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59 Commits

Author SHA1 Message Date
Tommi
87f7090fd9 Replace more instances of rtc::RefCountedObject with make_ref_counted. 
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}
 2021-04-27 17:01:59 +00:00
Mirko Bonadei
5686e3457e Optimize calls to std::string::find() and friends for a single char. 
The character literal overload is more efficient.

No-Presubmit: True
No-Try: True
Bug: None
Change-Id: Ice0b8478accd8a252ab81a0496d46c0f71db3db6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197810
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32841}
 2020-12-16 09:01:44 +00:00
Erik Språng
279f37052c Makes WebRTC-Pacer-SmallFirstProbePacket default enabled. 
This is expected to yield slightly higher bandwidth estimates when
probing is used, since it reduces a bias in how packet sizes are counted.

Bug: webrtc:11780
Change-Id: I6a4a3af0c50670d248dbe043a4d9da60915e3699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187491
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32394}
 2020-10-13 21:45:42 +00:00
Erik Språng
3e3e16682d Migrates probing end-to-end test to scenario test. 
The previous tests ran in real-time making them flaky, so they were
disabled on a number of platforms.
This CL ports the tests 1:1 (sort of) to use the scenario test
framework which runs with simulated time and much less risk of
flakiness.

Bug: webrtc:10155
Change-Id: I6281f57d73883c8aaa91964e9cfa58d9b47779fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32333}
 2020-10-06 20:51:35 +00:00
Philip Eliasson
2b068ce1b8 Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." 
This reverts commit a4f23ad0ce4382e3a11bc6a8c1f9f6183e722fd8.

Reason for revert: Downstream fix landed.

TBR=mflodman@webrtc.org

Original change's description:
> Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
>
> This reverts commit acb9d8365a5f9eb1e2a9e9902690d62dab1e5759.
>
> Reason for revert: Break downstream stuff.
>
> Original change's description:
> > Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory.
> >
> > Bug: webrtc:9106
> > Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31834}
>
> TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org
>
> Change-Id: I6cfdb85a154a78135839f84edf5f69673d5ab715
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9106
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180807
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31835}

TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org

# Not skipping CQ checks because this is a reland.

Bug: webrtc:9106
Change-Id: I03b3e68532107bec37bcc6e47a5489c84fe91ef9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180808
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31866}
 2020-08-06 11:50:08 +00:00
Philip Eliasson
a4f23ad0ce Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." 
This reverts commit acb9d8365a5f9eb1e2a9e9902690d62dab1e5759.

Reason for revert: Break downstream stuff.

Original change's description:
> Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory.
> 
> Bug: webrtc:9106
> Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31834}

TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org

Change-Id: I6cfdb85a154a78135839f84edf5f69673d5ab715
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9106
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180807
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31835}
 2020-08-03 15:45:41 +00:00
philipel
acb9d8365a Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory. 
Bug: webrtc:9106
Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31834}
 2020-08-03 14:10:37 +00:00
Markus Handell
a5a4be1118 Partly migrate test/ to webrtc::Mutex. 
This change migrates test/, except for subdirs
  - test/time_controller
  - test/pc/e2e

Bug: webrtc:11567
Change-Id: Ib6f7c062f1c66caf7083fb4ec60727d66299dbeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178819
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31675}
 2020-07-08 14:31:00 +00:00
Mirko Bonadei
06d3559b79 Replace std::string::find() == 0 with absl::StartsWith (part 2). 
This CL has been generated using clang-tidy [1] except for changes to
BUILD.gn files.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/abseil-string-find-startswith.html

Bug: None
Change-Id: Ibf75601065a53bde28623b8eef57bec067235640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172586
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30984}
 2020-04-02 14:38:30 +00:00
Nikita Zetilov
8e9fd4857e Fix FakeVp8Encoder name. 
Bug: None
Change-Id: Iaa11a452fcb6fb6f33d1396eb4e6fe9c050166ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169845
Commit-Queue: Nikita Zetilov <zetilovn@google.com>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30703}
 2020-03-06 11:12:21 +00:00
Sebastian Jansson
8ad3427d7f Use the last video stream for scenario tests stats. 
This makes slightly more sense when looking at video resolution etc.

Bug: webrtc:9510
Change-Id: I49d39cac23d2f5d7ca09f2a27152c7519ea639f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169344
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30632}
 2020-02-27 14:52:54 +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
Sebastian Jansson
3d4d94a832 Adds scenario test for transport wide feedback based retransmission. 
This ensures more end to end test coverage of the feature and captures
a wider class of regression then the existing unit test.

