Commit Graph

29278 Commits

Author SHA1 Message Date
7350a90237 RNN VAD: prepare for SIMD optimization
This CL adds the boilerplate for SIMD optimization of FC and GRU layers
in rnn.cc. The same scheme of AEC3 has been used. Unit tests for the
optimized architectures have been added (the same unoptimized
implementation will run).

Minor changes:
- unnecessary const removed in rnn.h
- FC and GRU test data in the anon namespace as constexpr

Bug: webrtc:10480
Change-Id: Ifae4e970326e7e7c603d49aeaf61194b5efdabd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141419
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29696}
2019-11-05 18:34:15 +00:00
ad04327df8 Add equals and hashCode method for IceCandidate class.
Bug: webrtc:11072
Change-Id: I03568c3290a49466d0f459b1de8c89afaaf020ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158860
Commit-Queue: Honghai Zhang <honghaiz@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29695}
2019-11-05 18:04:55 +00:00
be43b7c4d7 Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
This reverts commit 898ae5d74b3a7261c6993315bd2aa9e59cc493c0.

Reason for revert: This CL was just needed in order to
have a WebRTC commit to pin in Chromium to test the
component build (this CL enables symbol exports).

Original change's description:
> Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
> 
> This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed
> 
> I will revert this reland as soon as it lands because I just need
> to have a WebRTC commit to pin in Chromium in order to test the
> component build (this CL enables symbol exports).
> 
> Original change's description:
> > Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
> >
> > In order to land the component build support in Chromium, it is
> > easier to turn on symbols export every time that is_component_build=true
> > instead of setting rtc_enable_symbol_export=is_component_build in
> > Chromium (since is_component_build is not available in .gn).
> >
> > rtc_enable_symbol_export is still kept in the mix in order to turn
> > on symbol exports in any case a shared library will be added to the
> > WebRTC build.
> >
> > Bug: webrtc:9419
> > Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29540}
> 
> TBR: kwiberg@webrtc.org
> No-Try: True
> Bug: webrtc:9419
> Change-Id: I1b929a5a702ca8010c557612004f538256be8a4b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158889
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29693}

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

Change-Id: I39fe67e388857721f239b0042a33ef8ef90f2036
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29694}
2019-11-05 15:58:06 +00:00
898ae5d74b Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed

I will revert this reland as soon as it lands because I just need
to have a WebRTC commit to pin in Chromium in order to test the
component build (this CL enables symbol exports).

Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}

TBR: kwiberg@webrtc.org
No-Try: True
Bug: webrtc:9419
Change-Id: I1b929a5a702ca8010c557612004f538256be8a4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158889
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29693}
2019-11-05 15:55:54 +00:00
5ea5749a86 AEC3: Multichannel suppressor
This change adds multichannel support to the AEC3 suppressor.
Processing of mono capture is bit-exact to the previous code.

Bug: webrtc:10913
Change-Id: I89affe3e066021bc34e4b525edf44dd3bea68365
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158882
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29692}
2019-11-05 15:51:39 +00:00
3ee47de99b Roll chromium_revision 7ce0264138..4186f99f63 (710014:712562)
Change log: 7ce0264138..4186f99f63
Full diff: 7ce0264138..4186f99f63

Changed dependencies
* src/base: cccddae48a..dc5c15b4a7
* src/build: c0d6bd0031..9bc5ae11c3
* src/ios: 23c87f0723..c5b06b8f44
* src/testing: 09a3f2a9c4..6c3f7807fa
* src/third_party: 82bf503214..8b94058a0f
* src/third_party/bazel: 1794576f65a721eb0af320a0701e48d31f1b2415..tQPvsIj1Gtw5iXssKy7OREE-S02u7zItrw42l3DHUroC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/76918d0164..6be491b7bb
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d039ea0c17..a05f421623
* src/third_party/depot_tools: 27eb01c355..f6a2232b48
* src/third_party/freetype/src: 0a3d2bb99b..b75031a26e
* src/third_party/icu: b51014b962..88ea42af73
* src/third_party/libvpx/source/libvpx: 412547ad4b..9b73e21c0d
* src/third_party/r8: IOR6mtzOa3X07B0hIZ5U2prEf0GbTvCdN8no1FjNAtQC..7iz_2pdTN2RZRzgoVnxCi1Ro0iUSsEsvXGgmBgXG6z4C
* src/tools: 2d5d164a8d..5062d71604
DEPS diff: 7ce0264138..4186f99f63/DEPS

No update to Clang.

