36 Commits

Author SHA1 Message Date
Sebastian Jansson
49a7843030 Don't restart streams in scenario tests.
This CL changes the behavior for RunFor and RunUntil so they do not
anymore restart the underlying streams every time they are called.

This has a side effect on the semantics of the calls. Previously,
both RunUntil and RunFor would restart the session and run until the
given time had passed. Now RunFor will still run for the provided
duration, however, to make the name of RunUntil more correct, it
will run until the time since start is equal to the max_duration
parameter. An extra overload of RunUntil was added to allow using
this behavior without providing an ending condition.

Bug: webrtc:9510
Change-Id: I9fe56a44116907fba3d102894b5c96af2ba6cffb
Reviewed-on: https://webrtc-review.googlesource.com/c/111502
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25726}
2018-11-21 13:16:46 +00:00
Niels Möller
2222a80e79 Delete unneeded includes of common_types.h and gn deps on webrtc_common.
Bug: webrtc:5876
Change-Id: Iae14e5f1679067a5a5e0584ca830aee0870c8807
Reviewed-on: https://webrtc-review.googlesource.com/c/111463
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25715}
2018-11-20 16:28:39 +00:00
Sebastian Jansson
fd20171d28 Adds setup of RTP Extensions in Scenario tests.
This prevents printing warning messages when the extensions aren't
found. The real parsing is done deeper in the stack and is unaffected.

Bug: webrtc:9510
Change-Id: Idf09f0e69c223bd4217be7044d21d1d0bbbdab92
Reviewed-on: https://webrtc-review.googlesource.com/c/110615
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25612}
2018-11-13 09:34:09 +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
Jiawei Ou
c2ebe21ba9 Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer"
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782

This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.

Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.

One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.

Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 19:10:47 +00:00
Niels Möller
c572ff3c71 Add default constructor for rtc::Event
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
2018-11-07 08:57:50 +00:00
Sebastian Jansson
af6d741fe1 Makes send time information in feedback non-optional.
This makes it safer to reason about the common case where send
time information is available. We don't have to either assume that
it's available, or check it everywhere the PacketResult struct is used.

To achieve this, a new field is added to TransportPacketsFeedback
and a new interface is introduced to clearly separate which field is
used. A possible followup would be to introduce a separate struct.
That would complicate the signature of ProcessTransportFeedback.

Bug: webrtc:9934
Change-Id: I2b319e4df2b557fbd4de66b812744bca7d91ca15
Reviewed-on: https://webrtc-review.googlesource.com/c/107080
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25465}
2018-11-01 12:39:56 +00:00
Qingsi Wang
59844ce57e Revert "Use the factory instead of using the builtin code path in VideoCodecInitializer."
This reverts commit be142178aaf6ab4089b4d81c88c3d59c12cca567.

Reason for revert: breaking internal projects

Original change's description:
> Use the factory instead of using the builtin code path in `VideoCodecInitializer`.
> 
> Bug: webrtc:9513
> Change-Id: Ia299ae1044a3ff4c91e208200938cba540bdcea6
> Reviewed-on: https://webrtc-review.googlesource.com/c/94782
> Commit-Queue: Jiawei Ou <ouj@fb.com>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25456}

TBR=brandtr@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,tommi@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,tkchin@webrtc.org,shampson@webrtc.org,glaznev@webrtc.org,ouj@fb.com,qingsi@webrtc.org

Change-Id: I8040ccabe3ae6464d72c7696adb663c1dd275b63
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9513
Reviewed-on: https://webrtc-review.googlesource.com/c/108980
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25459}
2018-11-01 04:46:02 +00:00
Jiawei Ou
be142178aa Use the factory instead of using the builtin code path in VideoCodecInitializer.
Bug: webrtc:9513
Change-Id: Ia299ae1044a3ff4c91e208200938cba540bdcea6
Reviewed-on: https://webrtc-review.googlesource.com/c/94782
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25456}
2018-10-31 22:47:02 +00:00
Sebastian Jansson
ed45c57d98 Corrects audio overhead correction in Scenario test.
This makes the calculation more similar to the one in WebRTCVoiceEngine.

Bug: webrtc:9510
Change-Id: Ibca69842726e51c07b9cc9550ff9f15a24161e28
Reviewed-on: https://webrtc-review.googlesource.com/c/107653
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25448}
2018-10-31 11:06:39 +00:00
Yves Gerey
21cddffd99 Harmonize paths to dependent targets.
This CL consistently use:
 * relative paths for WebRTC dependent targets (test_support)
 * absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.