Bug: webrtc:9883
Change-Id: I6e74e571500c5c5d74caf8f661cac08bee8934f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164461
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30252}
 2020-01-14 16:00:14 +00:00
Sebastian Jansson
3e66a498c3 Use RTX SSRCs in scenario test framework. 
Using RTX SSRCs and payload type for retransmission of video. This
corresponds to the behavior when using the peer connection API.

Bug: webrtc:9883
Change-Id: Ic0e3964d097f42219ca225513a4bc771d70ccaa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164460
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30248}
 2020-01-14 12:04:56 +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
Danil Chapovalov
dc36829db0 Add VideoCodecType::kVideoCodecAV1 value 
Bug: webrtc:11042
Change-Id: I3c5151c9e47679760f8f7d79270488fa8f4c7db5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159282
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29927}
 2019-11-27 10:18:45 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique. 
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
 2019-09-17 15:47:29 +00:00
Niels Möller
7bf7a427bf Delete flag VideoReceiveStream::Config::Rtp::remb 
This flag became unused in https://codereview.webrtc.org/2789843002;
it was set, but the setting had no effect.

Bug: webrtc:7135
Change-Id: I012a7c3600bc7a371c7a589695823b30ed5647a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29192}
 2019-09-16 11:20:55 +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
Sebastian Jansson
c57b0ee11c Fix for NACK retransmission in Scenario tests. 
This fixes a bug where NACK mode was not properly enabled
due to missing send side configuration.

Bug: webrtc:9510
Change-Id: I318fdf44f17e57d30589115a452f6a64f81ee973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28391}
 2019-06-26 15:23:57 +00:00
Sebastian Jansson
e9cac4fe83 Improvements to scenario video stats for scenario tests. 
* Adds capture to decode time.
* Calculating PSNR only for delivered frames, keeping the old PSNR
  value including freezes as a separate field.
* Calculates end to end delay only for delivered frames.
* Adds Count member for stats collectors.
* Minor cleanups.

Bug: webrtc:10365
Change-Id: Iaa7b1f0666a10764a513eecd1a08b9b6e76f3bc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142812
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28355}
 2019-06-24 15:42:42 +00:00
Sebastian Jansson
342f98b117 Fixes for flexfec crash in scenario tests. 
Bug: webrtc:9510
Change-Id: I39bb4ed9afc4837f88f0db798495f34b685f4c24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142232
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28309}
 2019-06-18 15:03:41 +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
24cf2606e4 Adds TCP fairness test to receive side congestion controller. 
Bug: webrtc:9883
Change-Id: I3697491285e4f70b8f7857198e4e1ccb0097da5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28196}
 2019-06-07 13:49:37 +00:00
Sebastian Jansson
e82266836b Adds control of active layers in scenario tests. 
Bug: webrtc:9510
Change-Id: I889633650885d8ceb529c879febee811e4934b47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134647
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27821}
 2019-05-02 10:13:26 +00:00
Sebastian Jansson
5a000165d2 Cleanup: Using DCHECK comparison macros for unit types. 
This provides nicer error messages.

Bug: webrtc:9883
Change-Id: I7664c12f34bec2ba46a4057b1f45958daf3944b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132707
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27590}
 2019-04-12 13:01:03 +00:00
Sebastian Jansson
f4481c8f73 Adds GetStats to scenario receive streams. 
Bug: webrtc:10365
Change-Id: Ia17213701dee7de63018aa935261c17f07a40015
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132005
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27515}
 2019-04-09 11:44:20 +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
Sebastian Jansson
105a10aef0 Using TimeController for Scenario test framework 
As part of this change, a task queue is used to handle packet
processing in real time mode. This requires that we also do
most call and media stream related operation on the same task
queue to satisfy thread checkers.

Bug: webrtc:10365
Change-Id: Icdd9d56e4ca14f2c944dc655c91e29392e3765f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127544
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27379}
 2019-04-01 07:52:35 +00:00
Sebastian Jansson
d37307c561 Reland "Adds resource path support for video files in scenario tests." 
This is a reland of 8306a733f0dc45f19462268e29c90ada9f46b28e

Original change's description:
> Adds resource path support for video files in scenario tests.
> 
> Bug: webrtc:9510
> Change-Id: Id41a32325cc5b16b119e62fba483cec88f52975b
> Reviewed-on: https://webrtc-review.googlesource.com/c/123189
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26804}

Bug: webrtc:9510
Change-Id: I97a5568063569ca66d87f28204200a582d01e2e1
Reviewed-on: https://webrtc-review.googlesource.com/c/123960
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26836}
 2019-02-25 11:25:54 +00:00
Sergey Silkin
45af00f33b Revert "Adds resource path support for video files in scenario tests." 
This reverts commit 8306a733f0dc45f19462268e29c90ada9f46b28e.

Reason for revert: ReceivesFramesFromFileBasedStreams is flaky.

