c12db81e79
Add frame receive to frame rendered metric to video_quality_analyzer
...
Bug: webrtc:10975
Change-Id: I6b36566efbbb52d27ca6cb44cb3b40aaf0cacb7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29243}
2019-09-19 14:43:04 +00:00
f0be5b5380
Make GetBitstream non-virtual since it is no longer needed for testing.
...
Bug: webrtc:10979
Change-Id: Id313c7fddbec40b9f19dae95f736379b872e3082
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153663
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29242}
2019-09-19 14:04:09 +00:00
40de3cc5ef
Propagating TargetRate struct to BitrateAllocator.
...
Bug: webrtc:9883
Change-Id: I443ac7f1ef0f933e2165fdb2f912d314acc7f2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153345
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29241}
2019-09-19 14:03:04 +00:00
ac315b283c
Add support for max/min encode bitrate to peer connection quality test
...
Bug: webrtc:10975
Change-Id: I9be551040936d2e9b5e41dd1bbaea2ad4afd36ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153481
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29240}
2019-09-19 13:47:29 +00:00
6a092637f0
Delete obsolete isac "assign" api
...
Bug: None
Change-Id: I116e3f4b89e2c1e1f0d06e2ff5d58d2a50e2aadb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153665
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29239}
2019-09-19 13:26:26 +00:00
d8ffbb0bc4
Roll chromium_revision afdb2e7a8b..cf1a2beb4b (697871:697976)
...
Change log: afdb2e7a8b..cf1a2beb4b
Full diff: afdb2e7a8b..cf1a2beb4b
Changed dependencies
* src/build: 050608ea95..76a4ca1e1b
* src/ios: 797d97f55d..8de41b2ec5
* src/testing: c223ceca68..3f13136080
* src/third_party: d7037a0728..90b037fc6f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6a17fd7fb7..033994d4a3
* src/tools: efcee32728..97ef1887ae
DEPS diff: afdb2e7a8b..cf1a2beb4b
/DEPS
No update to Clang.
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I8653cfade8885b6166110d4d72819e30e7d21ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153700
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29238}
2019-09-19 12:32:23 +00:00
76161f7446
Move the call to GetBitstream out of the RtpFrameObject ctor.
...
Bug: webrtc:10979
Change-Id: I9159eb04d4a371e8ed8f932a989d6b884faa7be7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153661
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29237}
2019-09-19 12:08:04 +00:00
14137a1064
Adds logging of audio sessions status on the recording side in ADM for Android.
...
Goal is to be able to retrieve more details about possible microphone conflicts in
cases where Init/Start of audio recording fails.
Only supported on Android N and higher.
Also adds new boolean UMA histogram called WebRTC.Audio.SourceMatchesRecordingSession.
Its value is stored after the recording session has been stopped.
Does not affect the media flow or functionality of the ADM. Time to start audio should
not be affected either since the new check and logging takes place on a separate
ExecutorService thread.
See go/webrtc-adm-android for more details and examples.
Bug: webrtc:10971
Change-Id: Ia80c1534e326907a1582824225d5f58caa016922
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150793
Commit-Queue: Henrik Andreassson <henrika@webrtc.org >
Reviewed-by: Alex Glaznev <glaznev@webrtc.org >
Reviewed-by: Paulina Hensman <phensman@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29236}
2019-09-19 11:35:10 +00:00
86873f0cd3
Improve field trial error message.
...
Bug: None
Change-Id: I112cda6fead3d68136fd7be551686e40191fa87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153482
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29235}
2019-09-19 09:38:49 +00:00
e942b141d8
New build target api:media_interface
...
Bug: webrtc:8733
Change-Id: I84bbefb1a5ef8e592db29b79499d60ac80c23464
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29234}
2019-09-19 09:32:27 +00:00
0a5ed896e2
Adds remote estimates to rtc event log.
...
Bug: webrtc:10742
Change-Id: I0db998a05492603fcdeedca780d9ee3d64aa00d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151651
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29233}
2019-09-19 09:22:37 +00:00
6ed60e39dc
Implement Dependency Descriptor writer
...
Bug: webrtc:10342
Change-Id: I561825265c0990864e1d16aeed4afbdd98871940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153350
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29232}
2019-09-19 08:51:40 +00:00
489843f1b1
Improve trendline estimator logging.
...
Bug: None
Change-Id: I7cc6dc7f45ddb7325252516490436bea1ec8d250
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153521
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29231}
2019-09-19 08:33:11 +00:00
693bf1eea1
Delete modules/rtp_rtcp local DivideRoundToNearest in favor on one in rtc_base
...
