Create a server using:
./data_channel_benchmark --server --port 12345
Start the flow of data from the server to a client using:
./data_channel_benchmark --port 12345 --transfer_size 100
The throughput is reported on the server console.
The negotiation does not require a 3rd party server and is done over a
gRPC transport. No TURN server is configured, so both peers need to be
reachable using STUN only.
Bug: webrtc:13288
Change-Id: Iac9a96cf390ab465ea45a46bf0b40950c56dfceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36206}
This CL also removes dependency on the legacy field trial methods.
Bug: webrtc:11926
Change-Id: I53feeee86b92878cf0f2b8ebdce3d101f9e04014
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255381
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36205}
This is a reland of 43fb16921b29ecd3a2d87876dda75c575e05f66a
Original change's description:
> Refactor AnalyzerConfig to use Timestamps instead of microseconds.
>
> Add optional offset-to-UTC parameter to output. This allows aligning
> the x-axis in the generated charts to other UTC-based logs.
>
> Bug: b/215140373
> Change-Id: I65bcd295718acbb8c94e363907c1abc458067bfd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250203
> Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35992}
Bug: b/215140373
Change-Id: Id2b88cc4b8078a97275d49a617581cbbd02d2c6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252380
Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36066}
Add optional offset-to-UTC parameter to output. This allows aligning
the x-axis in the generated charts to other UTC-based logs.
Bug: b/215140373
Change-Id: I65bcd295718acbb8c94e363907c1abc458067bfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250203
Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35992}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This is part of a set of CLs to fix the Chromium roll.
Bug: None
Change-Id: I7cc1e5b84a2443f311fb4e047954d7d4d9c41189
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236761
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35284}
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}
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}
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}
This is a reland of e9ae4729e03f60dbe3b1828dd9009b401097cd3f
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}
This reverts commit e9ae4729e03f60dbe3b1828dd9009b401097cd3f.
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}
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}
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}
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}
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}
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}
This reverts commit cd5127b11e04b7f49031b2412625c36e4a86c3da.
Reason for revert: Fuzzer build problems fixed.
Original change's description:
> Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium"
>
> This reverts commit dfe19719e53abfd4d73722942445c5e1046b671b.
>
> 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}
This reverts commit dfe19719e53abfd4d73722942445c5e1046b671b.
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}
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}
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}
