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38 commits

Author SHA1 Message Date
Sebastian Jansson
7c1ac76f52 Adds binary proto ANA support in scenario tests. 
This makes it easier to reuse existing audio network adaptation
configurations in the scenario framework.

Bug: webrtc:9510
Change-Id: I06ab08684d449fef7fffe265d1078738d526a43d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169363
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30633}
 2020-02-27 14:53:59 +00:00
Danil Chapovalov
cad3e0e2fa Replace DataSize and DataRate factories with newer versions 
This is search and replace change:
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g"
git cl format

Bug: webrtc:9709
Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30545}
 2020-02-18 16:09:50 +00:00
Danil Chapovalov
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc 
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
 2020-02-10 12:21:17 +00:00
Sebastian Jansson
c9f42ad909 Simplifies transport overhead mechanism in Scenario test framework. 
This changes the behavior for adding virtual transport overhead so it
doesn't change the size of the actual payload buffer, only the
calculated packet size.

Bug: webrtc:9883
Change-Id: I6e24598378c4dd6a591d36ca3b162e933ff4ef7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164523
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30298}
 2020-01-17 11:30:02 +00:00
Artem Titov
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class 
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
 2019-12-07 00:54:26 +00:00
Erik Språng
014dd3c9f7 Trials should always be populated in call config. 
The trials are always set when a Call instead is created by a
CallFactory, but a lot of unit tests creates instances directly.
This CL updates those call site. There is still a fallback in place
in RtpTransportControllerSend, since there are down-stream usages that
need to be clean up. After that, we'll remove the fallback.

Bug: webrtc:10809
Change-Id: I0aacf0473317bcd64252dd43d93c42de730e2ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160408
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29978}
 2019-12-03 10:34:55 +00:00
Sebastian Jansson
24c678fd41 Adds test for loss based controller under cross traffic induced loss. 
Bug: webrtc:9883
Change-Id: I85a83dd15afe523e0ba5b3a723979317f0b98ab7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156501
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29465}
 2019-10-14 13:59:11 +00:00
Sebastian Jansson
ed0febf573 Add k prefix to FrameGenerator::OutputType enum values 
This prepares for using VideoFrameBuffer::Type as
FrameGenerator::OutputType, which will reduce the
number of redundant enums in the code.

Bug: webrtc:9883
Change-Id: I253f5f1ea7181e02a5cf1a92925f51da8ada6aa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146982
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28696}
 2019-07-29 09:41:31 +00:00
Jonas Olsson
857ad62721 Remove priority_rate from AudioStreamConfig. 
This API is going away, we'll use the WebRTC-Audio-Allocation field
trial flag to set this value in the future.

Bug: webrtc:10556
Change-Id: I2c4c1948a33f909fac069dd038cea36a793e4745
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145405
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28608}
 2019-07-19 08:29:55 +00:00
Jonas Olsson
a4d873786f Format almost everything. 
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
 2019-07-08 13:45:15 +00:00
Artem Titov
81e1bf0396 Remove using DegradationPreference from scenario_config.h 
DegradationPreference is already available in namespace webrtc so looks
like there is no reason to redeclare it. Also it cause compilation
error with GCC 5.4.0

Bug: webrtc:10792
Change-Id: I814e90000b8692de67ea477ea7d2769a34a14f01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144523
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28470}
 2019-07-03 14:25:36 +00:00
Sebastian Jansson
e112bb84ef Adds support for abs send time extension in scenario tests. 
Bug: webrtc:10742
Change-Id: I2fba97b23691b27c05dce17ca17c5cd13076616b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141871
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28291}
 2019-06-14 16:26:27 +00:00
Sebastian Jansson
5740afa0a4 Removes SimulatedTimeClient 
Bug: webrtc:9883
Change-Id: Id6e760b37360e7dafc67ded99e06128be20797d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141417
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28269}
 2019-06-13 15:37:10 +00:00
Sebastian Jansson
7ccaf8969d Cleanup of network controller handling in Scenario tests. 
Removing functionality to choose congestion controller implementation,
using injection instead. Also cleaning up some related functionality
that's no longer needed, such as the injection of event logs into the
factory.

