Commit graph

406 commits

Author SHA1 Message Date
Niels Möller
9def99487e Delete BasicPacketSocketFactory constructor with thread argument
In callers where it's non-trivial to explicitly pass the right
SocketFactory, pull the call to rtc::Thread::socketserver() into the
caller, with a TODO comment.

Bug: webrtc:13145
Change-Id: I029d3adca385d822180e089f016c3778e0d4fd0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231227
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35063}
2021-09-22 12:15:06 +00:00
Mirko Bonadei
824eebab8b Improve points visualization in metrics_plotter.
Bug: None
Change-Id: Id86c67ae9e1690817b98d8f62db0c9d05911a58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231680
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34963}
2021-09-10 10:59:58 +00:00
Danil Chapovalov
5ce7d14f81 Delete legacy rtp header parser as no longer used
Bug: None
Change-Id: I3c532eee7f2d9e5295874dd538730625c8d423ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227086
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34676}
2021-08-09 12:14:52 +00:00
Jonathan Lennox
03df29c100 Add -render_width and -render_height arguments to video_replay.
Bug: webrtc:12975
Change-Id: Ice8d704980a374378a1e20f526d5d8fb60e4db3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225960
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34605}
2021-07-30 16:56:12 +00:00
Artem Titov
54500adead Use backticks not vertical bars to denote variables in comments for /rtc_tools
Bug: webrtc:12338
Change-Id: Id47ef14982a6f31df6fc2e6d317e14f6e269e706
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226954
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34571}
2021-07-27 18:56:12 +00:00
Mirko Bonadei
25ab3228f3 Replace assert() with RTC_DCHECK().
CL partially auto-generated with:

git grep -l "\bassert(" | grep "\.[c|h]" | \
  xargs sed -i 's/\bassert(/RTC_DCHECK(/g'

And with:

git grep -l "RTC_DCHECK(false)" |  \
  xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'

With some manual changes to include "rtc_base/checks.h" where
needed.

A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.

The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.

This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).

Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
2021-07-09 07:49:43 +00:00
Jakob Ivarsson
c0a4a09fae Use default NetEq config for simulation in event log visualizer.
This disables fast accelerate mode but max buffer size is the same.

Bug: None
Change-Id: Iba883051c42b28ab094075948a43ec288b77ad5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224545
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34403}
2021-07-01 14:06:20 +00:00
Philipp Hancke
c060ce40bf video_replay: use abseil uint32_t for ssrc and ssrc_rtx
simplifying the validation

BUG=webrtc:12575

Change-Id: I3f43347aec653ac0297523cda88f9101c57fe1c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211867
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34392}
2021-06-30 10:28:46 +00:00
Mirko Bonadei
e99f6879f6 Move WebRTC to non deprecated jsoncpp APIs.
This will allow the removal of -Wno-deprecated-declarations from
WebRTC's BUILD.gn files even if [1] will still propagate in the
build graph, causing some ABSL_DEPRECATED to be ignored.

[1] - https://source.chromium.org/chromium/chromium/src/+/main:third_party/jsoncpp/BUILD.gn;l=15;drc=592d07510836410a1ec4833de342544d1b39ef08

Bug: webrtc:10770
Change-Id: I90193ac5cc3e41f00f1b5dd5dac3c462e4b5f9ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223666
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34375}
2021-06-28 12:09:14 +00:00
Danil Chapovalov
76a35d9ce2 Delete legacy RtpHeaderParser wrapper
Bug: None
Change-Id: I4deec4fab631488ef2d0706848cbbe4e085825bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221617
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34341}
2021-06-21 09:17:52 +00:00
Erik Språng
6a0a55907b Reland "Correctly handle retransmissions/padding in early loss detection."
This is a reland of e9ae4729e0

TBR=philipel@webrtc.org,terelius@webrtc.org

Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34293}

Bug: webrtc:12713
Change-Id: Iec123d71edafea98fe289acde007b57e212681f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222640
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34297}
2021-06-16 08:14:27 +00:00
Erik Språng
d6957c2eed Revert "Correctly handle retransmissions/padding in early loss detection."
This reverts commit e9ae4729e0.

Reason for revert: Internal test failure

Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34293}

TBR=danilchap@webrtc.org,terelius@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Iaca6dc7739d953e97add5f5d516139b4819e43ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34294}
2021-06-15 15:59:10 +00:00
Erik Språng
e9ae4729e0 Correctly handle retransmissions/padding in early loss detection.
This CL makes sure we don't cull packets from the history based on
incorrect ack mapping, just like it's predecessor:
https://webrtc-review.googlesource.com/c/src/+/218000

It also changes the logic to make sure retransmits counts towards
history pruning - and properly ignores padding/fec.