We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.

Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-31 10:04:59 +00:00
Danil Chapovalov
99b71dfd4a Use function_video_(en|de)coder_factory from api
Remove them from test.
It is completion of the move started with
https://webrtc-review.googlesource.com/c/src/+/107705

Bug: None
Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/107891
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25392}
2018-10-26 14:54:19 +00:00
Niels Möller
7d76a31f3d Use MediaTransportInterface, for audio streams.
Bug: webrtc:9719
Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239
Reviewed-on: https://webrtc-review.googlesource.com/c/104642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25385}
2018-10-26 11:40:57 +00:00
Sebastian Jansson
359d60a594 Adds target rate to audio send stream stats.
Bug: webrtc:9510
Change-Id: I8bd74fc115e3006f477b289edc58fa1f9d7b6bc6
Reviewed-on: https://webrtc-review.googlesource.com/c/107652
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25370}
2018-10-25 15:12:36 +00:00
Sebastian Jansson
0627e21a1a Removes unused DeliverPacket from CallClient.
Bug: webrtc:9510
Change-Id: Idfdce13ef407449c3896ad400ab4b8fb3ef589a1
Reviewed-on: https://webrtc-review.googlesource.com/c/107420
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25308}
2018-10-23 09:30:46 +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
Mirko Bonadei
2dfa998be2 Reland "Prefix flag macros with WEBRTC_."
This is a reland of 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d

Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

TBR=kwiberg@webrtc.org

Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
2018-10-19 15:06:43 +00:00
Mirko Bonadei
c538fc77b0 Revert "Prefix flag macros with WEBRTC_."
This reverts commit 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d.

Reason for revert: Breaks downstream project.

Original change's description:
> Prefix flag macros with WEBRTC_.
> 
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
> 
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
> 
> This CL adds the 'WEBRTC_' prefix to them.
> 
> Generated with:
> 
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
> 
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

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

Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
2018-10-19 15:04:13 +00:00
Mirko Bonadei
5ccdc1331f Prefix flag macros with WEBRTC_.
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).

They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.

This CL adds the 'WEBRTC_' prefix to them.

Generated with:

for x in DECLARE DEFINE; do
  for y in bool int float string FLAG; do
    git grep -l "\b$x\_$y\b" | \
    xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
  done
done
git cl format

Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
2018-10-19 10:55:20 +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
8285841e8f Adds handling of untracked data to congestion controller.
Bug: webrtc:9796
Change-Id: I097e8f72a6c8d323c3ea73dbb4ade60873dd4e8d
Reviewed-on: https://webrtc-review.googlesource.com/c/104883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25129}
2018-10-11 18:47:44 +00:00
Sebastian Jansson
0378997db3 Adds flags indicating presence in allocation and feedback per packet.
This CL adds flags to the PacketOptions and PacktInfo struct that are
intended to be used to indicate if the packet belongs to a media stream
that is part of bitrate allocation as well as if it is included in
transport wide packet feedback.

This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.

Bug: webrtc:9796
Change-Id: Icdf3e1e13d3f119574ee1b2c574f2d3329a7e303
Reviewed-on: https://webrtc-review.googlesource.com/c/104920
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25069}
2018-10-09 18:24:38 +00:00
Yves Gerey
2e00abc98e Reland "[cleanup] Remove useless includes."
Reason for reland: Downstream project fixed.

Original change's description:

> [cleanup] Remove useless includes.
>
> Manual cleanup guided by include-what-you-use diagnostic.
>
> Bug: webrtc:8311
> Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68
> Reviewed-on: https://webrtc-review.googlesource.com/c/103320
> Commit-Queue: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25013}

Bug: webrtc:8311
Change-Id: Id6ec4aeb798886a90ace640a190eaf16497ba31b
Reviewed-on: https://webrtc-review.googlesource.com/c/104120
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25034}
2018-10-08 07:44:19 +00:00
Stefan Holmer
64be7fa7d8 Move FecController to RtpVideoSender.
This also moves the packet feedback tracking to RtpVideoSender.

Bug: webrtc:9517
Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e
Reviewed-on: https://webrtc-review.googlesource.com/c/95920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25019}
2018-10-05 14:39:01 +00:00
Oleh Prypin
96a0f61917 Revert "[cleanup] Remove useless includes."
This reverts commit be8b5348c76105f8fe869b0cae4065ddca106419.