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

Change-Id: Ibb8f638a1df534ebf17750fdfb9d252748f9daf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158901
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@{#29691}
2019-11-05 15:50:34 +00:00
4e5c709ed4 Reland "Correct AEC3 multichannel functionality activation"
This is a reland of 9dda1b3a484ebeef921e419406402039f3852427

Original change's description:
> Correct AEC3 multichannel functionality activation
> 
> This CL corrects the AEC3 multichannel activation
> to also work for the case when a factory is used
> for the activation.
> 
> Bug: webrtc:10913
> Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29676}

Bug: webrtc:10913
Change-Id: I1cb3d0de61ea0b299158ca85433f2442c65c196f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158886
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29690}
2019-11-05 14:38:49 +00:00
4c04d8e10c Fixing autoroller/roll_deps.py.
This fixes the current autoroller breakage [1] by pre-updating the
cipd package version for 'chromium/third_party/jdk'.

Error was:

File "/b/s/w/ir/cache/builder/src/tools_webrtc/autoroller/roll_deps.py", line 264, in _FindChangedCipdPackages
    assert pkgs_equal, 'Old: %s\n New: %s' % (old_pkgs, new_pkgs)
AssertionError: Old: [{'version': 'rfJtuH296mzs7BYOgmQkpz-7ydXtpLKeO15qDDMaa5cC', 'package': 'chromium/third_party/jdk'}]
 New: [{'version': 'PfRSnxe8Od6WU4zBXomq-zsgcJgWmm3z4gMQNB-r2QcC', 'package': 'chromium/third_party/jdk'}, {'version': 'fkhuOQ3r-zKtWEdKplpo6k0vKkjl-LY_rJTmtzFCQN4C', 'package': 'chromium/third_party/jdk/extras'}]

[1] - https://ci.chromium.org/p/webrtc/builders/cron/Auto-roll%20-%20WebRTC%20DEPS/7897

Bug: None
Change-Id: I473410000186843221b941d3f8d62b4bb8bd5cac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158884
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29689}
2019-11-05 12:15:49 +00:00
b210eeb812 Reland "Refactors BitrateProber with unit types and absolute probe time."
This is a reland of 739a5b3692880cb6b41ae620fb9e755c39b044b1

Patchset 1 is the original CL, patchset 3 includes a fix

Original change's description:
> Refactors BitrateProber with unit types and absolute probe time.
>
> Using unit types improves readability and some conversion in PacedSender
> can be removed.
>
> TimeUntilNextProbe() is replaced by NextProbeTime(), so returning an
> absolute time rather than a delta. This fits better with the upcoming
> TaskQueue based pacer, and is also what is already stored internally
> in BitrateProber.
>
> Bug: webrtc:10809
> Change-Id: I5a4e289d2b53e99d3c0a2f4b36a966dba759d5cf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158743
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29670}

Bug: webrtc:10809
Change-Id: I033193c78474fdd82c109fdab0a8f09a05f7b30e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158841
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29688}
2019-11-05 11:02:22 +00:00
1983458981 Revert "Reland "Correct AEC3 multichannel functionality activation""
This reverts commit d5a7838926b839469db1072d72a92e6814f2faeb.

Reason for revert: Causing errors in downstream tests.

