's precision
> limitations. So only in edge cases of really large numbers or numbers
> with a silly number of digits will the two methods produce different
> results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> as opposed to "{foo:123}".
>
> Going forward we see no reason to maintain two different string
> converted paths that are this similar, so we only implement one
> Attribute::ToString() method which does what "ToJson" did.
>
> In the next CL we can delete RTCStatsMember<T>.
>
> Bug: webrtc:15164
> Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41544}

Bug: webrtc:15164
Change-Id: If34509ebf3d7c0291442ae11596e7c2d3978fb64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335240
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41566}
2024-01-19 07:21:51 +00:00
Henrik Boström
df0b363cf0 Reland "[Stats] Add value_or() and migrate from ValueOrDefault()."
This is a reland of commit 9e4a97bb02663604b02e219b9d501a8dd91b5614

Original change's description:
> [Stats] Add value_or() and migrate from ValueOrDefault().
>
> Yet another prerequisite for replacing RTCStatsMember<T> with
> absl::optional<T>, but this looks like the last one.
>
> Bug: webrtc:15164
> Change-Id: I2cde51e8c8c951f71b48ccd45e07146091a99616
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334647
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41541}

Bug: webrtc:15164
Change-Id: I5fdba499383e5d9efe0a1dcef6bf6c2e0a812857
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335102
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41564}
2024-01-18 22:07:18 +00:00
Mirko Bonadei
111e381822 Revert "[Stats] Add value_or() and migrate from ValueOrDefault()."
This reverts commit 9e4a97bb02663604b02e219b9d501a8dd91b5614.

Reason for revert: Breaks downstream project

Original change's description:
> [Stats] Add value_or() and migrate from ValueOrDefault().
>
> Yet another prerequisite for replacing RTCStatsMember<T> with
> absl::optional<T>, but this looks like the last one.
>
> Bug: webrtc:15164
> Change-Id: I2cde51e8c8c951f71b48ccd45e07146091a99616
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334647
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41541}

Bug: webrtc:15164
Change-Id: I89af6470c82d07981d8d064aa6ff8b50fae42b12
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334801
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41548}
2024-01-17 14:38:25 +00:00
Mirko Bonadei
1fee69cfff Revert "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."
This reverts commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb.

Reason for revert: Breaks downstream project.

Original change's description:
> [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
>
> Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> Attribute::ToString().
>
> The difference between "ToString" and "ToJson" is that the "ToJson"
> version converts 64-bit integers and doubles to floating points with no
> more than ~15 digits of precision as to not exceed JSON's precision
> limitations. So only in edge cases of really large numbers or numbers
> with a silly number of digits will the two methods produce different
> results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> as opposed to "{foo:123}".
>
> Going forward we see no reason to maintain two different string
> converted paths that are this similar, so we only implement one
> Attribute::ToString() method which does what "ToJson" did.
>
> In the next CL we can delete RTCStatsMember<T>.
>
> Bug: webrtc:15164
> Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41544}

Bug: webrtc:15164
Change-Id: I187d7dff6f330a4a440279e6c32d88eb6ddefac8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334820
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41546}
2024-01-17 14:06:34 +00:00
Henrik Boström
54be7084e0 [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
Attribute::ToString().

The difference between "ToString" and "ToJson" is that the "ToJson"
version converts 64-bit integers and doubles to floating points with no
more than ~15 digits of precision as to not exceed JSON's precision
limitations. So only in edge cases of really large numbers or numbers
with a silly number of digits will the two methods produce different
results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
as opposed to "{foo:123}".

Going forward we see no reason to maintain two different string
converted paths that are this similar, so we only implement one
Attribute::ToString() method which does what "ToJson" did.

In the next CL we can delete RTCStatsMember<T>.

Bug: webrtc:15164
Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41544}
2024-01-17 12:36:46 +00:00
Henrik Boström
9e4a97bb02 [Stats] Add value_or() and migrate from ValueOrDefault().
Yet another prerequisite for replacing RTCStatsMember<T> with
absl::optional<T>, but this looks like the last one.