To resolve a TODO
Bug: None
Change-Id: I90e10af24718e1aafd7e72076731b34c1110bb4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153524
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29230}
2019-09-19 08:10:12 +00:00
bd24260791
Roll chromium_revision eae7ecf757..afdb2e7a8b (697744:697871)
...
Change log: eae7ecf757..afdb2e7a8b
Full diff: eae7ecf757..afdb2e7a8b
Changed dependencies
* src/base: 01355d2940..3fe4a418db
* src/build: 71ed08eea6..050608ea95
* src/ios: e12f797b06..797d97f55d
* src/testing: 74e29479c8..c223ceca68
* src/third_party: a939fa9d82..d7037a0728
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/cbc960899a..6a17fd7fb7
* src/third_party/depot_tools: 3f79763629..c6be56eced
* src/tools: 8c0c370d66..efcee32728
DEPS diff: eae7ecf757..afdb2e7a8b
/DEPS
No update to Clang.
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: Ia0e447ae02cf1a0a9a1e1a49deca4762b486fd81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153620
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29229}
2019-09-19 02:33:30 +00:00
efa04efaaa
Roll chromium_revision 65274319fc..eae7ecf757 (697640:697744)
...
Change log: 65274319fc..eae7ecf757
Full diff: 65274319fc..eae7ecf757
Changed dependencies
* src/base: 0b98e1163e..01355d2940
* src/build: dd9f9cd163..71ed08eea6
* src/ios: c4c50839b0..e12f797b06
* src/testing: c684ea24f6..74e29479c8
* src/third_party: a9fa316c88..a939fa9d82
* src/third_party/depot_tools: 6f9a0238ce..3f79763629
* src/tools: db6a8ea870..8c0c370d66
DEPS diff: 65274319fc..eae7ecf757
/DEPS
No update to Clang.
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: Ic88b3d2dd39fd73f9f5db6ef22c0aac3d6f40b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153580
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29228}
2019-09-18 20:28:20 +00:00
93b1ea2168
Using struct for bitrate allocation limits.
...
Bug: webrtc:9883
Change-Id: I855c29808ffa14626d78842491fdf81cd00589e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153344
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29227}
2019-09-18 17:03:59 +00:00
1b83a9e400
Only handle each RTCP once.
...
Previously, each RTCP packet was handled several times in a row, once
per m-section. This caused various weirdness and log warning spam, in
particular when using unified plan.
The cause was that the packets were wired trough each BaseChannel
instance up to the Call class. With this fix, the RTCP packets are wired
once per RtpTransportInternal via the common peer connection class.
Bug: chromium:1002875
Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29226}
2019-09-18 16:54:39 +00:00
4bad650ba7
Roll chromium_revision 2bd75c72c1..65274319fc (697505:697640)
...
Change log: 2bd75c72c1..65274319fc
Full diff: 2bd75c72c1..65274319fc
Changed dependencies
* src/base: 107a963c0b..0b98e1163e
* src/build: aae0a7b1db..dd9f9cd163
* src/ios: 809c1d07f6..c4c50839b0
* src/testing: ae66c6e30c..c684ea24f6
* src/third_party: 0e5ec1d6e5..a9fa316c88
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/5ce7022394..a7d9ac2af4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b4e53c4a99..cbc960899a
* src/third_party/icu: faee8bc705..2ecd66c696
* src/tools: ee770715c2..db6a8ea870
DEPS diff: 2bd75c72c1..65274319fc
/DEPS
No update to Clang.
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I5602dbb9d7c7e4e208d5efadf6e764cfbb63c41b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153540
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29225}
2019-09-18 16:37:39 +00:00
7b04a91f4a
Delete almost all default methods on PeerConnectionInterface
...
Keeping default implementations only for methods involved in
ongoing transitions.
Intended to catch inconsistencies between the interface and the
PeerConnectionProxy class, at compile time.
Bug: webrtc:10716
Change-Id: I4cb126c353855f7288ba09273fa6f87aaa0f32eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140860
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29224}
2019-09-18 16:27:44 +00:00
e607a06338
Removed unused include from PacketBuffer.
...
Bug: none
Change-Id: I502f634e85421e38a02cd31d8ae5446cbe32d138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153520
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29223}
2019-09-18 15:35:59 +00:00
33b83fdc95
Introduce integer division helpers with non-default rounding
...
There are multiple places in webrtc code where alternative than
default rounding is desired. Typically this rounding is inlined.
e.g. as (<x> + <y>/2) / <y> making code more clumpsy (<y> might be long expression)
and unsafe for large values of <x>
This change introduce small helpers to address both concerns.
