12 Commits

Author SHA1 Message Date
Sebastian Jansson
bed801e005 Removes unnecessary destructor checks in tests.
Removes checks that are not relevant to the particular tests. The checks
create dependencies on the CallTest base class. This prepares for
further refactoring in CallTest.

Bug: webrtc:9510
Change-Id: Icd2bb4fe168aabd377d349b1de3de833fb7ae075
Reviewed-on: https://webrtc-review.googlesource.com/87823
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23902}
2018-07-10 11:29:45 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Niels Möller
4db138e889 Reland "Move creating encoder to VideoStreamEncoder."
This is a reland of fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80

Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}

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

Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
2018-04-19 08:48:58 +00:00
Niels Moller
0d650b44ef Revert "Move creating encoder to VideoStreamEncoder."
This reverts commit fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80.

Reason for revert: Appears to break Chromium, see https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Linux%20Tester/43756, where remoting_unittests failed.

Original change's description:
> Move creating encoder to VideoStreamEncoder.
> 
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
> 
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}

TBR=magjed@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

Change-Id: I47ee3ac42e62472d825a08c98e28f9ae53ec9fff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/70600
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22914}
2018-04-18 07:17:16 +00:00
Niels Möller
fb82fcc7f9 Move creating encoder to VideoStreamEncoder.
This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
One implication is that encoder is not created until the first
frame arrives, and some of the tests needed updates to emit a
frame or two.

Bug: webrtc:8830
Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
Reviewed-on: https://webrtc-review.googlesource.com/64885
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22905}
2018-04-17 15:04:33 +00:00
Niels Möller
259a497632 Reland "Reland "Move rtp-specific config out of EncoderSettings.""
This reverts commit 6c2c13af06b32778b86950681758a7970d1c5d9e.

Reason for revert: Intend to investigate and fix perf problems.

Original change's description:
> Revert "Reland "Move rtp-specific config out of EncoderSettings.""
> 
> This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.
> 
> Reason for revert: Regression in ramp up perf tests.
> 
> Original change's description:
> > Reland "Move rtp-specific config out of EncoderSettings."
> >
> > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
> >
> > Original change's description:
> > > Move rtp-specific config out of EncoderSettings.
> > >
> > > In VideoSendStream::Config, move payload_name and payload_type from
> > > EncoderSettings to Rtp.
> > >
> > > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > > and should perhaps be renamed in a follow up cl. It's no longer
> > > passed as an argument to VideoCodecInitializer::SetupCodec.
> > >
> > > The latter then needs a different way to know the codec type,
> > > which is provided by a new codec_type member in VideoEncoderConfig.
> > >
> > > Bug: webrtc:8830
> > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22532}
> >
> > Bug: webrtc:8830
> > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> > Reviewed-on: https://webrtc-review.googlesource.com/63721
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22595}
> 
> TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
> 
> Bug: webrtc:8830,chromium:827080
> Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
> Reviewed-on: https://webrtc-review.googlesource.com/65520
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22677}

TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org

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

Bug: webrtc:8830, chromium:827080
Change-Id: I9b62987bf5daced90dfeb3ebb6739c80117c487f
Reviewed-on: https://webrtc-review.googlesource.com/66862
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22751}
2018-04-05 14:30:09 +00:00
Niels Möller
6c2c13af06 Revert "Reland "Move rtp-specific config out of EncoderSettings.""
This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.

Reason for revert: Regression in ramp up perf tests.

Original change's description:
> Reland "Move rtp-specific config out of EncoderSettings."
>
> This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
>
> Original change's description:
> > Move rtp-specific config out of EncoderSettings.
> >
> > In VideoSendStream::Config, move payload_name and payload_type from
> > EncoderSettings to Rtp.
> >
> > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > and should perhaps be renamed in a follow up cl. It's no longer
> > passed as an argument to VideoCodecInitializer::SetupCodec.
> >
> > The latter then needs a different way to know the codec type,
> > which is provided by a new codec_type member in VideoEncoderConfig.
> >
> > Bug: webrtc:8830
> > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22532}
>
> Bug: webrtc:8830
> Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> Reviewed-on: https://webrtc-review.googlesource.com/63721
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22595}

TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org

Bug: webrtc:8830,chromium:827080
Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
Reviewed-on: https://webrtc-review.googlesource.com/65520
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22677}
2018-03-29 11:45:18 +00:00
Niels Möller
04dd176862 Reland "Move rtp-specific config out of EncoderSettings."
This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c

Original change's description:
> Move rtp-specific config out of EncoderSettings.
> 
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
> 
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
> 
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
> 
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}

Bug: webrtc:8830
Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
Reviewed-on: https://webrtc-review.googlesource.com/63721
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22595}
2018-03-26 08:39:39 +00:00
Niels Moller
92be1caf4f Revert "Move rtp-specific config out of EncoderSettings."
This reverts commit bc900cb1d1810fcf678fe41cf1e3966daa39c88c.

Reason for revert: Broke downstream projects.

Original change's description:
> Move rtp-specific config out of EncoderSettings.
> 
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
> 
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
> 
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
> 
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}

TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org

Change-Id: I01f06c1fcf21eb2cd40dca7d4f268614200ee490
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/63720
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22537}
2018-03-21 13:53:49 +00:00
Niels Möller
bc900cb1d1 Move rtp-specific config out of EncoderSettings.
In VideoSendStream::Config, move payload_name and payload_type from
EncoderSettings to Rtp.

EncoderSettings now contains configuration for VideoStreamEncoder only,
and should perhaps be renamed in a follow up cl. It's no longer
passed as an argument to VideoCodecInitializer::SetupCodec.

The latter then needs a different way to know the codec type,
which is provided by a new codec_type member in VideoEncoderConfig.

Bug: webrtc:8830
Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
Reviewed-on: https://webrtc-review.googlesource.com/62062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22532}
2018-03-21 12:55:08 +00:00
Emircan Uysaler
ef76e5a062 Add multiplex case to codec_tests
Bug: webrtc:7671
Change-Id: I38de24ea300d69757f5bd07111abc83b1f042751
Reviewed-on: https://webrtc-review.googlesource.com/50700
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21991}
2018-02-12 22:28:04 +00:00
Sebastian Jansson
c5017136c7 Split end to end tests into more cohesive test sets.
end_to_end_tests.cc was over 5000 lines and covered many different
areas in it's testing. In this change it is separated into multiple
smaller test sets separated by the functionality they are testing. The
reasoning behind this is that the fact that a test is working end to end
should be secondary to what functionality the test is actually testing.

A slight functional change is that for some of the tests the
parametrization over round robin pacing being controlled with a field
trial is removed since they are simple enough that they should not be
affected by the pacing method.

Bug: None
Change-Id: I4b7eba80fc142ecfc8fa642dab9b6f587d914048
Reviewed-on: https://webrtc-review.googlesource.com/46143
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21876}
2018-02-02 16:09:16 +00:00