Bug: webrtc:15164
Change-Id: I2cde51e8c8c951f71b48ccd45e07146091a99616
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334647
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41541}
2024-01-17 10:35:14 +00:00
Danil Chapovalov
972546ed50 Delete TestPeerFactory constructor that uses rtc::TaskQueue
Bug: webrtc:14169
Change-Id: Id1414ecc19c8ff9da826688684003560d9a3139d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334642
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41536}
2024-01-16 17:29:58 +00:00
Danil Chapovalov
e052eee7a3 Deprecate rtc::TaskQueue variant of AudioProcessing::CreateAndAttachAecDump
Bug: webrtc:14169
Change-Id: I63f40ec18b72cba89eb0b9b298f448ce7f7c4634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334201
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41528}
2024-01-15 13:36:35 +00:00
Sergey Silkin
2ab1997d9d Fix a crash in video codec tester
Bug: webrtc:14852
Change-Id: I282fd41f2c2486b4b788581221bf9811f6e918ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334221
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41514}
2024-01-12 11:15:07 +00:00
Sergey Silkin
5aea42860b Fix H264 simulcast in codec tester
It only worked with VP8 before.

Tested: out/debug/video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=openh264 --decoder=ffmpeg-h264 --num_frames=30 --scalability_mode=S2T2 --dump_encoder_output -> 360p and 720p streams with two temporal layers each were produced. Bitrate allocation across temporal layers is done by OpenH264 encoder (no API to control this).

Bug: webrtc:14852
Change-Id: I58e2e1f595bdd6653701a97874766752bd2e3d58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332342
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41508}
2024-01-12 08:23:03 +00:00
Tommi
c56052001d JsepTransportController: Remove raw pointers to description objects
Remove member variables that point to objects owned externally (in practice by SdpOfferAnswerHandler). The objects also live on the
signaling thread whereas JsepTransportController performs
operations on the network thread. Removing the raw pointers avoids
the risk of referencing the description objects after they've been
deleted or if the state is inconsistent across threads.

Bug: webrtc:1515832
Change-Id: I852b2a3993964be817f93c46b5bc4b03121cde86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334061
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41505}
2024-01-12 00:14:43 +00:00
Philipp Hancke
bb0044eb90 add VP8/VP9 packetization fuzzers
and ensure consistent behavior on empty input.

BUG=webrtc:15755

Change-Id: Id70ab5d55251b4dd10eed8ab67ea8e75545a7a8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332740
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41502}
2024-01-10 14:36:46 +00:00
henrika
b7ec05777a FrameCadenceAdapter: now sets queue_overload based on encoder load
Measures the time consumed by OnFrame (e.g. the encoding time) and
sets an overload flag during N subsequent frames if the time is
longer than the current frame time. N is set to the number of
received frames on the network thread while being blocked by
encoding.

The queue overload mechanism for zero hertz can be disabled using the
WebRTC-ZeroHertzQueueOverload kill switch.

Also adds a UMA called WebRTC.Screenshare.ZeroHz.QueueOverload.

Bug: webrtc:15539
Change-Id: If81481c265d3e845485f79a2a1ac03dcbcc3ffc3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332381
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41489}
2024-01-09 14:29:04 +00:00
Philipp Hancke
5d091cec5d Add H264 packetizer fuzzer
BUG=webrtc:15755

Change-Id: I384fbdfa3a2aea8faaf53eb161cecc2c8639401d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332600
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41487}
2024-01-09 13:32:42 +00:00
Danil Chapovalov
956879d86f Remove usage of rtc::TaskQueue in TimeController unittests
Bug: webrtc:14169
Change-Id: Ib3769bffe4642d9466c65dee3d46d81049642f40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333803
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41486}
2024-01-09 13:31:39 +00:00
Danil Chapovalov
0a96289bcb Replace legacy rtc::TaskQueue with TaskQueueBase in test Network helper
rtc::TaskQueue wrapper provide no extra feature there.

Bug: webrtc:14169
Change-Id: I91ec99e10400627f1fb6f534cc679e3b11534c0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333481
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41484}
2024-01-09 13:08:56 +00:00
Danil Chapovalov
55a61898a8 Pass Environment to custom FecController at construction
To allow custom FecController use propagated rather than global field trials
note that there is one FecControllerFactory per peer connection factory,
but FecController is created per peer connection and may use per peer connection field trials.