Bug: webrtc:9883
Change-Id: Ia528005625430ae31a15bc88881e2d4ac6ad1d42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133890
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27768}
 2019-04-25 12:40:00 +00:00
Sebastian Jansson
a4c22b9662 Using NetworkEmulationManager in Scenario tests. 
Bug: webrtc:9510
Change-Id: Ib619526269c58f0c46c0c1f01ba6c0efa5f79ba5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27635}
 2019-04-16 06:24:26 +00:00
Sebastian Jansson
ef86d1413e Refactor of SimulationNode. 
This prepares for using network emulation manager in Scenario tests.

Bug: webrtc:9510
Change-Id: I6ae1b21790d0bcd2b01a3b293231d0859afc1ac8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132719
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27623}
 2019-04-15 14:11:00 +00:00
Sebastian Jansson
7150d8c60f Refactoring scenario stats analysis. 
This CL just moves code around to prepare for an upcoming
CL where more stats collection is added to scenario tests.

Bug: webrtc:10365
Change-Id: I8a960e44fd11fc36047677c4d8dfc0af96aacb22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132002
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27519}
 2019-04-09 12:47:23 +00:00
Sebastian Jansson
cf2df2fb97 Bases scenario frame matching on similarity. 
Refactoring of quality measurement code, basing frame matching on
frame thumb likeness. This way the code is robust against variations
in timing and frame drops.

Bug: webrtc:9510
Change-Id: Ief7266e01f39ca621a529c0da736e5ed1df8560a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/124401
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27415}
 2019-04-02 14:15:49 +00:00
Sebastian Jansson
cc5be54c26 Support for injection of FEC controller in Scenario tests. 
Also adding sync group for video streams.

Bug: webrtc:10365
Change-Id: I9ef92de756f06bbbcd7b67524bbf51fe1365fa85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130508
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27390}
 2019-04-01 12:30:54 +00:00
Niels Möller
9d8eaac4ee Delete unneeded direct includes of common_types.h 
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:

api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/

There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.

Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
 2019-04-01 07:18:13 +00:00
Niels Möller
0cc2fe5e78 Delete transitional includes from common_types.h 
Also drop unneeded dependencies and a #pragma.

Bug: webrtc:5876, webrtc:7660
Change-Id: I3a46eaf60591b00e43c0647a6eb758e5443de466
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128773
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27311}
 2019-03-27 13:49:07 +00:00
Sebastian Jansson
2b08e3188e Adds CoDel implementation to network simulation. 
Adds an implementation of the CoDel active queue management algorithm
to the network simulation. It is loosely based on CoDel pseudocode
from ACMQueue: https://queue.acm.org/appendices/codel.html

Bug: webrtc:9510
Change-Id: Ice485be35a01dafa6169d697b51b5c1b33a49ba6
Reviewed-on: https://webrtc-review.googlesource.com/c/123581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26834}
 2019-02-25 09:54:03 +00:00
Sebastian Jansson
06c51455fc Adds support for VP9 scalability layers to scenario tests. 
Bug: webrtc:9510
Change-Id: I8d2823114bc921ed3412e3abda5501ce73f5a6fb
Reviewed-on: https://webrtc-review.googlesource.com/c/123042
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26743}
 2019-02-18 18:05:22 +00:00
Sebastian Jansson
f2727fb8d3 Adds slides support to scenario tests. 
Bug: webrtc:9510
Change-Id: I793fb9dbacc916b7b1a95d2fd30683d17a37f1b5
Reviewed-on: https://webrtc-review.googlesource.com/c/123041
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26741}
 2019-02-18 16:24:40 +00:00
Sebastian Jansson
f0c366b461 Cleanup of scenario test video stream setup. 
Removing simulcast stream support as it was broken.

