611 Commits

Author SHA1 Message Date
Per K
569849e885 Move call/simulated_network to test/network
Old target and call/simulated.h exist but refer to new target in test/network.

Bug: webrtc:14525
Change-Id: Ida04cef17913f2f829d7e925ae454dc40d5e8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349264
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42191}
2024-04-29 09:55:06 +00:00
Florent Castelli
0afde7614d Move webrtc::AudioProcessing include to api/ folder
Bug: webrtc:15874
Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-20 07:02:50 +00:00
Per K
29abba982c Cleanup WebRTC-SendPacketsOnWorkerThread
Experiment has been concluded and cleaned up.

Bug: webrtc:14502
Change-Id: I7f892538dc676056ca2e8969a1ef81ffa3d40014
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347645
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42095}
2024-04-17 11:20:58 +00:00
Victor Boivie
cdecc4e6df Expose bufferedAmountLowThreshold
This code was extracted to make the next following CL easier to review.

This CL simply exposes the getters, setters and callbacks to set the
buffered amount low threshold on a specific SCTP stream. It will be
used in a follow-up CL, but is just boilerplate.

Bug: chromium:40072842
Change-Id: Iccd72208b369ddc252cc5886f6446b9c2ceeb0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343360
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41943}
2024-03-21 19:59:39 +00:00
Jeremy Leconte
1a37aa197e Fix frame not found error when encoder is paused.
The problem occurs when a frame is sent again because the encoder was paused but the frame has already been received by all participants:
https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;l=2322

Change-Id: If8890986301c44a472db9bc4750d23761c150669
Bug: b/328175783
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343560
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41931}
2024-03-19 17:16:19 +00:00
Danil Chapovalov
dcc95081e1 Cleanup QualityAnalyzingVideoEncoderFactory::CreateVideoEncoder
And thus require Environment to be propagated to this test helper

Bug: webrtc:15860
Change-Id: Ia4796d7a6a8e6f5dcb947899617df43e991419e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343181
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41910}
2024-03-15 15:24:54 +00:00
Victor Boivie
fea41f540c pc: Include SCTP queued bytes in buffered_amount
Before this change, calling buffered_amount only included what was
buffered on top of what was already buffered in the SCTP socket. With
the defaults, the SCTP socket can buffer up to 2MB of data (that is not
put on the wire) before the additional external bufferering in
SctpDataChannel will be used. The buffering that I am working on
removing completely.

Until it's removed completely, to avoid the issue reported in
crbug.com/41221056, include the bytes buffered in the SCTP socket to
what is returned when calling RTCDataChannel::buffered_amount.

This means that when this value is zero, it can be safe to know that all
bytes have been sent, but not necessarily acknowledged. And calling
close will not discard any messages.

This is a stopgap solution, but as functional as the proper solution
that removes all additional buffering. Follow-up CLs will merely improve
this solution.

Bug: chromium:41221056
Change-Id: I06edd52188d3bf13a17827381a15a4730722685a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41898}
2024-03-13 15:44:17 +00:00
Per K
0fa90887c5 Deprecate VideoFrame::timestamp() and set_timestamp
Instead, add rtp_timestamp and set_rtp_timestamp.

Bug: webrtc:13756
Change-Id: Ic4266394003e0d49e525d71f4d830f5e518299cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342781
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41894}
2024-03-13 11:08:37 +00:00
Harald Alvestrand
afaae4e38a Remove remaining .cc files from rtc_media_base
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.

Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 14:09:38 +00:00
Danil Chapovalov
329f0ead43 Provide Environment when creating VideoEncoder in test code
Bug: webrtc:15860
Change-Id: I8c79ff58619716842e02f33e78a0529c631494e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342280
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41884}
2024-03-12 11:09:31 +00:00
Victor Boivie
cd54fd8606 sctp: Pass webrtc::Environment to DcSctpTransport
The DcSctpTransport will soon use field trials to conditionally enable
some options.

And overall, there is a migration project to start using the Environment
and this CL is in that direction, also setting the boundary; The dcSCTP
library should not depend on it. But the transport is allowed to.