Bug: webrtc:10335
Change-Id: Id25bfaf4b49d4f6d551730c8fd55596ddc49ab47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41478}
2024-01-08 12:57:48 +00:00
Danil Chapovalov
b64eef1234 In AecDump take raw pointer to TaskQueueBase instead of legacy rtc::TaskQueue
Bug: webrtc:14169
Change-Id: I1e50a945a7637da07bec00ccd7b6b1847a7481cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333480
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41477}
2024-01-08 12:17:06 +00:00
Philipp Hancke
de17252e8e Reland "Unify access to SDP codec parameters"
This is a reland of commit 63d03f586bb668f72113b61030ec0930aa192010

Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
>   std::map<std::string, std:string>
> with three aliases,
>   cricket::CodecParameterMap
>   SdpAudioFormat::Parameters
>   SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}

Bug: None
Change-Id: I5f8f45688df232eb37b12fa3e56a893a1c754e17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331402
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41467}
2024-01-03 12:03:11 +00:00
Dor Hen
764ac7ec0a Allowing to set PCF options via peer configurer
Bug: webrtc:15752
Change-Id: I408cf2e118d09504d59a09ef4c2767ab89982db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332645
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41462}
2024-01-02 10:59:05 +00:00
Sergey Silkin
e6f244e003 Write codec FourCC to IVF dump
Bug: webrtc:14852
Change-Id: I00bf828bf4d3f38bf6215320dec2189202a0b2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331520
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41441}
2023-12-22 15:50:39 +00:00
Sergey Silkin
0a01ffcd3f Add SVC support to the video codec tester
Bug: webrtc:14852
Change-Id: Iaa060fea396b8ec317d8f20d0c1bdad21bf739db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331502
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41440}
2023-12-22 15:18:53 +00:00
Sergey Silkin
9e5c979743 Add simulcast support to the video codec tester
Create one decoder per simulcast stream and pass encoded frame to a dedicated decoder.

Bug: webrtc:14852
Change-Id: I2a0baaa1e28b38507993eb4269b15ae89695d670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41439}
2023-12-22 14:40:36 +00:00
Sergey Silkin
66344aca9c Update video codec tester unit tests
* Add VideoCodecTesterTest and move Run[EncodeDecode]Test() into it. The class will be extended with functionality necessary for testing simulcast/SVC (it will collect and store decode input frame sizes in particular) in follow-up CLs, which will add simulcast/SVC support to the tester.
* Add TestVideoEncoder and TestVideoDecoder classes.
* Use frame size instead of timestamp in checks in Slice test. Unlike timestamp, which has the same value for spatial layer frames within a temporal unit, frame size is a unique frame property in these tests.

Bug: webrtc:14852
Change-Id: I2386183688dd4988ca56e0ab53edbb9f5fcf6c9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331362
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41438}
2023-12-22 11:02:20 +00:00
Mirko Bonadei
6c9c958c69 Revert "Unify access to SDP codec parameters"
This reverts commit 63d03f586bb668f72113b61030ec0930aa192010.

Reason for revert: Breaks downstream project (not backwards compatible API change)

Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
>   std::map<std::string, std:string>
> with three aliases,
>   cricket::CodecParameterMap
>   SdpAudioFormat::Parameters
>   SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}

Bug: None
Change-Id: I841735d98533d3b66850b9cfcf7ee0a99ddde078
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41377}
2023-12-13 16:28:44 +00:00
Christoffer Jansson
399a9768f2 remove landrey@ from OWNERS
Bug: NONE
Change-Id: I515ca02376087656167cdbe234250b1a0100abbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331380
Auto-Submit: Christoffer Dewerin <jansson@google.com>
Commit-Queue: Christoffer Dewerin <jansson@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41376}
2023-12-13 16:10:11 +00:00
Philipp Hancke
63d03f586b Unify access to SDP codec parameters
which come from the a=fmtp:<pt> lines in the SDP and were used as either
  std::map<std::string, std:string>
with three aliases,
  cricket::CodecParameterMap
  SdpAudioFormat::Parameters
  SdpVideoFormat::Parameters

Use webrtc::CodecParameterMap in all places.

BUG=None

Change-Id: If47692bde7347834c349c6539b43309d8770e67b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41375}
2023-12-13 14:22:15 +00:00
Danil Chapovalov
7b4b39809f Remove DCHECK when transport feedback on request can't be produced
Bug: chromium:1507210
Change-Id: I840b91dd7143ce6a0d3c9a17df6c187e01a145f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330320
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41351}
2023-12-11 11:15:47 +00:00
Danil Chapovalov
151003d341 Deprecate RtcEventLogFactory constructor taking unused parameter
Bug: webrtc:15656
Change-Id: I22ed4cca4c0ce7ebf9c533ed7434617bf0a0f4a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41338}
2023-12-07 21:46:56 +00:00