Reason for revert: Breaks downstream project

Original change's description:
> [cleanup] Remove useless includes.
> 
> Manual cleanup guided by include-what-you-use diagnostic.
> 
> Bug: webrtc:8311
> Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68
> Reviewed-on: https://webrtc-review.googlesource.com/c/103320
> Commit-Queue: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25013}

TBR=phoglund@google.com,phoglund@webrtc.org,yvesg@webrtc.org

Change-Id: I7a6e1cdfef685173b76f234ad598083043dcd9a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8311
Reviewed-on: https://webrtc-review.googlesource.com/c/104022
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25015}
2018-10-05 13:13:45 +00:00
Yves Gerey
be8b5348c7 [cleanup] Remove useless includes.
Manual cleanup guided by include-what-you-use diagnostic.

Bug: webrtc:8311
Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68
Reviewed-on: https://webrtc-review.googlesource.com/c/103320
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25013}
2018-10-05 11:51:06 +00:00
Artem Titov
8ea1e9def1 Switch webrtc from deprecated usages of NetworkSimulationInterface
Bug: webrtc:9630
Change-Id: I42222261676b0c260c1aab81523a23988d3cd1c1
Reviewed-on: https://webrtc-review.googlesource.com/c/103780
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25011}
2018-10-05 11:01:42 +00:00
Mirko Bonadei
e0d455b409 Remove runtime_enabled_feature.
This features is not needed anymore, with this CL it is also possible
to address two issues:
- The need to pick a default implementation.
- The need to use -Wno-global-constructors.

Bug: webrtc:9631, webrtc:9693
Change-Id: Id3daf34179fbc8db26969fc701ccbfa7182c6a9b
Reviewed-on: https://webrtc-review.googlesource.com/102543
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24904}
2018-10-01 07:03:25 +00:00
Sebastian Jansson
156d11ddd9 Adds packet_size to rtc::SentPacket in testing code.
Bug: webrtc:9796
Change-Id: Id67bb02858164dba696474b1b60ebfa1597a2577
Reviewed-on: https://webrtc-review.googlesource.com/102685
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24901}
2018-09-29 22:06:07 +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
Niels Möller
cbcbc22568 Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
This is a reland of 529d0d9795b81dbed5e4231f15d3752a5fc0df32

Original change's description:
> Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
> 
> Preparation for deleting EnableFrameRecordning, and also a step
> towards landing of the new VideoStreamDecoder.
> 
> Bug: webrtc:9106
> Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
> Reviewed-on: https://webrtc-review.googlesource.com/97660
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24861}

Bug: webrtc:9106
Change-Id: I2eb894773b3f33ff6a980e8008e8248607e32668
Reviewed-on: https://webrtc-review.googlesource.com/102480
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24882}
2018-09-28 08:48:02 +00:00
Mirko Bonadei
17f4878419 Remove deprecated field_trial_default and metrics_default.
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default

It also refreshes all the dependencies on field_trial.h and metrics.h.

A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm

Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 07:21:07 +00:00
Sebastian Jansson
377b26ec65 Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
This reverts commit efb94d57eb88638c323d93dddc281390dada5021.

Reason for revert: Investigate and fix build errors.

Original change's description:
> Revert "Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.""
>
> This reverts commit 7961dc2dbdb3391a003d63630d5107e258ff3e78.
>
> Reason for revert: WebRTC does not build
>
> Original change's description:
> > Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
> >
> > This reverts commit 529d0d9795b81dbed5e4231f15d3752a5fc0df32.
> >
> > Reason for revert: Seems to break perf tests, likely some breakage in video_quality_tests decoder configuration.
> >
> > Original change's description:
> > > Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
> > >
> > > Preparation for deleting EnableFrameRecordning, and also a step
> > > towards landing of the new VideoStreamDecoder.
> > >
> > > Bug: webrtc:9106
> > > Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
> > > Reviewed-on: https://webrtc-review.googlesource.com/97660
> > > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#24861}
> >
> > TBR=brandtr@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org
> >
> > Change-Id: Id34e4a3452a7dbc06167a4df5bb4c2825ebd7bd0
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:9106
> > Reviewed-on: https://webrtc-review.googlesource.com/102421
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24866}
>
> TBR=brandtr@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org
>
> Change-Id: I23a439e1ceef79109b1f966b80b2663203968269
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9106
> Reviewed-on: https://webrtc-review.googlesource.com/102422
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24867}

TBR=brandtr@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org

Change-Id: I9dafbc070e7f39dcb0ddbd61cb620164258fe894
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9106
Reviewed-on: https://webrtc-review.googlesource.com/102460
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24872}
2018-09-27 16:04:50 +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