Original change's description:
> Reland "Correct AEC3 multichannel functionality activation"
> 
> This is a reland of 9dda1b3a484ebeef921e419406402039f3852427
> 
> Original change's description:
> > Correct AEC3 multichannel functionality activation
> > 
> > This CL corrects the AEC3 multichannel activation
> > to also work for the case when a factory is used
> > for the activation.
> > 
> > Bug: webrtc:10913
> > Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29676}
> 
> Bug: webrtc:10913
> Change-Id: Ibfe4e8a51183390a4054514bb294c89c2ea201e9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158880
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29685}

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

Change-Id: I6e27bc7fd1c9d4d5550fdc6ae14c39ca84fb03f8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158883
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29687}
2019-11-05 09:54:06 +00:00
3ac6375bb3 Add 3 missing RTC_EXPORT.
These two annotations are now needed to correctly compile Chromium
with is_component_build=true and the WebRTC component.

TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: Id5603cf747357c0c2a4b41684eb4fd607cccfdea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158881
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@{#29686}
2019-11-05 09:40:03 +00:00
d5a7838926 Reland "Correct AEC3 multichannel functionality activation"
This is a reland of 9dda1b3a484ebeef921e419406402039f3852427

Original change's description:
> Correct AEC3 multichannel functionality activation
> 
> This CL corrects the AEC3 multichannel activation
> to also work for the case when a factory is used
> for the activation.
> 
> Bug: webrtc:10913
> Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29676}

Bug: webrtc:10913
Change-Id: Ibfe4e8a51183390a4054514bb294c89c2ea201e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158880
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29685}
2019-11-05 09:11:23 +00:00
ca585bb457 Make some DecisionLogic functions virtual.
Bug: webrtc:11005
Change-Id: I86d1eadc85162abf77010d97917e5ab20f644d66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158783
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29684}
2019-11-04 16:29:17 +00:00
054f18513e Use template instantiation declaration/definition for RTCStatsMember<T>.
This CL works around an "Explicit specialization after instantiation
error" when building with clang-cl and is_component_build=true (see
crbug.com/1018579). On top of that it uses "template instantiation
declarations/declarations" in order to avoid to instantiate the
template in clients code.

TBR: hbos@webrtc.org
Bug: webrtc:9419, chromium:1018579
Change-Id: I1b2862de678586afc81e8f7a407947322f8a06c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158795
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29683}
2019-11-04 16:24:37 +00:00
bb56d4b0e2 Revert "Refactors BitrateProber with unit types and absolute probe time."
This reverts commit 739a5b3692880cb6b41ae620fb9e755c39b044b1.

Reason for revert: Speculate revert due to perf alerts.

Original change's description:
> Refactors BitrateProber with unit types and absolute probe time.
> 
> Using unit types improves readability and some conversion in PacedSender
> can be removed.
> 
> TimeUntilNextProbe() is replaced by NextProbeTime(), so returning an
> absolute time rather than a delta. This fits better with the upcoming
> TaskQueue based pacer, and is also what is already stored internally
> in BitrateProber.
> 
> Bug: webrtc:10809
> Change-Id: I5a4e289d2b53e99d3c0a2f4b36a966dba759d5cf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158743
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29670}

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

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10809
Change-Id: Ic0ad7d45031bf33c24583dfde308bdd8087a62aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158799
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29682}
2019-11-04 14:37:03 +00:00
ebf4552c8f Adds WebRTC-Audio-AgcMinMicLevelExperiment to AGC1
Bug: webrtc:11065
Change-Id: Id07ebab7bfa12980187a5847d4f11c8a57450147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158784
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29681}
2019-11-04 14:10:03 +00:00
2b9317ad76 Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame.
- Centralize field trial string reading to RateControlSettings
- Cache RateControlSettings at all production code use sites

Bug: None
Change-Id: I0dbce9cc97fea0bc780982e7ef270b417a8c15bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158664
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29680}
2019-11-04 13:50:59 +00:00
a06048a41e Return status instead of CHECKing in event log parser.
This CL adds ParseStatus/ParseStatusOr classes and returns those instead
of CHECKing that the log is well formed. Some refactoring was required.

We also add a allow_incomplete_logs parameter to the parser which by
default is false. Setting it to true will make the parser log a warning
but return success for errors that typically indicate that the log has
been truncated. "Deeper" errors indicating log corruption still return
an error.

