Commit graph

31 commits

Author SHA1 Message Date
Danil Chapovalov
eb90e6ffe3 Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest
That allows to use SingleThreadedTaskQueueForTesting via TaskQueueBase interface
but still have access to test-only SendTask function.

Bug: webrtc:10933
Change-Id: I3cc397e55ea2f1ed9e5d885d6a2ccda412beb826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29480}
2019-10-15 09:17:36 +00:00
Sebastian Jansson
f34116e356 Replacing bandwidth adaptation trial with stable target in Opus encoder.
This also means that the NetworkEstimate::bandwidth can be deprecated
as it's currently just a copy of the target_rate.

Bug: webrtc:10981
Change-Id: I1bc57b98480bd77ce052736b19d630c775428546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29288}
2019-09-24 16:35:02 +00:00
Sebastian Jansson
ee5ec9a93a Replacing local closure classes with C++14 moving capture lambdas.
Bug: webrtc:10945
Change-Id: I569b9495cae98f204065911e13c37c31f35da372
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153241
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29214}
2019-09-17 19:43:05 +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
Tommi
25eb47ccf1 Make the RtpHeaderParserImpl available to tests and tools only.
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
  (a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
  See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
  sufficient for most production cases.

Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
2019-08-29 15:56:40 +00:00
Florent Castelli
4e615d590a Wire the stable target bitrate from GoogCC to the BitrateAllocator
Deprecated the field BitrateAllocationUpdate::link_capacity since it is only
used by the Opus codec in order to smooth the target bitrate, which is
equivalent to the stable_target_bitrate field.

The unused field trial WebRTC-Bwe-StableBandwidthEstimate is also removed.

Bug: webrtc:10126
Change-Id: Ic4a8a9ca4202136d011b91dc23c3a27cfd00d975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149839
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28941}
2019-08-22 15:25:15 +00:00
Tommi
e6b7b6678c Fix CallClient so that it calls Call::GetStats() on the right thread.
Bug: webrtc:10847
Change-Id: Id23a389b4d5bad8f2211b5ec87b37aefc81a9292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148065
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28772}
2019-08-06 11:16:41 +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
Sebastian Jansson
49167de0be Adds interface for remote network estimates to NetworkControllerInterface.
Bug: webrtc:10742
Change-Id: I593fc17ce5d42c5dc17fd289f0621230319f9752
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144039
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28405}
2019-06-27 16:57:32 +00:00
Sebastian Jansson
a7d70ab0fe Adds network update state cache to scenario tests.
Bug: webrtc:9510
Change-Id: I1ffbc21770f0442086491a07ac7970eaf594a05f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141400
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28223}
2019-06-11 11:16:04 +00:00
Sebastian Jansson
58c71db1b3 Fix for crash in event log when using scenario tests.
Scenario tests runs all its activities on task queues. This is not
allowed by the default event log writer, causing a DCHECK failure.
This CL makes it possible to stop the event asynchronously,
thereby avoiding the need for the DCHECK.

Bug: webrtc:10365
Change-Id: I1206982b29fd609ac85b4ce30ae9291cbec52041
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136685
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28027}
2019-05-22 15:22:49 +00:00
Danil Chapovalov
b32f2c7f57 Publish rtc event log api and default factory for it in api/
Bug: webrtc:10206
Change-Id: I34194ddb6fd2b0a3d7c553fadc9ddc1ea9740da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28023}
2019-05-22 13:38:25 +00:00
Sebastian Jansson
871ac42597 Refactor of GoogCC debug printer.
Simplifying the code to better fit with how it is used.

Bug: webrtc:9883
Change-Id: I2bd52f26b829413e516dee4f551cf36574275019
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136681
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27994}
2019-05-20 18:40:26 +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
Artem Titov
4cd433ed5f Switch routing from dest endpoint id to dest ip in network emulation layer
Bug: webrtc:10138
Change-Id: I8659777820ec7244c1435d71504b0c01fb438192
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130103
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27386}
2019-04-01 10:04:41 +00:00
Danil Chapovalov
359fe33163 Move TaskQueueFactory from Call::Create parameter to CallConfig
to decouple it from other optional parameters
and with plan to make it mandatory

Bug: webrtc:10284
Change-Id: I71c1d3d9eaf09d00b99b0bc4c811ab173ea5f01f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130473
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27385}
2019-04-01 09:27:44 +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
1e42761b39 Removes verbose extension warning in Scenario tests.
Bug: webrtc:9510
Change-Id: I017119a899c68d27fe4b4376afb4070ff89b4f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125540
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26991}
2019-03-06 14:00:11 +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
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
40d55331d7 Include absl/memory/memory.h if absl::make_unique is used
Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Iaf4533d2ce0e80b351a8a664ef8cf7ba0e5ec583
Reviewed-on: https://webrtc-review.googlesource.com/c/115746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26168}
2019-01-08 20:08:32 +00:00
Artem Titov
40f511540b Extract NetworkReceiverInterface and introduce EmulatedIpPacket
It is a step in the big refactoring to introduce new network emulation
layer for peer connection level e2e test, which will be based on system
sockets level injection.

Bug: webrtc:10138
Change-Id: Ie3854d22aa3eec289617bc432026ea670646556a
Reviewed-on: https://webrtc-review.googlesource.com/c/115943
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26137}
2019-01-04 15:19:13 +00:00
Sebastian Jansson
f65309cd47 Removes return value and Try prefix from TryDeliverPacket.
The return value is not used. This change prepares for future
refactoring by removing the requirement that TryDeliverPacket must be
synchronous. Also renaming to DeliverPacket as we no longer need to
indicate the meaning of the return value.

Bug: webrtc:9510
Change-Id: I78536434b198fa7bf4df88b10d6add23684767f1
Reviewed-on: https://webrtc-review.googlesource.com/c/115181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26066}
2018-12-20 10:04:17 +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
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
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
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
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