Bug: webrtc:9510
Change-Id: I42ba285bbea81e6ffd5b1d1a1aec4e5eb0990b1e
Reviewed-on: https://webrtc-review.googlesource.com/c/123040
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26684}
 2019-02-14 13:03:15 +00:00
Sebastian Jansson
ad8719442b Adds audio DTX and mute support to scenario tests. 
Bug: webrtc:9510
Change-Id: I50a12c319141dd505309830afdc169c6811c5eca
Reviewed-on: https://webrtc-review.googlesource.com/c/117920
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26288}
 2019-01-17 09:53:48 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
 2019-01-11 17:11:39 +00:00
Sebastian Jansson
9a4f38ec5c Adds optional video quality metrics to scenario tests. 
Bug: webrtc:9510
Change-Id: I448e7156cc8f56930f58c4d25bd167df83a2ba85
Reviewed-on: https://webrtc-review.googlesource.com/c/114885
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26065}
 2018-12-20 08:50:12 +00:00
Sebastian Jansson
f0d031240c Allows injection of network controller in scenarios. 
This makes it possible to test custom network controllers without
requiring update to test framework. Also updating BBR performance
test to use this feature.

Bug: webrtc:9510
Change-Id: I0446de0403fe9d1f6dc3710c1d114887a6c359c5
Reviewed-on: https://webrtc-review.googlesource.com/c/114640
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26046}
 2018-12-18 15:15:05 +00:00
Christoffer Rodbro
5d4740170a Reduce pacing buffer padding rate during pushback. 
Bug: webrtc:10112
Change-Id: I2cd2d07bd5bcbff5b3808ee63eea251a52e45b79
Reviewed-on: https://webrtc-review.googlesource.com/c/113808
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25968}
 2018-12-11 15:22:27 +00:00
Yves Gerey
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.

Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
 2018-11-28 18:25:07 +00:00
Sebastian Jansson
0c32e33b48 Allows change of fake encoder max rate in scenarios tests. 
Bug: webrtc:9510
Change-Id: I13010c7febe8c31de78178611915a2b9e2f9869f
Reviewed-on: https://webrtc-review.googlesource.com/c/110612
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25608}
 2018-11-12 16:50:58 +00:00
Sebastian Jansson
985ee68dc4 Add support for screenshare content type in scenario tests. 
Bug: webrtc:9510
Change-Id: Icd15696e5a57a8e93223933f6ccd23687115e29a
Reviewed-on: https://webrtc-review.googlesource.com/c/110613
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25607}
 2018-11-12 16:43:48 +00:00
Sebastian Jansson
2b101d2c9e Simplifies audio priority rate config in scenario tests. 
Bug: webrtc:9510
Change-Id: Iecd2caa8d4353c64ec351969f999c8ed59c3a07d
Reviewed-on: https://webrtc-review.googlesource.com/c/110614
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25606}
 2018-11-12 16:30:21 +00:00
Sebastian Jansson
800e121dca Adds support to change transport routes in Scenario tests. 
This CL makes it possible to change transport routes while running
a scenario based test.

To make this possible in a consistent manner, the scenario test
framework is modified to only allow shared transport for all streams
between two CallClients. This is what typically is done in practice and
it is quite complex to even reason about the implications of using
mixed transports for a single call.

Bug: webrtc:9718
Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b
Reviewed-on: https://webrtc-review.googlesource.com/c/107200
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25287}
 2018-10-22 11:14:37 +00:00
Sebastian Jansson
b9972fa37b Adds AudioNetworkAdaptation support to Scenario tests. 
Bug: webrtc:9718
Change-Id: I6cb976df5767797fec670134d29e030ec0f9d3a2
Reviewed-on: https://webrtc-review.googlesource.com/c/106340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25236}
 2018-10-17 15:42:58 +00:00
Sebastian Jansson
71a091e24e Adds simulated time scenario client. 
Adds SimulatedTimeClient, a class that simulates time so congestion
controllers can be tested using the Scenario test framework without
running in real time.

This allows using simplified scenario tests as unit tests, narrowing
the gap between end to end tests and unit tests.

Bug: webrtc:9510
Change-Id: I61ab388bd610f636b926675b1f14b8d85e3c1114
Reviewed-on: https://webrtc-review.googlesource.com/99801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24890}
 2018-09-28 12:30:44 +00:00
Sebastian Jansson
98b07e9180 Adds scenario test framework. 
Bug: webrtc:9510
Change-Id: I387aab4211f520a1c54832f82032ee724479e89e
Reviewed-on: https://webrtc-review.googlesource.com/89342
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24864}
 2018-09-27 12:31:33 +00:00