Original change's description:
> Adds resource path support for video files in scenario tests.
> 
> Bug: webrtc:9510
> Change-Id: Id41a32325cc5b16b119e62fba483cec88f52975b
> Reviewed-on: https://webrtc-review.googlesource.com/c/123189
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26804}

TBR=ilnik@webrtc.org,srte@webrtc.org

Change-Id: I3b157a58bfaf6bcd3dfd9a9d2573a0edd3e6eeab
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9510
Reviewed-on: https://webrtc-review.googlesource.com/c/123880
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26807}
 2019-02-22 08:11:57 +00:00
Sebastian Jansson
8306a733f0 Adds resource path support for video files in scenario tests. 
Bug: webrtc:9510
Change-Id: Id41a32325cc5b16b119e62fba483cec88f52975b
Reviewed-on: https://webrtc-review.googlesource.com/c/123189
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26804}
 2019-02-21 18:52:52 +00:00
Sebastian Jansson
5fbebd585e Adds support for VP8 simulcast to scenario tests. 
Bug: webrtc:9510
Change-Id: Ice98e7bd98a1a8e4fd3b1a1c7c053a65de3f56e3
Reviewed-on: https://webrtc-review.googlesource.com/c/123380
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26764}
 2019-02-20 10:51:39 +00:00
Elad Alon
d8d3248d95 Reland "Delete test/constants.h" 
This reverts commit 4f36b7a478c2763463c7a9ea970548ec68bc3ea6.

Reason for revert: Failing tests fixed.

Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}

TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}
 2019-02-19 08:51:20 +00:00
Oleh Prypin
4f36b7a478 Revert "Delete test/constants.h" 
This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de.

Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate

Original change's description:
> Delete test/constants.h
>
> It's not possible to use constants.h for all RTP extensions
> after the number of extensions exceeds 14, which is the maximum
> number of one-byte RTP extensions. This is because some extensions
> would have to be assigned a number greater than 14, even if the
> test only involves 14 extensions or less.
>
> For uniformity's sake, this CL also edits some files to use an
> enum as the files involved in this CL, rather than free-floating
> const-ints.
>
> Bug: webrtc:10288
> Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26728}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

No-Presubmit: True
Bug: webrtc:10288, chromium:933127
Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
Reviewed-on: https://webrtc-review.googlesource.com/c/123381
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26744}
 2019-02-18 18:09:22 +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
Elad Alon
389b1672a3 Delete test/constants.h 
It's not possible to use constants.h for all RTP extensions
after the number of extensions exceeds 14, which is the maximum
number of one-byte RTP extensions. This is because some extensions
would have to be assigned a number greater than 14, even if the
test only involves 14 extensions or less.

For uniformity's sake, this CL also edits some files to use an
enum as the files involved in this CL, rather than free-floating
const-ints.

Bug: webrtc:10288
Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/123048
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26728}
 2019-02-17 21:47:41 +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
bdfadd666e Adds Stop methods to media streams in scenario framework. 
Bug: webrtc:9510
Change-Id: If011e701496850dd67394052edd5a6d14a3998be
Reviewed-on: https://webrtc-review.googlesource.com/c/121951
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26609}
 2019-02-08 13:21:20 +00:00
Christoffer Rodbro
5f6abcfbd2 Fix for RttBackoff when sending of packets with TWCC stops. 
Bug: webrtc:10290
Change-Id: Ia825cbde070214e5ec9f5439246ea43f58c3c2b7
Reviewed-on: https://webrtc-review.googlesource.com/c/121561
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26605}
 2019-02-08 10:47:03 +00:00
Mirko Bonadei
739baf097b [clang-tidy] Apply performance-for-range-copy fixes. 
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.

All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html

Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
 2019-01-28 09:53:50 +00:00
Sebastian Jansson
52de8b0270 Adds functionality to write logs to memory. 
This makes it possible to save log outputs from scenario tests to
either files or memory.

Bug: webrtc:9510
Change-Id: I883bd8240ab712d31d54118adf979041bd83481a
Reviewed-on: https://webrtc-review.googlesource.com/c/116321
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26284}
 2019-01-16 17:36:31 +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
Niels Möller
8eeccbe6a6 Delete Start and Stop methods from TestVideoCapturer. 
Preparation for replacing use of TestVideoCapturer as an interface,
instead using VideoSourceInterface.

Methods kept as non-virtual on the subclass FrameGeneratorCapturer,
but it's changed to be started on creation.

Bug: webrtc:6353
Change-Id: Iae1c9a0ee55d730d4992204f62227ef2f057d58e
Reviewed-on: https://webrtc-review.googlesource.com/c/114425
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26037}
 2018-12-18 09:29:52 +00:00
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
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