Only user was the replay.cc tool, when dumping frames to a file. It is
changed to instead inject a special decoder.
Bug: None
Change-Id: I521fbba1a0ef440cff7d786f6f4c6397e33f764f
Reviewed-on: https://webrtc-review.googlesource.com/83121
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23675}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
This replaces webrtc::VideoSendStream::DegradationPreference with
webrtc::DegradationPreference, and adds "DISABLED".
It's still not wired up from RtpSenderInterface::SetParameters to the
underlying video engine; that would be the next step.
Bug: webrtc:8830
Change-Id: I582ffd04eaef33c73d9892e52e789804c933b864
Reviewed-on: https://webrtc-review.googlesource.com/77024
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23276}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
This CL adds an end to end test testing that jumps in receive time are
properly filtered when the receive time correction field trial is enabled.
Bug: webrtc:9054
Change-Id: I1d52594b6559e752c04c997ba56c6a3e20e629cd
Reviewed-on: https://webrtc-review.googlesource.com/64727
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22977}
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}
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}
This is a reland of dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8
Original change's description:
> Storing frame if encoder is paused.
>
> Adds a pending frame to VideoStreamEncoder that is used to store frames
> that are not sent because the encoder is paused. If the encoder is
> resumed within 200 ms, the pending frame will be encoded and sent. This
> ensures that resuming a stream instantly starts sending frames if it is
> possible.
>
> This also protects against a race between submitting the first frame
> and enabling the encoder that caused flakiness in end to end tests
> when using the task queue based congestion controller.
>
> Bug: webrtc:8415
> Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
> Reviewed-on: https://webrtc-review.googlesource.com/67141
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22781}
Bug: webrtc:8415
Change-Id: I0ea7d4d679e7845907cfbe9a120f128ff2180e4b
Reviewed-on: https://webrtc-review.googlesource.com/68580
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22810}
This reverts commit dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8.
Reason for revert: breaks downstream projects
Original change's description:
> Storing frame if encoder is paused.
>
> Adds a pending frame to VideoStreamEncoder that is used to store frames
> that are not sent because the encoder is paused. If the encoder is
> resumed within 200 ms, the pending frame will be encoded and sent. This
> ensures that resuming a stream instantly starts sending frames if it is
> possible.
>
> This also protects against a race between submitting the first frame
> and enabling the encoder that caused flakiness in end to end tests
> when using the task queue based congestion controller.
>
> Bug: webrtc:8415
> Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
> Reviewed-on: https://webrtc-review.googlesource.com/67141
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22781}
TBR=sprang@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8415
Change-Id: I4449eca65a64e2bc2fb25d866e5775e9a085cee9
Reviewed-on: https://webrtc-review.googlesource.com/68280
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22788}
Adds a pending frame to VideoStreamEncoder that is used to store frames
that are not sent because the encoder is paused. If the encoder is
resumed within 200 ms, the pending frame will be encoded and sent. This
ensures that resuming a stream instantly starts sending frames if it is
possible.
This also protects against a race between submitting the first frame
and enabling the encoder that caused flakiness in end to end tests
when using the task queue based congestion controller.
Bug: webrtc:8415
Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
Reviewed-on: https://webrtc-review.googlesource.com/67141
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22781}
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}
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}
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}
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}
Adding TaskQueueCongestionControl field trial to parametrized end to
end tests. This ensures that enabling the field trial will not break the
functionality tested in the tests.