Bug: webrtc:12713
Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34293}
2021-06-15 15:39:19 +00:00
Henrik Boström
58126f92bf Update the only 3 remaining kFilterBilinear to kFilterBox.
Bilinear is faster but lesser quality, box is best quality. Our code
base has disagreed about which filter to use for quite some time,
causing aliasing bug reports. In an effort to avoid aliasing artifacts
and make our scaling filters more predictable, we're updating all uses
to kFilterBox.

WebRTC already uses kFilterBox everywhere except for these three
places. The main discrepency was between Chromium and WebRTC but that
has already been fixed. This CL fixes the last remaining bilinears.

This brings the WebRTC kFilterBox use count up from 11 to 14 and the
kFilterBilinear use count down from 3 to 0.

Bug: chromium:1212630
Change-Id: I5fe4aa92b9275d65b91ea97925533055d190d317
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221372
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34248}
2021-06-08 13:19:23 +00:00
Paul Hallak
46fbefa302 Convert to NTP time using the real clock.
Bug: webrtc:11327
Change-Id: I523b111c72569580b8b27d47ad648e7887bea872
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219793
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Paul Hallak <phallak@google.com>
Cr-Commit-Position: refs/heads/master@{#34082}
2021-05-21 19:29:32 +00:00
Per Kjellander
fe2063ebc7 Remove REMB throttling funcionality from PacketRouter
This removes PacketRouter inheritance from  RemoteBitrateObserver and TransportFeedbackSenderInterface.
Call binds methods for sending REMB and transport feedback messages from RemoteCongestionController to PacketRouter.
This is needed until the RTCPTranseiver is used instead of the RTP modules.

Bug: webrtc:12693
Change-Id: I7088de497cd6d1e15c98788ff3e6b0a2c8897ea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215965
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33993}
2021-05-12 11:24:58 +00:00
Tommi
87f7090fd9 Replace more instances of rtc::RefCountedObject with make_ref_counted.
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.

Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
2021-04-27 17:01:59 +00:00
Philipp Hancke
f3a687a175 video_replay: add --start-timestamp and --stop-timestamp
to allow filtering of the frames that are being decoded.
Timestamp wraparound is not supported.

BUG=webrtc:12575

Change-Id: If08c46a377336e559475caefa934f6b82c46decc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211864
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33812}
2021-04-23 08:36:15 +00:00
Johannes Kron
bb52bdf095 Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium"
This reverts commit cd5127b11e.

Reason for revert: Fuzzer build problems fixed.

Original change's description:
> Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium"
>
> This reverts commit dfe19719e5.
>
> Reason for revert: Breaks fuzzers in Chromium builds. See https://ci.chromium.org/ui/p/chromium/builders/try/linux-libfuzzer-asan-rel/685438/overview. I am reverting since this blocks the roll but I will be in touch for a fix.
>
> Original change's description:
> > Enable use of rtc::SystemTimeNanos() provided by Chromium
> >
> > This is the third CL out of three to enable overriding
> > of the function SystemTimeNanos() in rtc_base/system_time.cc
> >
> > When WebRTC is built as part of Chromium the rtc::SystemTimeNanos()
> > function provided by Chromium will be used. This is controlled
> > by the build argument rtc_exclude_system_time which directly
> > maps to the macro WEBRTC_EXCLUDE_SYSTEM_TIME.
> >
> > By doing this we are making sure that the WebRTC and Chromium
> > clocks are the same.
> >
> > Bug: chromium:516700
> > Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33337}
>
> TBR=kron@webrtc.org
>
> Bug: chromium:516700
> Change-Id: I9ecd1784a6c1cdac8bae07d34f7df20c62a21a95
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208740
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33340}

Bug: chromium:516700
Change-Id: I4cd68bac1cc4befdb46351f5d6fb2cf1ef5c3062
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208742
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33341}
2021-02-25 10:48:55 +00:00
Mirko Bonadei
cd5127b11e Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium"
This reverts commit dfe19719e5.

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

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

TBR=kron@webrtc.org

Bug: chromium:516700
Change-Id: I9ecd1784a6c1cdac8bae07d34f7df20c62a21a95
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208740
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33340}
2021-02-25 08:47:19 +00:00
Johannes Kron
dfe19719e5 Enable use of rtc::SystemTimeNanos() provided by Chromium
This is the third CL out of three to enable overriding
of the function SystemTimeNanos() in rtc_base/system_time.cc

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

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

Bug: chromium:516700
Change-Id: If7f749c4aadefb1cfc07ba4c7e3f45dc6c31118b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208223
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33337}
2021-02-24 22:25:33 +00:00
Danil Chapovalov
e904161cec Replace RTC_DEPRECATED with ABSL_DEPRECATED
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.

Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
2021-02-22 12:53:23 +00:00
Artem Titov
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
Per Åhgren
879d33b9f8 Add more refined control over dumping of data and the aecdump content
This CL adds the ability in audioproc_f and unpack_aecdump to:
-Clearly identify the Init events and when those occur.
-Optionally only process a specific Init section of an aecdump.
-Optionally selectively turn on dumping of internal data for a
 specific init section, and a specific time interval.
-Optionally let unpack_aecdump produce file names based on inits.

Bug: webrtc:5298
Change-Id: Id654b7175407a23ef634fca832994d87d1073239
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196160
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33181}
2021-02-06 00:36:10 +00:00
Björn Terelius
8db9534909 Support event log visualization in python3
Bug: webrtc:12431
Change-Id: I54910e862ab8de013879af632efc2f3834d80552
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205526
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33170}
2021-02-04 19:01:58 +00:00
Björn Terelius
4ef5638871 Parse and plot RTCP BYE in RTC event log.
Bug: webrtc:12432
Change-Id: I9a98876044e0e75ee4f3ef975ae75237606d108d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204380
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33161}
2021-02-04 11:28:46 +00:00
Artem Titov
c8421c4c3e Replace rtc::ThreadChecker with webrtc::SequenceChecker
Bug: webrtc:12419
Change-Id: I825c014cc1c4b1dcba5ef300409d44859e971144
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33136}
2021-02-02 14:56:27 +00:00
Andrey Logvin
e7c79fd3d6 Remove from chromium build targets that are not compatible with it.
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.

`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.

Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 13:46:19 +00:00
Andrey Logvin
7864600a6e Add absl_deps field for rtc_test and rtc_executable
To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl".

Bug: webrtc:12404
Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33108}
2021-01-29 16:40:49 +00:00
Philipp Hancke
fae4fb1345 video_replay: add support for IVF file output
adding a -decoder_ivf_filename command line flag.

BUG=none

Change-Id: I895a6bf8093d5f36c17462d97240b17ada4dc9f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33077}
2021-01-26 19:51:08 +00:00
Mirko Bonadei
e5f4c6b8d2 Reland "Refactor rtc_base build targets."
This is a reland of 69241a93fb

Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).

Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
>   break a circular dependency (is has been extracted from
>   //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
>   break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}

Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 17:00:05 +00:00
Mirko Bonadei
7acc2d9fe3 Revert "Refactor rtc_base build targets."
This reverts commit 69241a93fb.

Reason for revert: Breaks WebRTC roll into Chromium.

Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
>   break a circular dependency (is has been extracted from
>   //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
>   break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}

TBR=mbonadei@webrtc.org,hta@webrtc.org

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

No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
2021-01-14 21:27:38 +00:00
Mirko Bonadei
69241a93fb Refactor rtc_base build targets.
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.

This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).

The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
  break a circular dependency (is has been extracted from
  //rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
  break another circular dependency.

Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
2021-01-11 18:32:30 +00:00
Ilya Nikolaevskiy
6a646905b9 Use a task queue for call interaction in video_replay tool
After some recent change current thread while creating the receive stream is
used as a task queue for stats calculation.

Currently, video_replay tool doesn't create streams inside a task queue, so
it ends up posting tasks to a "dead" task queue, which doesn't run message
processing loop at all.

Bug: webrtc:12204
Change-Id: Ieb97a10f44a11e92e2ac08df5b39b7cd695c852e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196860
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32824}
2020-12-14 15:25:43 +00:00
Niels Möller
0d863f72a8 Cleanup of bwe_defines.h
Delete unused macros BWE_MIN and BWE_MAX.

Move enum RateControlState: Make it a private enum class in
AimdRateControl, the only user.

Change users of the header file that only need BandwidthUsage, to
instead include api/network_state_predictor.h, the file defining this
class. As a result, fewer dependencies on
modules/remote_bitrate_estimator.

Bug: None
Change-Id: I4450c79dd58e1875d64dd74d1ae2cb7d911a14b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195222
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32704}
2020-11-26 12:26:02 +00:00
Mirko Bonadei
20e4c80fbe Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
This is a reland of f5e261aaf6

This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:29:36 +00:00
Karl Wiberg
c95b939667 Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.

Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
2020-11-09 10:47:55 +00:00
Jakob Ivarsson
ddd41919c0 Remove simulated neteq packet loss plot.
This was missed in https://webrtc-review.googlesource.com/c/src/+/183620

Bug: webrtc:11622
Change-Id: I2815aa972d1051da70494d08d3db8eb0080b70bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191442
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32541}
2020-11-03 17:25:50 +00:00
Mirko Bonadei
8cc6695652 Reformat python files checked by pylint (part 1/2).
After recently changing .pylintrc (see [1]) we discovered that
the presubmit check always checks all the python files when just
one python file gets updated.

This CL moves all these files one step closer to what the linter
wants.

