Commit Graph

24974 Commits

Author SHA1 Message Date
583d6d9d4f Add missing directory to api/DEPS and PRESUBMIT.py.
TBR: kwiberg@webrtc.org
Bug: webrtc:9887
Change-Id: Ib285005fc2a25549e72922bc38b05a170c6ef228
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/107707
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#25342}
2018-10-24 14:03:04 +00:00
62ae178357 Remove deprecated pipe field from VideoQualityTestFixtureInterface::Params
To be landed after 23th October

Bug: webrtc:9630
Change-Id: I8de460d093438c8b72bca44cdfce49b72cbcc2d0
Reviewed-on: https://webrtc-review.googlesource.com/c/104481
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25341}
2018-10-24 13:21:28 +00:00
825f83b99e Revert "Encode RTC event logs in new format."
This reverts commit ece3c228a2cbd1c1b05eee3a7f55dbb6f020acbc.

Reason: Breaks downstream project.

Bug: webrtc:8111
Change-Id: Ia264802b35a576d74b8a249ed742a8177e5cbe24
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107721
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25340}
2018-10-24 13:18:03 +00:00
e943d43926 Remove deprecated DefaultNetworkSimulationConfig
To be landed after 23th October

Bug: webrtc:9630
Change-Id: Ie322fe5428824b29ad51edaaa446121c5511b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/104600
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25339}
2018-10-24 12:57:31 +00:00
a418e67c53 Use checkdeps to ensure API headers don't include internal headers.
This CL updates the checkdeps configuration for the api/ folder in
order to explicitly avoid to #include non API headers from API headers.

In order to force a careful review of potential exceptions to this
rule, the CL also adds mbonadei@ and kwiberg@ as OWNERS of api/DEPS.

Bug: webrtc:9887
Change-Id: I0ada6f1020186b2782c7d060af36079c452ba1aa
Reviewed-on: https://webrtc-review.googlesource.com/c/106800
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25338}
2018-10-24 12:55:01 +00:00
ec9b77bc42 Remove deprecated API: NetwrokSimulationInterface.
To be landed after 23th October

Bug: webrtc:9630
Change-Id: Ibf9c09d16e86789284491b16812ce57a3cad0624
Reviewed-on: https://webrtc-review.googlesource.com/c/104061
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25337}
2018-10-24 12:52:51 +00:00
257ed437f0 Add support for optional fields in FixedLengthDeltaEncoder
Optional fields are those which only occur sometimes. For example,
the sequence number field in an RTP packet always occurs, but
fields in optional RTP extensions only occur sometimes.

Bug: webrtc:8111
Change-Id: Iff2c35b73530c0a1db68e547b4caf34434aa4ace
Reviewed-on: https://webrtc-review.googlesource.com/c/103362
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25336}
2018-10-24 12:48:44 +00:00
c6ec4b1fa9 Fix w3c URL for RTCIceTransport
No-Try: true
Tbr: hta@webrtc.org
Bug: None
Change-Id: I51a6ad8c30e19cf999c5356e619770cfeee0068f
Reviewed-on: https://webrtc-review.googlesource.com/c/107638
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25335}
2018-10-24 12:36:09 +00:00
5e58bcbf29 Forward audio rtp frequency to Rtcp sender and use it for SR packets
Process video rtp frequency in the same way.

Bug: webrtc:6458
Change-Id: Ia22768e1242d686c2b3e2b911f3e5e492cf8b895
Reviewed-on: https://webrtc-review.googlesource.com/c/107651
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25334}
2018-10-24 12:27:09 +00:00
ece3c228a2 Encode RTC event logs in new format.
This CL adds the encoder and wires it up to the event log.
Parser and unit tests are uploaded in a separate CL.

Bug: webrtc:8111
Change-Id: I6470003e55c2c4006cd8349a2c4bdc3f9491d869
Reviewed-on: https://webrtc-review.googlesource.com/c/106708
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25333}
2018-10-24 12:21:43 +00:00
fb5c1eca30 AEC3: Included missing parsing of config parameter
Bug: webrtc:9912,chromium:898462
Change-Id: I8efb60367964d3880ba15ffd18349abd288d7307
Reviewed-on: https://webrtc-review.googlesource.com/c/107654
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25332}
2018-10-24 11:38:40 +00:00
8e6749e0dd Improve fileutils_override implementation internal API.
Use absl::optional instead of special constant to show, that we failed
to get OutputPath in fileutils_override

Bug: webrtc:9792
Change-Id: Ice19a9bf425e88a747dd9b07e82dbb5bdc59685b
Reviewed-on: https://webrtc-review.googlesource.com/c/107630
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25331}
2018-10-24 10:24:14 +00:00
e068ad6262 Use a sufficiently large bitmask.
The fuzzer uses a bitmask to construct the field trials string.
Now that there's 33 relevant field trials it's no longer large enough, so switch to a 64-bit type.