Bug: webrtc:8415
Change-Id: Ieac75b840f18af2d9d5d35f976e119a8b3e7bfc0
Reviewed-on: https://webrtc-review.googlesource.com/61722
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22498}
This is a reland of d90a7e842437f5760a34bbfa283b3c4182963889
Original change's description:
> Add multiplex case to webrtc_perf_tests
>
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
>
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}
Bug: webrtc:7671
Change-Id: Ib6e37ce4bc0bae903dd72f49ffdc2ee583d75491
TBR: niklas.enbom@webrtc.org, phoglund@webrtc.org, sprang@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/61120
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22376}
This is a reland of d90a7e842437f5760a34bbfa283b3c4182963889
Original change's description:
> Add multiplex case to webrtc_perf_tests
>
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
>
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}
Bug: webrtc:7671
Change-Id: Iba2e89aee73a73a0372edea26933d6a7ea2e0ec9
TBR: niklas.enbom@webrtc.org, phoglund@webrtc.org, sprang@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/60600
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22336}
This CL adds two new tests to perf, covering I420 and I420A input to multiplex
codec. In order to have the correct input, it adds I420A case to
SquareGenerator and corresponding PSNR and SSIM calculations.
Bug: webrtc:7671
Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
Reviewed-on: https://webrtc-review.googlesource.com/52180
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22330}
Moving usage of bitrate configuration related interface from Call
interface to the corresponding methods in the RtpSendTransportController
interface.
SetBitrateConfig was replaced with SetSdpBitrateParameters
SetBitrateConfigMask was replaced with SetClientBitratePreferences
OnNetworkRouteChanged was replaced with OnNetworkRouteChanged
This makes it more clear that RtpSendTransportController owns bitrate
configuration and fits a longer term ambition to reduce the scope of
the Call class.
Bug: webrtc:8415
Change-Id: I6d04eaad22a54ecd5ed60096e01689b0c67e9c65
Reviewed-on: https://webrtc-review.googlesource.com/54365
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22131}
Modified the StopsAndResumesMediaWhenCongestionWindowFull test so it
detects when it's test condition is fulfilled rather than waiting for a
set amount of time. This makes it less sensitive to timing changes in
the underlying code and makes it return earlier if possible.
The number of padding packets to wait for were also reduced to save some
runtime.
Bug: webrtc:8415
Change-Id: Ib72524591e976f8bea7ef8b856e2c2c87b2e13ff
Reviewed-on: https://webrtc-review.googlesource.com/56041
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22129}
This is a reland of 5897fe27abcbe70f706cc23adc26147e0581f97e.
Adding back CallConfig::kDefaultStartBitrateBps as deprecated.
Also making BitrateContraints::kDefaultStartBitrateBps private to stop
it from being used in other places.
Original change's description:
> Moved BitrateConfig out of Call::Config.
>
> This prepares for a CL extracting the bitrate configuration logic from
> the Call class.
>
> Also renaming BitrateConfig to BitrateConstraints.
>
> Bug: webrtc:8415
> Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
> Reviewed-on: https://webrtc-review.googlesource.com/54400
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22104}
Bug: webrtc:8415
Change-Id: Iacfe2d6daedff710832ab89210c7c66d4403c93b
Reviewed-on: https://webrtc-review.googlesource.com/55980
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22123}
This reverts commit 5897fe27abcbe70f706cc23adc26147e0581f97e.
Reason for revert: Breaking internal builds
Original change's description:
> Moved BitrateConfig out of Call::Config.
>
> This prepares for a CL extracting the bitrate configuration logic from
> the Call class.
>
> Also renaming BitrateConfig to BitrateConstraints.
>
> Bug: webrtc:8415
> Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
> Reviewed-on: https://webrtc-review.googlesource.com/54400
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22104}
TBR=nisse@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I598040edba7f1ff8b39d2d9c3c3ceca5627aaa0c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/55740
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22106}
This prepares for a CL extracting the bitrate configuration logic from
the Call class.
Also renaming BitrateConfig to BitrateConstraints.
Bug: webrtc:8415
Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
Reviewed-on: https://webrtc-review.googlesource.com/54400
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22104}
Expanded congestion window test to test that media is resumed after
being stopped due to congestion window getting filled. Previously only
the behavior that padding packets was sent in congested state was tested,
but not that media actually was resumed when feedback from the padding
packets was received.
Bug: None
Change-Id: Icd494c2e5386926f92c4d5dd0e8bc80c81608325
Reviewed-on: https://webrtc-review.googlesource.com/46262
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21941}
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}