Bug: None
Change-Id: Icd8dcee80a697b7c50ba0b2e50295087d2be8670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153354
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29222}
2019-09-18 15:08:39 +00:00
b6a45dda4c
Revert "Fix minor regression caused by a8336d3"
...
This reverts commit 809198edfff416fce8d75b574a43afab5e67b1cd.
Reason for revert: Performance regressions that need to be addressed.
Original change's description:
> Fix minor regression caused by a8336d3
>
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
>
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org >
> Commit-Queue: Evan Shrubsole <eshr@google.com >
> Cr-Commit-Position: refs/heads/master@{#29208}
TBR=sprang@webrtc.org ,eshr@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10126
Change-Id: I133cbe5d8cb894ed944ae8a2d0f63a78bbed72ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153484
Commit-Queue: Erik Språng <sprang@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29221}
2019-09-18 14:38:15 +00:00
53227ccba9
Remove webrtc::MinPositive from api/.
...
Follow-up of https://webrtc-review.googlesource.com/c/src/+/153220 ,
where during code review it was suggested to move webrtc::MinPositive
out of the api/ directory.
Bug: None
Change-Id: I0c3b87a9ffd1cd205a85dddd9f44cfd95eb02206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153480
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29220}
2019-09-18 12:52:09 +00:00
1162ba285d
Add max/min encode bitrates to video config of peer connection tests
...
Extend PeerConnectionE2EQualityTestFixture::VideoConfig with
min_encode_bitrate_bps and max_encode_bitrate_bps.
These are needed to be able to specify the bitrate to be used in tests.
Bug: None
Change-Id: I8af88020e9b364d924e2cecb2bdcc12bf287394d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153352
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29219}
2019-09-18 09:15:03 +00:00
7cfde54849
Roll chromium_revision 51a0808947..2bd75c72c1 (697405:697505)
...
Change log: 51a0808947..2bd75c72c1
Full diff: 51a0808947..2bd75c72c1
Changed dependencies
* src/base: 8b3a663d3b..107a963c0b
* src/ios: 2b489af222..809c1d07f6
* src/third_party: 04016a4f18..0e5ec1d6e5
* src/tools: 5ed44acf45..ee770715c2
DEPS diff: 51a0808947..2bd75c72c1
/DEPS
No update to Clang.
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I524f06254bc0287f538f92aa34a48107c0c7a24b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153463
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29218}
2019-09-18 08:42:34 +00:00
738bfa7bab
Remove api/bitrate_constraints.h.
...
Bug: webrtc:8733
Change-Id: Iaeb26e07d399f25dc18b0c4af38ed400577a5d3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29217}
2019-09-18 06:37:58 +00:00
c128df14ee
Update style guide for absl::make_unique.
...
No-Try: True
Bug: webrtc:10945
Change-Id: I707aefda5d5b224d78b97ce3122e095c7b9b1f1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153356
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29216}
2019-09-18 06:10:58 +00:00
95c4b916ce
Roll chromium_revision 31d9542abc..51a0808947 (697288:697405)
...
Change log: 31d9542abc..51a0808947
Full diff: 31d9542abc..51a0808947
Changed dependencies
* src/base: 9d1bb9a333..8b3a663d3b
* src/build: 48ea8b8e18..aae0a7b1db
* src/ios: fb0b52197c..2b489af222
* src/testing: 24e33d5203..ae66c6e30c
* src/third_party: cf41eae8a8..04016a4f18
* src/third_party/depot_tools: 2c210a4908..6f9a0238ce
* src/third_party/freetype/src: 99f23d6ff2..04ebb2a000
* src/tools: f7c3756749..5ed44acf45
DEPS diff: 31d9542abc..51a0808947
/DEPS
No update to Clang.
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: Ibb815b17d9359814b3a4b364b58e4f61c08f0bd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153440
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29215}
2019-09-18 00:37:04 +00:00
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
4d461ba298
Reusing MediaStreamAllocationConfig struct in ObserverConfig.
...
This makes it easier to follow the code and reduces the risk of
accidents in the mapping of fields.
Also renaming the ObserverConfig struct to AllocatableTrack to better
reflect what it represents.
Bug: webrtc:9883
Change-Id: Ia320363813db2b4bf7b37852882a1ccb7644ae0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153342
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29213}
2019-09-17 19:31:25 +00:00
86314cfb5d
Cleaning up C++14 move into lambda TODOs.
...