Bug: webrtc:11064
Change-Id: Id5bd6e321de07e250662ae3aaa5ef15f48db6d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158746
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29679}
2019-11-04 12:42:57 +00:00
cc9bf6398c Revert "Correct AEC3 multichannel functionality activation"
This reverts commit 9dda1b3a484ebeef921e419406402039f3852427.

Reason for revert: The CL is causing downstream issues

Original change's description:
> Correct AEC3 multichannel functionality activation
> 
> This CL corrects the AEC3 multichannel activation
> to also work for the case when a factory is used
> for the activation.
> 
> Bug: webrtc:10913
> Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29676}

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

Change-Id: Ic487f77f5c11485a0f25a2a1d3797d0ec956f913
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158797
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29678}
2019-11-04 08:49:30 +00:00
191e38fb47 Delete gturn support
Delete enum RelayType and classes RelayPort and RelayServer.

See also PSA: https://groups.google.com/forum/?#!msg/discuss-webrtc/0ROpUXpw3Gs/eikIN-eEBwAJROpUXpw3Gs/eikIN-eEBwAJ

Bug: webrtc:10998
Change-Id: I1eab760dc73df9156cd1224cf99ad4a4c12ed882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154522
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29677}
2019-11-04 08:31:07 +00:00
9dda1b3a48 Correct AEC3 multichannel functionality activation
This CL corrects the AEC3 multichannel activation
to also work for the case when a factory is used
for the activation.

Bug: webrtc:10913
Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29676}
2019-11-04 08:23:27 +00:00
8846c8af85 RNN VAD: cast and scale quantized weights at init
This CL has two goals: (i) avoid casting and scaling of the NN weights
for every processed feature vector and (ii) prepare for SIMD
optimizations.

Bug: webrtc:10480
Change-Id: Ice7bac5657123354714cc7c63b00abbb8a76c7d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141413
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Fredrik Hernqvist <fhernqvist@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29675}
2019-11-01 16:44:59 +00:00
26452ff7db Cleanup of TransportFeedbackAdapter.
* Removes legacy defines from rtp_rtcp_defines.
* Simplifies the feedback adaptation logic, this is achieved
  by using the ability to preserve lost packets information
  from the RTCP message.
* Extracts in flight data tracking to a separate helper class.
* Removes legacy fields and constructors from the PacketFeedback
  structure.
* Removes the legacy GetTransportFeedbackVector method.

Apart from reducing total LOC, this prepares for moving the adaptation
to run on a TaskQueue.

Bug: webrtc:9883
Change-Id: I5ef4eace0948f119f283cd71dc2b8d0954a1449b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29674}
2019-11-01 11:55:16 +00:00
c3d1f9b0cd Enable injection of a custom NetEqFactory into PeerConnectionFactory.
Injecting both a custom NetEqFactory and an AudioDecoderFactory is not
supported, in that case the AudioDecoderFactory should be wrapped inside
the NetEqFactory.

Bug: webrtc:11005
Change-Id: I4e311eb1bfa03c91bca587d70540e81829f881c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158720
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29673}
2019-11-01 11:30:36 +00:00
2ebbff83ee do not offer gcm as the preferred cipher suite
Move the GCM srtp cipher suites below the default SRTP_AES128_CM_SHA1_80 one.
This will not negotiate them by default since they have an impact on packet overhead for audio-only calls.
GCM can still be negotiated if the peer offers it as preferred cipher suite or answers with just that cipher suite.

BUG=chromium:713701

Change-Id: I79bd4ab827e5c7f55f5550d14db3f4217a7eff86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158404
Reviewed-by: Justin Uberti <juberti@google.com>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Justin Uberti <juberti@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29672}
2019-10-31 20:59:42 +00:00
3ce44a3540 Move NetEq headers to api/
This CL also introduces NetEqFactory and NetEqControllerFactory
interfaces, as well as several convenience classes for working with
them: DefaultNetEqFactory, DefaultNetEqControllerFactory and
CustomNetEqFactory.