Bug: webrtc:14997
Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41872}
2024-03-08 09:45:12 +00:00
Philipp Hancke
bbff58d935 Introduce "well-known" SdpVideoFormat codecs
describing video codecs with their parameters as static members of SdpVideoFormat:
  static const SdpVideoFormat VP8();
  static const SdpVideoFormat H264();
  static const SdpVideoFormat VP9Profile0();
  static const SdpVideoFormat VP9Profile1();
  static const SdpVideoFormat VP9Profile2();
  static const SdpVideoFormat VP9Profile3();
  static const SdpVideoFormat AV1Profile0();
  static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.

BUG=webrtc:15703

Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41833}
2024-02-28 06:57:10 +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
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
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
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
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
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
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
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
Danil Chapovalov
3d9c3687a4 Delete CallFactoryInterface as no longer needed
Replace CallFactory class with a factory function

Bug: webrtc:15574
Change-Id: Ib1d8cff8d7550da3af01693a7bc117a7bd342258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330000
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41321}
2023-12-05 15:44:43 +00:00
Danil Chapovalov
4fd1cc70da Add EnableMediaWithDefaultsAndTimeController
To replace CreateTimeControllerBasedCallFactory

Update webrtc tests to use this new function

Bug: webrtc:15574
Change-Id: I2b74cd930ecc4f72dd1e7aa853764ca298b66ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325527
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41076}
2023-11-03 13:26:32 +00:00
Danil Chapovalov
554f7db01c Add EnableMediaWithDefaults to replace SetMediaEngineDefaults
Update most of the webrtc tests to use EnableMediaWithDefaults instead of SetMediaEngineDefaults

Bug: webrtc:15574
Change-Id: I489a09e4ea3479dc26829ee0c1235e67bcbca7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325485
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41059}
2023-11-01 11:47:59 +00:00
Danil Chapovalov
1d586debab In PCLF remove ability to inject TaskQueueFactory and CallFactory
Instead rely on TaskQueueFactory and Clock provided by the internal TimeController of the PCLF framework.

Bug: webrtc:15574
Change-Id: I473e1f12ead97f866dbd45771ed5a59541c0c47c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325182
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41026}
2023-10-27 13:03:09 +00:00
Danil Chapovalov
f2443a7971 Replace WebRTC-QuickPerfTest field trial with a flag
This field trial is configured via command line flag, so may use flag system directly, reducing dependency on global field trial string.

Bug: webrtc:7101, webrtc:10335
Change-Id: I1e48e0e3fdc251b73a375c6d7f1a46fa4f8a179b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322624
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40897}
2023-10-10 08:59:10 +00:00
Danil Chapovalov
2d508f10d3 Deprecate old names for EncodedImage::RtpTimestamp accessor and setter
Replace remaining webrtc usage of the deprecated names.

Bug: webrtc:9378
Change-Id: Ie5bd2d3eaf68316e7c827fc35c7c7d8e2eadeb9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321584
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40824}
2023-09-28 07:29:22 +00:00
Danil Chapovalov
9c58483b5a Rename EncodedImage property Timetamp to RtpTimestamp
To avoid name collision with Timestamp type,
To avoid confusion with capture time represented as Timestamp

Bug: webrtc:9378
Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40796}
2023-09-24 20:06:48 +00:00
Tony Herre
55b593fb6b Remove EncodedFrame::MissingFrame and start removing Decode() param
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.

Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
2023-08-30 10:38:35 +00:00
Harald Alvestrand
4d25a77fd3 Deprecate AsyncResolver config fields and remove internal usage.
Bug: webrtc:12598
Change-Id: Ic43cbcd13e4de44b02351c89da12844606368623
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317604
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40627}
2023-08-25 14:02:27 +00:00
Artem Titov
1997837d16 Add stream label to test video source for better debugablity and testability
Bug: b/294812400
Change-Id: I830515b797100ca2dc0e68dd3b79d5a1bb4068da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316221
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40581}
2023-08-21 08:25:38 +00:00
Mirko Bonadei
2d7ccb4149 Revert "FrameGeneratorCapturer: don't generate video before Start is called"
This reverts commit 00a8576a67c9e37de52a9d0c18042b4d4fd339a2.