Bug: chromium:898373
Change-Id: I1ea68d451ceadbd9b720079a577b573866293e4b
Reviewed-on: https://webrtc-review.googlesource.com/c/107650
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25330}
2018-10-24 09:27:18 +00:00
511fe0b2ca Roll chromium_revision 5e5003737d..869181c2dc (602066:602275)
Change log: 5e5003737d..869181c2dc
Full diff: 5e5003737d..869181c2dc

Changed dependencies
* src/base: 1fa2a3d59a..fc75e9da03
* src/build: 97454d191e..4b0fe3afe2
* src/ios: 25ccb1048b..b480083c51
* src/testing: fdf8fc0704..a3a1f924e0
* src/third_party: 49cf955ddc..106ec94e49
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/df56c1dae1..685b5de113
* src/third_party/depot_tools: 03d6d11896..879d5e3796
* src/tools: 6b4b60ca40..2444eb8ba4
DEPS diff: 5e5003737d..869181c2dc/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I2c3cbff203aa19db8ebb136405b7184bcfa643d2
Reviewed-on: https://webrtc-review.googlesource.com/c/107678
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25329}
2018-10-24 08:31:11 +00:00
d38a2b860b Increase the UDP receive buffer for video
Lost packets have been seen in high-bitrate applications and increasing
the UDP receive buffer reduced the problems.

Bug: b/115713113
Change-Id: I671f528afeaea525150fdc2013f2b245778e5d16
Reviewed-on: https://webrtc-review.googlesource.com/c/107580
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25328}
2018-10-24 07:54:12 +00:00
f0c449e3ff APM: Correct includes required for the data dumping functionality
Bug: webrtc:5298
Change-Id: Ia8b8e6a308f1812216651efaf0e2249e9d0cbfd8
Reviewed-on: https://webrtc-review.googlesource.com/c/107631
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25327}
2018-10-24 07:38:28 +00:00
700b4a4e65 AEC3: Allow limiting dominant nearend to the non-initial phase
This CL allows control over the dominant nearend functionality so that
it is not active during the initial phase, when estimates are less
certain.

Bug: webrtc:9906,chromium:898273
Change-Id: I5f61dac806ec3b1ebc1a3ec72f0a16d07a67f14a
Reviewed-on: https://webrtc-review.googlesource.com/c/107632
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25326}
2018-10-24 07:15:49 +00:00
41ed3e083e Roll chromium_revision b1cb85713b..5e5003737d (601125:602066)
Change log: b1cb85713b..5e5003737d
Full diff: b1cb85713b..5e5003737d

Changed dependencies
* src/base: 703032f2cd..1fa2a3d59a
* src/build: dc93a673cf..97454d191e
* src/ios: b3f0c9c0fa..25ccb1048b
* src/testing: a02afa425c..fdf8fc0704
* src/third_party: 1234d52325..49cf955ddc
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a0b50a9a05..df56c1dae1
* src/third_party/depot_tools: 93277a7fc8..03d6d11896
* src/third_party/icu: ccad447212..b029971f1f
* src/third_party/libvpx/source/libvpx: e188b5435d..137d99c91f
* src/tools: c634df121c..6b4b60ca40
* src/tools/swarming_client: 486c9b53c4..f78187ab77
DEPS diff: b1cb85713b..5e5003737d/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,marpan@webrtc.org,
BUG=None

Change-Id: I082f35c474dba26fbd102a1638858c20e4d6955e
Reviewed-on: https://webrtc-review.googlesource.com/c/107665
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25325}
2018-10-23 20:47:55 +00:00
73f3917e89 Add support for signed deltas in FixedLengthDeltaEncoder
Signed deltas can yield a more efficient encoding when the encoded
sequence sometimes moves backwards.

Bug: webrtc:8111
Change-Id: Ib1a50192851214ccc3f2bd7eaf88f4be97e4beb0
Reviewed-on: https://webrtc-review.googlesource.com/c/100423
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25324}
2018-10-23 16:52:31 +00:00
4b31cf571f Disable CertificateTest.CertificateIsUsedInConfig
TBR=magjed@webrtc.org

Bug: webrtc:9763
Change-Id: Id0c3c4b16f300714c637606043c4357682196980
Reviewed-on: https://webrtc-review.googlesource.com/c/107647
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25323}
2018-10-23 16:46:49 +00:00
087e9bed41 AGC2 Limiter class renamed.
Limiter has been renamed to LimiterDbGainCurve, which is a more correct name
and will allow in a follow-up CL to reuse the Limiter name for GainCurveApplier.
This is done to allow to use the limiter without instancing the fixed digital
gain controller and then to fix an AGC2 issue (namely, fixed gain applied after
the adaptive one).