Bug: webrtc:10945
Change-Id: I4d2f358b0e33b37e4b4f7bfcf3f6cd55e8d46bf9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153240
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29212}
2019-09-17 19:18:26 +00:00
368d002e48
Roll chromium_revision dbd1569418..31d9542abc (697157:697288)
...
Change log: dbd1569418..31d9542abc
Full diff: dbd1569418..31d9542abc
Changed dependencies
* src/base: c4f644b627..9d1bb9a333
* src/build: 3bf1aad87c..48ea8b8e18
* src/ios: e5e3e08174..fb0b52197c
* src/testing: 79f8c8e672..24e33d5203
* src/third_party: d482089f63..cf41eae8a8
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b19a360c12..b4e53c4a99
* src/third_party/freetype/src: cc17f852d5..99f23d6ff2
* src/tools: da9f4cfafe..f7c3756749
DEPS diff: dbd1569418..31d9542abc
/DEPS
No update to Clang.
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I00231a7c0097659fd6bed17b00299957f80d8715
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153403
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29211}
2019-09-17 18:36:01 +00:00
9fa8ef1f4f
absl::make_unique presubmit check.
...
Starting from [1], WebRTC has been migrated to std::make_unique, in
order to keep the codebase consistent, absl::make_unique is now
banned.
Output example:
** Presubmit ERRORS **
Please use std::make_unique instead of absl::make_unique.
Affected files:
call/rtp_demuxer.cc
[1] - https://webrtc-review.googlesource.com/c/src/+/153221
Bug: webrtc:10945
Change-Id: I5b727ecc5ea8ac2ecd89cbd5fba866baf6de9012
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153355
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29210}
2019-09-17 17:47:31 +00:00
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
809198edff
Fix minor regression caused by a8336d3
...
VideoEncoder::SetRates was being called unnessesarily when the fields
appended to RateControlParameters were changed. Since SetRates only
cares about RateControlParameters, it should have only been called if
the RateControlParameters themselves were actually changed.
Bug: webrtc:10126
Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Evan Shrubsole <eshr@google.com >
Cr-Commit-Position: refs/heads/master@{#29208}
2019-09-17 13:34:18 +00:00
7d00342f66
Remove old packet socket factory header.
...
Bug: webrtc:7447
Change-Id: I367e624070561349a2e98c00d1ce97ad8d12edeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153347
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Patrik Höglund <phoglund@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29207}
2019-09-17 11:21:45 +00:00
e1b777717b
Removing deprecated min_pacing_rate alias in StreamsConfig.
...
Bug: webrtc:9883
Change-Id: I8ca9f51b60b5fc24233f14404c13b411a5f2c253
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153343
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29206}
2019-09-17 11:20:34 +00:00
4a822f4b3c
Roll chromium_revision 2e4ccff8a8..dbd1569418 (696956:697157)
...
Change log: 2e4ccff8a8..dbd1569418
Full diff: 2e4ccff8a8..dbd1569418
Changed dependencies
* src/base: 6815595428..c4f644b627
* src/build: f44258b883..3bf1aad87c
* src/ios: 7178c1b623..e5e3e08174
* src/testing: e4bf8aa501..79f8c8e672
* src/third_party: ff1e5ce5a6..d482089f63
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d835968072..b19a360c12
* src/third_party/depot_tools: 4a60db4c3e..2c210a4908
* src/tools: eb67b7ca40..da9f4cfafe
DEPS diff: 2e4ccff8a8..dbd1569418
/DEPS
No update to Clang.
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: I81acf7232d9e43c92f6bafbdae686ff7dc4cc6ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153380
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29205}
2019-09-17 10:35:11 +00:00
2c6ea52369
In TaskQueueTest::PostDelayedAfterDesctruct increase timeout
...
from 2x expected time to 10x.
To decrease flakiness for task queue implemntations that destroy tasks
after destruction of the task queue.
Bug: chromium:1000531
Change-Id: Ieb37ff782ead585e0aa2c84472e3993107c5c072
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152830
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29204}
2019-09-17 09:36:47 +00:00
c1c6284cd1
New (empty) build target api:media_stream_interface
...
Will be populated in a later cl.
Bug: webrtc:8733
Change-Id: I7e136645380d2264697c72f2d49403b3b9f0f044
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153341
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29203}
2019-09-17 08:59:15 +00:00
172218295b
Roll chromium_revision 3cf04dec00..2e4ccff8a8 (696812:696956)
...