Bug: webrtc:11005
Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 15:43:59 +00:00
739a5b3692 Refactors BitrateProber with unit types and absolute probe time.
Using unit types improves readability and some conversion in PacedSender
can be removed.

TimeUntilNextProbe() is replaced by NextProbeTime(), so returning an
absolute time rather than a delta. This fits better with the upcoming
TaskQueue based pacer, and is also what is already stored internally
in BitrateProber.

Bug: webrtc:10809
Change-Id: I5a4e289d2b53e99d3c0a2f4b36a966dba759d5cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158743
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29670}
2019-10-31 15:34:39 +00:00
8d65e9ab98 Fixes pacing interval dependency and race in BandwidthEndToEndTest
BandwidthEndToEndTest failed when I tested it with the new task-queue
based paced sender. This turned out to be issues with this test.
Problems fixed by this CL:

1. Send-side BWE not set up correctly. Caused probing to fail.
2. Test waited for non-zero pacer delay, but the new pacer will not
   generate any delay in this scenario.
3. Race condition during shutdown of test.

1) Is just a matter of configiuring the right header extension.
2) Set up test with high encoder bitrate to trigger pacer delay.
3) TaskQueue outlives the Call instances used in test, so make sure
   they are not referenced from lambda during teardown.

Bug: webrtc:10809
Change-Id: I6393975691dfa05eb5b25d9283e476062e23a876
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158722
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29669}
2019-10-31 15:27:21 +00:00
caaa9e73d7 AEC3: Handle multichannel audio in single CNG instance
Instead of having a comfort noise generator (CNG) instance per capture
channel, one instance handles CNG for all capture channels.

Bug: webrtc:10913
Change-Id: I897471be6d203ad750c517c5076d421f2ae3879b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158780
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29668}
2019-10-31 14:51:35 +00:00
cd2a92f8e0 Removes RPLR based FEC controller.
This is not used and adds a lot of maintenance overhead to
the code since it requires that the transport feedback adapter
communicates directly with audio send stream.

This also means that the packet loss tracker used as input for
this can be removed and a lot of wiring up code overall.

Bug: webrtc:9883
Change-Id: I25689fb622ed89cbb378c27212a159485f5f53be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156502
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29667}
2019-10-31 13:56:44 +00:00
d1ea4c93d3 Update comments on Audio Level RTP header extension.
Bug: None
Change-Id: Id9f10ea2236ba4a154cd82f2e2b05e3fa03442f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158745
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29666}
2019-10-31 13:11:41 +00:00
577c580cd0 Do not stop SingleThreadedTaskQueueForTestingTest near the end of the tests
That brings usage of that queue closer to the production.
In particular that should surface race conditions on destruction.
Those should be fixed rather than avoided.

Bug: webrtc:10933
Change-Id: Iff60cf5a4b87bd848117ef543ffc97f6504dc979
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157898
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29665}
2019-10-31 13:07:30 +00:00
fb075d558d Removing unused Opus wrapper API: WebRTCOpus_DecodePlc.
Bug: None
Change-Id: I5b613b4c13ec5f6ad13d8430043d006f6d83c11f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158671
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29664}
2019-10-31 12:01:31 +00:00
0cbb58e046 Reland "Refactoring of the noise suppressor and adding true multichannel support"
This is a reland of 87a7b82520b83a6cf42da27cdc46142c2eb6248c

Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
> 
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
> 
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
> 
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
> 
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}

Bug: webrtc:10895, b/143344262
Change-Id: I236f1e67bb0baa4e30908a4cf7a8a7bb55fbced3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158747
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29663}
2019-10-31 11:56:01 +00:00
159b417c98 Keep the video send stream alive if the encoder drop frames.
The encoder can drop all frames for extended periods if it has produced over budget. After 2 seconds without any encoded frames, the video send stream times out and deallocates the stream.

Ideally the send stream should keep track if frames are captured instead of encoded, but keeping the stream alive using OnDroppedFrame can work as a proxy for that.