Bug: webrtc:7494
Change-Id: Icd7050e3e51b832bfbf35e5cc61109215c5b1ca6
Reviewed-on: https://webrtc-review.googlesource.com/c/106901
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25322}
2018-10-23 15:20:52 +00:00
4842c78e93 Increasing APM fuzzer coverage.
Add AecDump to the list of fuzzed stuff. Attaches an AecDump to the
Audio Processing Module in the APM-fuzzer. The AecDump writes to
/dev/null.

Bug: webrtc:7820
Change-Id: I03916ce4d1c69906ca8bb7e6fbe29c11e4ea55e5
Reviewed-on: https://webrtc-review.googlesource.com/c/107622
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25321}
2018-10-23 15:11:02 +00:00
c6de47ec8c Added supported H264 profiles for new iPhones
Bug: webrtc:9134, webrtc:7992
Change-Id: Ic5e92764ccd02803e626eb0db21175a13123dc33
Reviewed-on: https://webrtc-review.googlesource.com/c/107625
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25320}
2018-10-23 14:59:13 +00:00
8f726be415 Add ability to override detection of resource location and source root
Bug: webrtc:9792
Change-Id: I944d2e1c1b4b4154a90eba6fbe9c417aad17498d
Reviewed-on: https://webrtc-review.googlesource.com/c/107401
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25319}
2018-10-23 14:21:24 +00:00
877dc89f92 Fix errors in AEC3 JSON parsing
Some issues were surfaced during testing:
 - Config validation clamping silently passed NaNs
 - Config validation only fixed the first out-of-bounds parameter, and
   not any subsequent ones
 - Config validation did not check all values in the config
 - use_stationarity_properties_at_init is misspelled in JSON parsing

These changes are identical to those in this CL:
https://webrtc-review.googlesource.com/c/src/+/107120

Bug: webrtc:9535
Change-Id: I36c5e7c69ffdc2c0c24a9be86ccb1df59683c0fe
Reviewed-on: https://webrtc-review.googlesource.com/c/107640
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25318}
2018-10-23 13:29:07 +00:00
6e8e2993dd Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
This reverts commit 80cd25bcfb2264fa0f1192de942a6f063879dd42.

Reason for revert: Breaks downstream project

Original change's description:
> Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
>
> Bug: None
> Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/107303
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25312}

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

No-Try: true
Bug: None
Change-Id: I77b66bc032e2d95d1bd408c6cdeceb4dcd511699
Reviewed-on: https://webrtc-review.googlesource.com/c/107643
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25317}
2018-10-23 13:21:27 +00:00
7e6b528e5d Removes FakeBaseEngine.
This CL removes FakeBaseEngine and the currently not used functionality
of FakeMediaEngine that depends on it.

Bug: webrtc:9883
Change-Id: I9daa853dedefdf4b4c64b815a7d575eb8ba63c93
Reviewed-on: https://webrtc-review.googlesource.com/c/107581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25316}
2018-10-23 12:36:36 +00:00
362cb50f92 Remove redundant RTC_DCHECK of max/min RTP header extension id
This RTC_DCHECK is no longer valid when we start sending two-byte header
extensions.

Bug: webrtc:7990
Change-Id: Id6cda6f3647838a0d93aa48528bb0bb4300304d3
Reviewed-on: https://webrtc-review.googlesource.com/c/105104
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25315}
2018-10-23 12:33:46 +00:00
93922dca97 Fix flaky unit test in rtc_unittests
Bug: webrtc:9902
Change-Id: I1f6a794a7b473b02764edda486864b6fda94ce39
Reviewed-on: https://webrtc-review.googlesource.com/c/107623
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25314}
2018-10-23 12:23:16 +00:00
848273aa6a Revert "Increase coverage of AEC3 JSON config unit tests, fix bugs"
This reverts commit 8ee06a7b0cc22a486ad924e00034b95dbecd70ce.