Change log: 3cf04dec00..2e4ccff8a8
Full diff: 3cf04dec00..2e4ccff8a8
Changed dependencies
* src/base: 03bceb0723..6815595428
* src/build: 36c63090ad..f44258b883
* src/ios: 06351f1c5f..7178c1b623
* src/testing: 13f4bfd01e..e4bf8aa501
* src/third_party: a624039fb7..ff1e5ce5a6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/914862e8ec..d835968072
* src/third_party/depot_tools: 73ec83f0fe..4a60db4c3e
* src/third_party/googletest/src: cad3bc46c2..f2fb48c3b3
* src/tools: 1133315201..eb67b7ca40
DEPS diff: 3cf04dec00..2e4ccff8a8
/DEPS
No update to Clang.
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: Ic1c98d90d4490c149594c3cfb716acf582208840
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153280
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29202}
2019-09-16 22:34:28 +00:00
7262fc29a0
Refactor Rtp Receivers to accept SSRC 0.
...
Changes Rtp Receivers to use a null value of ssrc to mean a default
receive stream.
Bug: webrtc:8694
Change-Id: I835199345f7add993b9078c8b0e7988d5cdd6646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152425
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Commit-Queue: Saurav Das <dinosaurav@chromium.org >
Cr-Commit-Position: refs/heads/master@{#29201}
2019-09-16 21:29:58 +00:00
3d1647412c
in RtcpTransciever use lambdas with move capture.
...
Now that c++14 allows that.
Bug: webrtc:10945
Change-Id: I218bebeb549b66c9ad3760762f2783c76d30143d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153200
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29200}
2019-09-16 17:30:48 +00:00
3462793296
Roll chromium_revision 1d12ff693d..3cf04dec00 (696696:696812)
...
Change log: 1d12ff693d..3cf04dec00
Full diff: 1d12ff693d..3cf04dec00
Changed dependencies
* src/base: 4e24f6c092..03bceb0723
* src/build: e7f81b6504..36c63090ad
* src/ios: d8a0bae322..06351f1c5f
* src/testing: 15e0bc2f47..13f4bfd01e
* src/third_party: 3355b26c6e..a624039fb7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ecd10922ee..914862e8ec
* src/third_party/freetype/src: 3de1b8d0b0..cc17f852d5
* src/tools: 3692d5fe84..1133315201
DEPS diff: 1d12ff693d..3cf04dec00
/DEPS
Clang version changed 8455294f2ac13d587b13d728038a9bffa7185f2b:b4160cb94c54f0b31d0ce14694950dac7b6cd83f
Details: 1d12ff693d..3cf04dec00
/tools/clang/scripts/update.py
TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=None
Change-Id: Ia39d25cc4bb0666ae08bb94763f79e07de2849e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153030
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com >
Cr-Commit-Position: refs/heads/master@{#29199}
2019-09-16 16:34:50 +00:00
68ef259c30
Delete deprecated rtc_event.h file
...
Bug: webrtc:10206
Change-Id: I6fe19bfb0b6dbef5ce73711b22fd903432f87810
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152485
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29198}
2019-09-16 15:59:54 +00:00
f5dec1c6af
Implement Dependency Descriptor reader
...
Bug: webrtc:10342
Change-Id: I671bf57368016b633546966cc994646095433519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152823
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29197}
2019-09-16 15:55:54 +00:00
d9cc8c08dc
Encoder switching based on network and/or resolution conditions.
...
In this CL:
- Renamed EncoderFailureCallback to EncoderSwitchRequestCallback. An encoder
switch request can now also be made with a configuration that specifies which
codec/implementation to switch to.
- Added "WebRTC-NetworkCondition-EncoderSwitch" field trial that specifies
switching conditions and desired codec to switch to.
- Added checks to trigger the switch based on these conditions.
Bug: webrtc:10795
Change-Id: I9d3a9a39a7c4827915a40bdceed10b581d70b90a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151900
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29196}
2019-09-16 13:43:29 +00:00
73ceed58f8
Update simulcast bitrate calculations for non-standard resolutions.
...
* Increase 540p bitrate to 1.2mbps from 0.9mpbs.
960x540 bitrate was by far smallest in terms of bits per pixel. This change
brings it closer to other resolutions.
* Interpolate max/target/min bitrates for non-standard resolutions based
on number of pixels.
Bug: webrtc:10965
Change-Id: If0aa56bb4c614ca09ee39d3a2b700aab2ffa1a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152828
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29195}
2019-09-16 13:40:59 +00:00
1b6a30ddcc
Update WebRTC's C++ style guide to reflect the switch to C++14.
...
No-Try: True
Bug: webrtc:10945
Change-Id: Ife5d5c12144e00aeefd5ccfe8470c8741ad8eb54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151460
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29194}
2019-09-16 11:45:35 +00:00