Bug: webrtc:11062
Change-Id: Id7ec1ff333427643453c4a36d1db03ca826cd9ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29662}
2019-10-31 11:30:47 +00:00
c35333d1fd Add RTC_EXPORT_TEMPLATE_{DECLARE,DEFINE} macros.
This CL adds macros to correctly export template instantiation
declarations and definitions. These macros have been borrowed from
Chromium's //base/export_template.h [1] and are supposed to be used
together with RTC_EXPORT [2].

The goal is to start using explicit template instatiation declarations
(introduced in C++11) [3] and remove workarounds that are not compatible
with all the compilers. An example can be found in [4], where in order
to workaround crbug.com/1018579, another workaround was almost created
before being stopped at code review time.

[1] - https://cs.chromium.org/chromium/src/base/export_template.h
[2] - https://cs.chromium.org/chromium/src/third_party/webrtc/rtc_base/system/rtc_export.h
[3] - https://en.cppreference.com/w/cpp/language/class_template#Explicit_instantiation
[4] - https://webrtc-review.googlesource.com/c/src/+/158674

Bug: webrtc:9419
Change-Id: I2e9287a15e28f619462e0b9a5deb0b672be248c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158742
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29661}
2019-10-31 11:12:52 +00:00
de365955dc Added new Apple devices.
Added new apple devices to corresponding enumeration.
Added H264 profile level infromation.
Previous update was done as part of:
https://webrtc-review.googlesource.com/c/src/+/107625
Device machine names obtained from:
https://gist.github.com/adamawolf/3048717

Bug: None
Change-Id: I14aca9dbf495cf50835b388caf38b43145724bd4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158744
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29660}
2019-10-31 10:09:15 +00:00
32913c128a Removes the flakiness in PeerConnectionUsageHistogramTest.
Bug: webrtc:9494, webrtc:11048
Change-Id: I5e6498f10259ee76af682d7019b89bf1f5bb9699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29659}
2019-10-31 00:18:16 +00:00
5bd8cb74a6 Revert "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""
This reverts commit a0adf3d4409036d095480e9bfa0fc06990362f84.

Reason for revert: Suspected of breaking chromium trybots, blocking
webrtc from rolling into chromium.

- First failed roll:
https://chromium-review.googlesource.com/c/chromium/src/+/1889997

- Second failed roll:
https://chromium-review.googlesource.com/c/chromium/src/+/1890837

Example failure:
https://ci.chromium.org/p/chromium/builders/try/linux-rel/230122
Log:
https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8898155661969383856/+/steps/browser_tests__with_patch_/0/logs/Deterministic_failure:_DesktopCaptureApiTest.ChooseDesktopMedia__status_FAILURE_/0

Including lines like:
[12413:12413:1030/102514.183135:INFO:CONSOLE(0)] "[FAIL] screenShareWithAudioPermissionGetStream: NotReadableError: Could not start video source
Error", source: chrome-extension://knldjmfmopnpolahpmmgbagdohdnhkik/_generated_background_page.html (0)

Original change's description:
> Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."
> 
> This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3
> 
> Original change's description:
> > Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
> >
> > Bug: chromium:396091
> > Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> > Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> > Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> > Cr-Commit-Position: refs/heads/master@{#29083}
> 
> Bug: chromium:396091
> Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29655}

TBR=zijiehe@chromium.org,tommi@webrtc.org,julien.isorce@chromium.org,sergeyu@chromium.org,trevor.axiom@gmail.com,jonringle@gmail.com

Change-Id: I2af6a0d5eaf74a0ee536d1c5440049a21d6f7dbf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:396091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158740
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29658}
2019-10-30 18:14:52 +00:00
54d027843a Renaming opus_interface.c to opus_interface.cc.
This is to allow advanced features of WebRTC/Chrome e.g., field trials.

More style compliant changes may follow up. Only a minimal (not in terms of line changes) is applied, so that presubmit does not complain. These changes include

1. removing unused headers.
2. eliminating c-style casting.