Reason for revert: ubsan triggers on config randomization

Original change's description:
> Increase coverage of AEC3 JSON config unit tests, fix bugs
> 
> The new test checks that json strings are unchanged when parsing to a
> config and back to a string. This ensures that everything in the json
> representations is parsed when created a config from the json.
> 
> This CL also adds the render_levels config substruct to the JSON parser.
> 
> Some issues were surfaced by the new test:
>  - Config validation clamping silently passed NaNs
>  - Config validation only fixed the first out-of-bounds parameter, and
>    not any subsequent ones
>  - Config validation did not check all values in the config
> 
> Bug: webrtc:9535
> Change-Id: Ie7b588731dc1fe26ba71d1eb2f177f3b3b8139e3
> Reviewed-on: https://webrtc-review.googlesource.com/c/107120
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25310}

TBR=saza@webrtc.org,peah@webrtc.org

Change-Id: I12d4a6e35110241c51c13eff547ee5a640d141bc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/107624
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25313}
2018-10-23 12:14:08 +00:00
80cd25bcfb Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
Bug: None
Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107303
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25312}
2018-10-23 12:13:02 +00:00
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
8ee06a7b0c Increase coverage of AEC3 JSON config unit tests, fix bugs
The new test checks that json strings are unchanged when parsing to a
config and back to a string. This ensures that everything in the json
representations is parsed when created a config from the json.

This CL also adds the render_levels config substruct to the JSON parser.

Some issues were surfaced by the new test:
 - Config validation clamping silently passed NaNs
 - Config validation only fixed the first out-of-bounds parameter, and
   not any subsequent ones
 - Config validation did not check all values in the config

Bug: webrtc:9535
Change-Id: Ie7b588731dc1fe26ba71d1eb2f177f3b3b8139e3
Reviewed-on: https://webrtc-review.googlesource.com/c/107120
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25310}
2018-10-23 11:08:54 +00:00
f7a7c8a8bc Stop adding RTT delay if there was not packet loss for enough time
Bug: none
Change-Id: I9105c0ee6a4f75381e133a496c3728dece9aca4b
Reviewed-on: https://webrtc-review.googlesource.com/c/23261
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25309}
2018-10-23 10:38:39 +00:00
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
9581bc4c52 Rename too long variable name to extmap_allow_mixed
Bug: webrtc:7990
Change-Id: I990111e473553163cecb9c73fec90d07c24aca02
Reviewed-on: https://webrtc-review.googlesource.com/c/107362
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25307}
2018-10-23 09:25:19 +00:00
2edab4c026 Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.
Bug: webrtc:5876
Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda
Reviewed-on: https://webrtc-review.googlesource.com/c/106820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25306}
2018-10-23 09:24:15 +00:00
01cf44d397 AEC3: Adding missing elements to the json parser
Bug: webrtc:9894
Change-Id: I10c23c5df5b262cfb07fc89715e12bc99fc7be3f
Reviewed-on: https://webrtc-review.googlesource.com/c/107201
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25305}
2018-10-23 09:23:11 +00:00
3583693638 3 TLs: add full stack test for short pattern + base heavy alloc.
A previous CL split an existing field trial in two. This CL
ensures that we have perf coverage for the two interesting
cases: short pattern and short pattern + base heavy TL alloc.

Tested: autoninja -C out/Release; and out/Release/webrtc_perf_tests --gtest_filter="*3TL*"
Bug: chromium:893500
Change-Id: I0585d67860d8a10122793fa1336440c13ebd0c57
Reviewed-on: https://webrtc-review.googlesource.com/c/107561
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25304}
2018-10-23 09:09:21 +00:00
6ed37ba1b5 AEC3: Enable fuzzer testing of old render buffering code.
The old render buffering code has been replaced, but can still be
activated by a killswitch. This change enables fuzzer testing of
the old code path.

Bug: webrtc:9726
Change-Id: I6e91cd4b4a95388cc63d1a65dade21b3c44be71b
Reviewed-on: https://webrtc-review.googlesource.com/c/107562
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25303}
2018-10-23 08:43:00 +00:00
d34597cf3d Update test::CreateVideoStreams to use num_temporal_layers.
MultiCodecReceiveTest/VideoSendStreamTest:
Configure num_temporal_layers via VideoEncoderConfig (and remove
implementations of VideoStreamFactoryInterface used to override
the default num_temporal_layers configuration).

Bug: none
Change-Id: I9855245477fe3c6fe48d1a755d401d6a35a17c70
Reviewed-on: https://webrtc-review.googlesource.com/c/107301
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25302}
2018-10-23 08:36:25 +00:00
98f5f6cdea In RtcpTransceiver functions with callback avoid relying on PostTaskAndReply
deprecated version guarantees (using PostTaskAndReply) callback task will run on the task queue,
and thus doesn't guarantee to run it if task queue is destroyed,

new callback versions instead guarantee callback will always run,
but may run off the task queue if task queue is destroyed.

Both keep guarantee observer callbacks will not run after on_destroyed/on_removed is called.