Reason for revert: Speculative rollback (performance metrics change)

Original change's description:
> FrameGeneratorCapturer: don't generate video before Start is called
>
> Bug: b/272350185
> Change-Id: I3c264df49e952c8f852feb08607b8d4e320b15fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309860
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40336}

Bug: b/272350185, b/288515909
Change-Id: I66fc61d5d4d1c17f46f1f5b4fc6ff64a9b2012f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310681
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
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>
Cr-Commit-Position: refs/heads/main@{#40372}
2023-06-28 19:58:41 +00:00
Jeremy Leconte
9a3ab3dcca Add a method to log AnalyzingVideoSink metrics.
Change-Id: I19a954f4341c6581d89a8fecf8f2646bb3fe46f4
Bug: b/282154243
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#40353}
2023-06-27 08:39:53 +00:00
Jeremy Leconte
34589929fe Add audio energy metric.
More details on audio energy can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcaudiosourcestats-totalaudioenergy

Change-Id: Ie8b543c0c3d2136f453c6731945f93de4c38218c
Bug: b/272781101
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310121
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#40352}
2023-06-27 08:25:32 +00:00
Jeremy Leconte
93008bde6c DefaultVideoQualityAnalyzer::GetCpuUsagePercent is const.
Change-Id: I46216217ccfcd58775bb9e872ad5d8c7ebc80ead
Bug: b/272781101
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310000
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#40338}
2023-06-22 16:21:40 +00:00
Artem Titov
00a8576a67 FrameGeneratorCapturer: don't generate video before Start is called
Bug: b/272350185
Change-Id: I3c264df49e952c8f852feb08607b8d4e320b15fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309860
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40336}
2023-06-22 14:00:22 +00:00
Artem Titov
5246ae20a2 Fix TestVideoCapturer and subclasses to support pause/resume video
Bug: b/272350185
Change-Id: I8e2e1a833430f78627ec6301ea23f2f8337a01ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309622
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40329}
2023-06-21 17:46:33 +00:00
Henrik Boström
c929ab49b9 Reland "[Stats] Remove enum-like structs in favor of strings."
This is a reland of commit ccc87ea3c625e43ab138e00ba2ef1a2d99756199

Downstream project has been updated.

Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}

Bug: webrtc:15245
Change-Id: Ibc7aeb518ed0bd7f1d725f140132c99e5a89bcf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308880
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@{#40305}
2023-06-17 15:41:44 +00:00
Christoffer Jansson
45666d4b01 Revert "[Stats] Remove enum-like structs in favor of strings."
This reverts commit ccc87ea3c625e43ab138e00ba2ef1a2d99756199.

Reason for revert: Breaks downstream project

Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}

Bug: webrtc:15245
Change-Id: I05db80ba9f29460239de82cea9d95136e4c708e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308860
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40298}
2023-06-16 06:29:26 +00:00
Henrik Boström
ccc87ea3c6 [Stats] Remove enum-like structs in favor of strings.
Due to a limitation of RTCStatsMember<T> not supporting enums, as well
as the fact that in JavaScript enums are represented as basic strings,
the stats enums have always been represented by T=std::string.

Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
values are simply string-copied (example: [2]) it seems safe to assume
that "stats enums are just strings" is here to stay.

Therefore there is little value in having C++ structs that look like
enums so I'm deleting those in favor of std::string operator==()
comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
lines of code from our code base.

I mostly want to get rid of these because they were taking up about 20%
of the rtcstats_objects.h real estate...

[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755

Bug: webrtc:15245
Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40295}
2023-06-15 16:11:27 +00:00
Artem Titov
eba7cee1da Extract TestADM into a separate target
Bug: b/272350185, webrtc:15104
Change-Id: I091b81d81506e0caad665522e872c5cccf45d8d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301980
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39906}
2023-04-20 10:45:37 +00:00