609 Commits

Author SHA1 Message Date
Mirko Bonadei
b1b6129944 Revert "Each spatial layer can only have 1 QP value."
This reverts commit 962b3935e44053641764beb0bd095540fe8cbd64.

Reason for revert: Breaks downstream tests.

Original change's description:
> Each spatial layer can only have 1 QP value.
>
> As the code is now, it looks like it accepts that a spatial layer can have more than 1 QP value. These QP values according to the code are summed. However, to my best knowledge this cannot be the case and makes the code hard to read. Therefore, updating it with a check such that it checks that each spatial layer only have 1 QP value and would be easier for a future code reader.
>
> Bug: webrtc:357636606
> Change-Id: I650cac724811a1ddc7ab8933c1e1ac5fe844b61c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358101
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
> Cr-Commit-Position: refs/heads/main@{#42736}

Bug: webrtc:357636606
Change-Id: I60a2d4e1285f961f2ed2ea4c1d2e5942ea68b365
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358721
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@{#42737}
2024-08-07 08:42:20 +00:00
Emil Vardar
962b3935e4 Each spatial layer can only have 1 QP value.
As the code is now, it looks like it accepts that a spatial layer can have more than 1 QP value. These QP values according to the code are summed. However, to my best knowledge this cannot be the case and makes the code hard to read. Therefore, updating it with a check such that it checks that each spatial layer only have 1 QP value and would be easier for a future code reader.

Bug: webrtc:357636606
Change-Id: I650cac724811a1ddc7ab8933c1e1ac5fe844b61c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358101
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#42736}
2024-08-07 07:59:22 +00:00
Artem Titov
eb3da2b1ec Extract video writing into separate target
Bug: None
Change-Id: I3af192606eb623e21a4d648fb69bb62c14ab8b0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42541}
2024-06-26 12:47:15 +00:00
Dor Hen
aefed55c25 [iwyu][1\n] Applying to api/[a-s]*
First batch of applying iwyu to the repo.
Done with:
> ./tools_webrtc/iwyu/apply-iwyu api
> git add api/[a-s]*
> python3 gn_autodeps.py ~/local/webrtc/src out/Default

Last step is a custom script I wrote to automatically apply new required
dependencies for target in gn, which saved tons of time manually going
over the files and fixing.
If this is something that interest others, I can submit it as well.

Bug: webrtc:42226242
Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42512}
2024-06-19 06:19:20 +00:00
Florent Castelli
99c519b3fd Mass removal of absl_deps in all BUILD.gn files
Bug: webrtc:341803749
Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 15:09:46 +00:00
Per K
5566b91356 Reland "Replace usage of link_capacity_kbps with DataRate link_capacity"
This reverts commit ff2dd50fd88e07affc4b070ce535935409f6673a.

Reason for revert: Temporary fix for downstream breakage in patch 2

Original change's description:
> Revert "Replace usage of link_capacity_kbps with DataRate link_capacity"
>
> This reverts commit 6186c0226e41dabbfc5cc8527e6b350b62f39f02.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > Replace usage of link_capacity_kbps with DataRate link_capacity
> >
> > Replace usage of BuiltInNetworkBehaviorConfig.link_capacity_kbps in tests with  DataRate link_capacity.
> >
> > Bug: webrtc:14525
> > Change-Id: Id1c1b8d20eb2be5e9d1461704bb7c37c61c491f0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350300
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42306}
>
> Bug: webrtc:14525
> Change-Id: I09ede3e89d065061cb4133bff96dbf242ff70832
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350621
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42309}

Bug: webrtc:14525
Change-Id: Ie35cd97a158d008a80ed007b27d2c6b1a9affff9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350541
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42320}
2024-05-16 10:39:10 +00:00
Jeremy Leconte
f9da667dca [DVQA] Sum the encoded_image_size for simulcast scenario.
This aligns actual_encode_bitrate with target_encode_bitrate

Change-Id: I8c460a5883e5eacbee8e67b5b6625f65792efb77
Bug: b/340790240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42310}
2024-05-15 11:23:45 +00:00
Mirko Bonadei
ff2dd50fd8 Revert "Replace usage of link_capacity_kbps with DataRate link_capacity"
This reverts commit 6186c0226e41dabbfc5cc8527e6b350b62f39f02.

Reason for revert: Breaks downstream project.

Original change's description:
> Replace usage of link_capacity_kbps with DataRate link_capacity
>
> Replace usage of BuiltInNetworkBehaviorConfig.link_capacity_kbps in tests with  DataRate link_capacity.
>
> Bug: webrtc:14525
> Change-Id: Id1c1b8d20eb2be5e9d1461704bb7c37c61c491f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350300
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42306}

Bug: webrtc:14525
Change-Id: I09ede3e89d065061cb4133bff96dbf242ff70832
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350621
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42309}
2024-05-15 11:09:33 +00:00
Per K
6186c0226e Replace usage of link_capacity_kbps with DataRate link_capacity
Replace usage of BuiltInNetworkBehaviorConfig.link_capacity_kbps in tests with  DataRate link_capacity.

Bug: webrtc:14525
Change-Id: Id1c1b8d20eb2be5e9d1461704bb7c37c61c491f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350300
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42306}
2024-05-15 08:44:20 +00:00
Sergey Sukhanov
26a082ce36 Introduce a mode that lets NetworkEmulationManager ignore DTLS handshake sizes.
Bug: b/169531206
Change-Id: I02c19385ff7078944f7509ecc07358b4315f7b08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350181
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42261}
2024-05-08 13:20:20 +00:00
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
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
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
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