Bug: None
Change-Id: I61bf52127f3084c0186aa8bc89037bf9296801d8
Reviewed-on: https://webrtc-review.googlesource.com/c/107305
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25301}
2018-10-23 08:29:34 +00:00
b0ab2ce256 Reland "Remove the HighPassFilter interface"
Downstream Chromium dependencies fixed here:
https://chromium-review.googlesource.com/c/chromium/src/+/1286449

This is a reland of e2405c1a823f3baf90a9c72f2e058f91eb659c20

Original change's description:
> Remove the HighPassFilter interface
>
> The functionality remains unaffected.
> Filter toggling is still available via webrtc::AudioProcessing::Config.
> Example:
> webrtc::AudioProcessing::Config config = apm.GetConfig();
> // Read settings
> if (config.high_pass_filter.enabled) { ... }
> // Apply setting
> config.high_pass_filter.enabled = true;
> apm.ApplyConfig();
>
> Bug: webrtc:9535
> Change-Id: Ib4c4b04078bbb490ebdab9721b8c7811d73777a8
> Reviewed-on: https://webrtc-review.googlesource.com/c/102541
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25198}

Bug: webrtc:9535
Change-Id: I0017193ad3ca1762e186f3ad79f29d33ef468202
Reviewed-on: https://webrtc-review.googlesource.com/c/106681
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25300}
2018-10-23 07:44:09 +00:00
c9f9b8711f AEC3: Improve dominant nearend detection
This change makes the dominant nearend detection more accurate.
- The hangover is increased not leave nearend state between words.
- The SNR requirement is increased to not enter nearend state without
  speech activity.
- An early exit mechanism has been added to leave nearend state quickly
  when the echo is strong.

Bug: chromium:897701,webrtc:9897
Change-Id: I9e0f3e6ecb80eee1c0c917d4835f110555f74acf
Reviewed-on: https://webrtc-review.googlesource.com/c/107347
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25299}
2018-10-23 07:05:46 +00:00
ac19414512 Export symbols needed by the Chromium component build (part 6).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: I67a4d016a11deca5ac5459826741dd2d3f7931d5
Reviewed-on: https://webrtc-review.googlesource.com/c/107400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25298}
2018-10-23 06:48:51 +00:00
9a5da497b5 Split out a separate target for SimulcastEncoderAdapter
This will allow downstream projects to use it to construct their own
injected codecs without pulling in dependencies on the software codecs.

Bug: webrtc:7925
Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8
Reviewed-on: https://webrtc-review.googlesource.com/c/107027
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-22 22:36:59 +00:00
165148d84d Reland "Remove deprecated barcode scanning functionality"
This is a reland of ff292f30d9a4b7a56aea872fe488d342f47202a3

I'm leaving empty .py files in place in order to not break downstream client builds.

Original change's description:
> Remove deprecated barcode scanning functionality
>
> This code is not used anymore, but it's not possible to land this CL
> until issue webrtc:9665 is fixed.
>
> Bug: webrtc:9642,webrtc:9665
> Change-Id: Idb68e9bdf51b4239788cd6869dcb44dae87d7c56
> Reviewed-on: https://webrtc-review.googlesource.com/c/95951
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25289}

TBR=phensman@webrtc.org,phoglund@webrtc.org

Bug: webrtc:9642, webrtc:9665
Change-Id: I248f8656b14c89b0b92e777f4408ee6a6dad41f9
Reviewed-on: https://webrtc-review.googlesource.com/c/107360
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25296}
2018-10-22 21:30:58 +00:00
39feabe35c Enables FrameDecryptor to do an initial key request on frame decryption.
This change enables the FrameDecryptor attached to an RtpVideoReceiver to do
an initial request for a KeyFrame if the first successfully decrypted payload
is not a key frame.

Bug: webrtc:9795
Change-Id: I401ce1f513cb51ce520b60dcaf8b825a68d00c7f
Reviewed-on: https://webrtc-review.googlesource.com/c/107246
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25295}
2018-10-22 21:12:35 +00:00
201596f004 Make packet max buffer size configurable via field trial flag
Bug: webrtc:9851
Change-Id: I2d1960e64d5ff11529317088f838032cecb8ae01
Reviewed-on: https://webrtc-review.googlesource.com/c/107346
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25294}
2018-10-22 16:25:10 +00:00
68b2df7cf1 Make protection_overhead_rate configurable through field trial.
Bug: None
Change-Id: I0a4e03d5f8134809c44d2d0ff88c4b4c8b8a9eaa
Reviewed-on: https://webrtc-review.googlesource.com/c/107341
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25293}
2018-10-22 15:44:23 +00:00