Bug: b/143582588
Change-Id: I6d0fd926c542ab0afdc38cc4bf03aaf584ec13dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158670
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29657}
2019-10-30 15:48:28 +00:00
09860e0bc3 Split out counting unique rtp timestamps from packet_buffer
Bug: None
Change-Id: Ia6fd05f284e8304cf56ab9ddf944fb222a4c9573
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158676
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29656}
2019-10-30 15:27:48 +00:00
a0adf3d440 Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."
This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3

Original change's description:
> Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
>
> Bug: chromium:396091
> Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#29083}

Bug: chromium:396091
Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29655}
2019-10-30 12:20:20 +00:00
9560d7dc58 Make update_rect optional in VideoFrame
For the automatic content type detection we need to know if the update
rect is trusted or just not available.

Currently we only care if it's not empty, so in case of no update rect
available, full frame resolution was set as a changed region.

This CL makes the update_rect field optional but should be a no-op in the
current code, as absence of update_rect is treated as a full update via
a new getter method |update_rect_or_full_frame()|.

Bug: webrtc:11058
Change-Id: I913545b71ac2fc845861549ac34eb1b630012109
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158673
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29654}
2019-10-30 11:27:54 +00:00
4e19670d3a [PeerConnection] Implement parameterless SetLocalDescription().
For background, motivation, requirements and implementation notes, see
https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing

The parameterless SetLocalDescription() will implicitly create an
offer or answer to be set by chaining create offer or answer with
setting the session description, as per spec:
https://w3c.github.io/webrtc-pc/#dom-peerconnection-setlocaldescription

Bug: chromium:980885
Change-Id: Ia430160869df18fd47b756b9adf9e7e23ba8e969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157444
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29653}
2019-10-30 10:24:44 +00:00
9b66114878 Disable rendering statistics while video is paused.
Bug: b/142685093
Change-Id: Ie350335f139a82ae247271c3a5a7a9b78a236084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157887
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29652}
2019-10-30 10:04:21 +00:00
4778f6ce7a Revert "Refactoring of the noise suppressor and adding true multichannel support"
This reverts commit 87a7b82520b83a6cf42da27cdc46142c2eb6248c.

Reason for revert: Speculative revert. Breaks downstream projects.

Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
> 
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
> 
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
> 
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
> 
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}

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

Change-Id: I4d4025bda01f484979961fe57380a705e4d78397
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10895, b/143344262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158701
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29651}
2019-10-30 09:49:31 +00:00
9c712bb404 Fix invalid @Nullable handling in TextureBufferImpl.
Bug: None
Change-Id: Ic0b75c62512e9bcb88d562c754e4ed38058a5ece
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157886
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29650}
2019-10-30 09:18:54 +00:00
f298855981 Cleanup of feedback observer interface
Removes all unused features, reducing the exposed interface surface.
This makes refactoring and maintenance simpler as we can change
TransportFeedbackAdapter without making corresponding changes
to RtpVideoSender.

Bug: webrtc:9883
Change-Id: If372a868e0765e94df52b4de52d3bb619ce11471
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156943
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29649}
2019-10-30 07:50:29 +00:00
470b2d5144 Stop relying on GN's sources_assignment_filter.
Recently, on the gn-dev mailing list [1] and on chromium-dev [2] a
consensus about not using "sources_assignment_filter" [3] has been
reached.

This CL removes the implicit dependency on this feature from the
WebRTC codebase in order to make it easier to remove it from GN [4].

[1] - https://groups.google.com/a/chromium.org/forum/#!topic/gn-dev/oQcYStl_WkI
[2] - https://groups.google.com/a/chromium.org/forum/#!topic/chromium-dev/hyLuCU6g2V4
[3] - https://gn.googlesource.com/gn/+/master/docs/reference.md#func_set_sources_assignment_filter
[4] - https://bugs.chromium.org/p/gn/issues/detail?id=125

Bug: webrtc:11057
Change-Id: Ia77820f1b4f9dbc47df2b670148b90928860111a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158677
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29648}
2019-10-30 07:37:39 +00:00
ae40e19805 AEC3: Adding a configurable render signal gain
Bug: webrtc:8671
Change-Id: I405d669517382ce195065caa3147eabace5ec18a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158669
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29647}
2019-10-29 23:26:38 +00:00