Autogenerated with:

# Added all the files under pylint control to ~/Desktop/to-reformat
cat ~/Desktop/to-reformat | xargs sed -i '1i\\'
git cl format --python --full

This is part 1 out of 2. The second part will fix function names and
will not be automated.

[1] - https://webrtc-review.googlesource.com/c/src/+/186664

No-Presubmit: True
Bug: webrtc:12114
Change-Id: Idfec4d759f209a2090440d0af2413a1ddc01b841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32530}
2020-10-30 10:13:11 +00:00
Ilya Nikolaevskiy
38e9b06151 Reland "Add scaling interface to VideoFrameBuffer"
(Reland with no changes after the fix to the downstream project)

This can be overriden for kNative frame types to perform scaling efficiently.

Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.

Originally Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303

(Landing with TBR as it's unchaged reland of already approved CL)
TBR=nisse@webrtc.org,sakal@webrtc.org

Bug: webrtc:11976, chromium:1132299
Change-Id: Ia23f7d3e474bd9cdc177104cc5c6d772f04b210f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187345
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32362}
2020-10-09 08:30:50 +00:00
Ilya Nikolaevskiy
441dbf9a56 Revert "Add scaling interface to VideoFrameBuffer"
This reverts commit c79f1d8cfb.

Reason for revert: Breaks downstream project.

Original change's description:
> Add scaling interface to VideoFrameBuffer
>
> This can be overriden for kNative frame types to perform scaling efficiently.
>
> Default implementations for existing buffer types require actual
> buffer implementation, thus this CL also merges "video_frame"
> with "video_frame_I420" build targets.
>
> Bug: webrtc:11976, chromium:1132299
> Change-Id: I3bf5f6bf179db5e7ab165b1c2301980043a08765
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#32352}

TBR=mbonadei@webrtc.org,sakal@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,eshr@google.com

Change-Id: I86ac697bf963ef7e2c4f2ed34c3a7bf04f4f1ce1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11976
Bug: chromium:1132299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187344
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32354}
2020-10-08 14:16:23 +00:00
Ilya Nikolaevskiy
c79f1d8cfb Add scaling interface to VideoFrameBuffer
This can be overriden for kNative frame types to perform scaling efficiently.

Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.

Bug: webrtc:11976, chromium:1132299
Change-Id: I3bf5f6bf179db5e7ab165b1c2301980043a08765
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32352}
2020-10-08 13:33:00 +00:00
Mirko Bonadei
0abd518abd Revert "Introduce RTC_NO_UNIQUE_ADDRESS."
This reverts commit f5e261aaf6.

Reason for revert: Breaks downstream projects.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

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

Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
2020-10-07 07:37:01 +00:00
Bjorn Terelius
7634ea7240 Add method to extract triage alerts from RTC event log analyzer.
Bug: webrtc:11566
Change-Id: I8315895be4fe93513247c49452c50ec23e9d1e11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186560
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32292}
2020-10-02 13:41:30 +00:00
Mirko Bonadei
f5e261aaf6 Introduce RTC_NO_UNIQUE_ADDRESS.
This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.

The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.

Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
2020-09-30 09:52:49 +00:00
Artem Titov
0b9e354d61 Improve perf metrics plotter
Add ability to specify which metrics to plot on the plotter level and
add sorting of plottable data because there is no guarantee on the perf
writer side that output is sorted by time.

Bug: webrtc:11959
Change-Id: I87e6f5720fff2b259f58e3fc5f7ed2462568e0d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32233}
2020-09-29 13:51:59 +00:00
Niels Möller
de95329daa Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.

Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
2020-09-29 10:19:20 +00:00
Mohamed Heikal
3a5612a8de Clean up references to deprecated create_srcjar in android_resources
`create_srcjar = false` was needed during the transition to moving
R.java generation to android_library targets. Now this variable is
unused (the variable is asserted to be false), clean up all references.

Bug: chromium:1073476
Change-Id: I4c09ea05ded27ea2360392aacbce036bc1a2f928
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mohamed Heikal <mheikal@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32178}
2020-09-23 14:11:50 +00:00
Niels Möller
4461f059d1 Delete unused NetEq stats currentPacketLossRate, currentDiscardRate and addedSamples
Bug: webrtc:11622
Change-Id: I097bb7284d952ada41f4f38dd7adf3536bd040ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183620
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32148}
2020-09-21 12:19:16 +00:00
Mohamed Heikal
3f94fc53d4 Migrate android_resources targets to not create R.java files
R.java file creation responsibilities will be moved to android_library
and android_apk targets and creating R.java files in the
android_resources targets is now deprecated. This cl migrates webrtc
targets to the new way.

Bug: chromium:1073476
Change-Id: I0a2fa759d3ff1d8e201e5719c9238701a58171e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183060
Commit-Queue: Mohamed Heikal <mheikal@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32026}
2020-09-01 20:29:36 +00:00