19 Commits

Author SHA1 Message Date
Ali Tofigh
b7821cea6b Remove unnecessary overload in RtcEventLogOutput
Bug: webrtc:13579
Change-Id: I3ea4b8ce8d111ae6b9ce7e92f75bd4196bc9656b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268420
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37508}
2022-07-12 22:09:36 +00:00
Björn Terelius
63299a3124 Add absl::string_view overload for RtcEventLogOutput::Write
Bug: webrtc:13579
Change-Id: I13f63fb6be6aa62c2e011c18327499fa16b5824e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267641
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37440}
2022-07-05 10:47:47 +00:00
Tommi
3176ef79e9 Rename AudioReceiveStream to AudioReceiveStreamInterface
Bug: webrtc:7484
Change-Id: I22eaa7a9e082fc575cf7471d7a2f4f706564d54f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262805
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36965}
2022-05-23 08:44:26 +00:00
Tommi
f6f4543304 Rename VideoReceiveStream to VideoReceiveStreamInterface
Bug: webrtc:7484
Change-Id: I653cfe46486e0396897dd333069a894d67e3c07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262769
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36958}
2022-05-22 10:54:38 +00:00
Henrik Boström
2deee4bbb2 Mark rtc::Thread's versions of PostTask/PostDelayedTask deprecated.
Because rtc::Thread inherits from TaskQueueBase, it already implements
a pair of PostTask/PostDelayedTask methods that we want to keep. But in
addition to those, rtc::Thread defines its own PostTask/PostDelayedTask
using templates. These are the versions that we want to deprecate.

They were originally implemented prior to rtc::Thread inheriting from
TaskQueueBase. We want to deprecate them because...
- We don't want to have multiple code paths that do the same thing.
- We want to move away from rtc::Thread to TaskQueueBase long-term.
- These versions are not overridable in Chromium.
- These versions don't have high/low precision versions of PDT.

Helper methods are added to rtc::Thread so that callers don't have to
wrap every lambda in webrtc::ToQueuedTask() and update dependencies.

Bug: webrtc:13582
Change-Id: I58702c53f4cb3705681bd9f1ea16b7aaa5052c18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35750}
2022-01-20 12:59:27 +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
6e07cde22c Accept undecoded frame pairs in VideoLayerAnalyzer
Bug: webrtc:9883
Change-Id: I651bf21ebbf547389b36df077f6ff619c5e670b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168043
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30442}
2020-02-03 09:46:55 +00:00
Niels Möller
a837030f8f Split out RtpSource from libjingle_peerconnection_api
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.

Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
2019-09-02 14:04:47 +00:00
Sebastian Jansson
7cbee84610 Reland "Adds PeerConnection scenario test framework."
This is a reland of ad5c4accad00e04de08e2b62d366cc1f8e0320a5

It was flaky due to starting ICE signaling before SDP negotiation
finished. This was solved by adding an helper for adding ice candidates
which will wait until the peer connection is ready if needed.

Original change's description:
> Adds PeerConnection scenario test framework.
>
> Bug: webrtc:10839
> Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28754}

Bug: webrtc:10839
Change-Id: I6eb8f482561c87e7b0f20d2431d21a41b26c91d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147877
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28777}
2019-08-06 16:12:12 +00:00
Sebastian Jansson
3d351c6885 Revert "Adds PeerConnection scenario test framework."
This reverts commit ad5c4accad00e04de08e2b62d366cc1f8e0320a5.

Reason for revert: Breaks downstream bots.

Original change's description:
> Adds PeerConnection scenario test framework.
> 
> Bug: webrtc:10839
> Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28754}

TBR=steveanton@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Change-Id: I35576b4afe100a3220c3c01a6a6d5fbdf48a258b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147876
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28755}
2019-08-05 10:46:25 +00:00
Sebastian Jansson
ad5c4accad Adds PeerConnection scenario test framework.
Bug: webrtc:10839
Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28754}
2019-08-05 10:12:43 +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
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
72b7524d87 Adds more stats to CallStatsCollector.
Also adding checks to avoid adding empty stats.

Bug: webrtc:10365
Change-Id: I37ab32a3d4271fcad419f17841a8d2e524d73245
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133020
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27625}
2019-04-15 14:47:56 +00:00
Sebastian Jansson
9a2ca0a9d8 Reland "Adds more performance stats collection to scenario tests."
This is a reland of 63b0b2cf307b47bae3c10b295ece9a5f6d9bd7a4

Original change's description:
> Adds more performance stats collection to scenario tests.
> 
> Bug: webrtc:10365
> Change-Id: I66dce6ff21242c30af674f89fc9fd19172d4a3af
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131948
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27585}

Bug: webrtc:10365
Change-Id: Id7ddb64ac17ecbb4de223dec497bc562040ba7c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132711
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27618}
2019-04-15 12:13:45 +00:00
Artem Titov
80bea1eeaa Revert "Adds more performance stats collection to scenario tests."
This reverts commit 63b0b2cf307b47bae3c10b295ece9a5f6d9bd7a4.

Reason for revert: ScenarioAnalyzerTest.* are broken at HEAD.

Original change's description:
> Adds more performance stats collection to scenario tests.
> 
> Bug: webrtc:10365
> Change-Id: I66dce6ff21242c30af674f89fc9fd19172d4a3af
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131948
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27585}

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

Change-Id: Idb60431dfd859c4328c5c81d0570948463ec3262
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10365
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132709
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27587}
2019-04-12 11:31:07 +00:00
Sebastian Jansson
63b0b2cf30 Adds more performance stats collection to scenario tests.
Bug: webrtc:10365
Change-Id: I66dce6ff21242c30af674f89fc9fd19172d4a3af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131948
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27585}
2019-04-12 10:14:34 +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