Commit graph

94 commits

Author SHA1 Message Date
Markus Handell
0cd0dd3b07 rtc::Event: Finalize migration to TimeDelta.
Bug: webrtc:14366
Change-Id: Icd8792a2f9efa5609dd13da2e175042fac101d36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272101
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37844}
2022-08-19 13:44:57 +00:00
Ali Tofigh
82c29716c0 Adopt absl::string_view in modules/audio_device/
Bug: webrtc:13579
Change-Id: I6e8a90281a9d70a40364b6df5fee4f0a55b4a797
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269060
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37607}
2022-07-25 10:35:17 +00:00
Björn Terelius
7534ebd2bf Revert "Reland "Reland "Delete old Android ADM."""
This reverts commit db30009304.

Reason for revert: ... and it's out again :(
 
Original change's description:
> Reland "Reland "Delete old Android ADM.""
>
> This reverts commit 38a28603fd.
>
> Reason for revert: Attempt to reland, now that WebRTC dependency cycle has been broken.
>
> Original change's description:
> > Revert "Reland "Delete old Android ADM.""
> >
> > This reverts commit 6e4d7e606c.
> >
> > Reason for revert: Still breaks downstream build (though in a different way this time)
> >
> > Original change's description:
> > > Reland "Delete old Android ADM."
> > >
> > > This is a reland of commit 4ec3e9c988
> > >
> > > Original change's description:
> > > > Delete old Android ADM.
> > > >
> > > > The schedule move Android ADM code to sdk directory have been around
> > > > for several years, but the old code still not delete.
> > > >
> > > > Bug: webrtc:7452
> > > > Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > > Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> > > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#37174}
> > >
> > > Bug: webrtc:7452
> > > Change-Id: Icabad23e72c8258a854b7809a93811161517266c
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
> > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#37236}
> >
> > Bug: webrtc:7452
> > Change-Id: Ide8fbd55fadd7aed9989053afff7c63c04f1320f
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266023
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Owners-Override: Björn Terelius <terelius@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37242}
>
> Bug: webrtc:7452
> Change-Id: I6946d0fc28cf4c08387e451e6a07765f7410ce7c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266980
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37356}

Bug: webrtc:7452
Change-Id: I1ef4004e89c8bea322bda0dc697a7ba45abeffcc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267067
Owners-Override: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37359}
2022-06-28 14:37:43 +00:00
Björn Terelius
db30009304 Reland "Reland "Delete old Android ADM.""
This reverts commit 38a28603fd.

Reason for revert: Attempt to reland, now that WebRTC dependency cycle has been broken.

Original change's description:
> Revert "Reland "Delete old Android ADM.""
>
> This reverts commit 6e4d7e606c.
>
> Reason for revert: Still breaks downstream build (though in a different way this time)
>
> Original change's description:
> > Reland "Delete old Android ADM."
> >
> > This is a reland of commit 4ec3e9c988
> >
> > Original change's description:
> > > Delete old Android ADM.
> > >
> > > The schedule move Android ADM code to sdk directory have been around
> > > for several years, but the old code still not delete.
> > >
> > > Bug: webrtc:7452
> > > Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#37174}
> >
> > Bug: webrtc:7452
> > Change-Id: Icabad23e72c8258a854b7809a93811161517266c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37236}
>
> Bug: webrtc:7452
> Change-Id: Ide8fbd55fadd7aed9989053afff7c63c04f1320f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266023
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37242}

Bug: webrtc:7452
Change-Id: I6946d0fc28cf4c08387e451e6a07765f7410ce7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266980
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37356}
2022-06-28 12:58:23 +00:00
Björn Terelius
38a28603fd Revert "Reland "Delete old Android ADM.""
This reverts commit 6e4d7e606c.

Reason for revert: Still breaks downstream build (though in a different way this time)

Original change's description:
> Reland "Delete old Android ADM."
>
> This is a reland of commit 4ec3e9c988
>
> Original change's description:
> > Delete old Android ADM.
> >
> > The schedule move Android ADM code to sdk directory have been around
> > for several years, but the old code still not delete.
> >
> > Bug: webrtc:7452
> > Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37174}
>
> Bug: webrtc:7452
> Change-Id: Icabad23e72c8258a854b7809a93811161517266c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37236}

Bug: webrtc:7452
Change-Id: Ide8fbd55fadd7aed9989053afff7c63c04f1320f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266023
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37242}
2022-06-16 16:07:49 +00:00
Yaowen Guo
6e4d7e606c Reland "Delete old Android ADM."
This is a reland of commit 4ec3e9c988

Original change's description:
> Delete old Android ADM.
>
> The schedule move Android ADM code to sdk directory have been around
> for several years, but the old code still not delete.
>
> Bug: webrtc:7452
> Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37174}

Bug: webrtc:7452
Change-Id: Icabad23e72c8258a854b7809a93811161517266c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37236}
2022-06-16 13:22:29 +00:00
Erik Språng
7dbfad613c Revert "Delete old Android ADM."
This reverts commit 4ec3e9c988.

Reason for revert: Causes downstream build error.

Original change's description:
> Delete old Android ADM.
>
> The schedule move Android ADM code to sdk directory have been around
> for several years, but the old code still not delete.
>
> Bug: webrtc:7452
> Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37174}

Bug: webrtc:7452
Change-Id: If094e0a3ef5a3d340cbd5dfa0a8a88c3e97ba0bf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265393
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37180}
2022-06-10 15:10:04 +00:00
Yaowen Guo
4ec3e9c988 Delete old Android ADM.
The schedule move Android ADM code to sdk directory have been around
for several years, but the old code still not delete.

Bug: webrtc:7452
Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37174}
2022-06-10 08:55:04 +00:00
Niels Möller
ea1e6f44f8 Delete rtc_base/format_macros.h
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.

Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
2022-05-09 12:03:21 +00:00
Björn Terelius
0c68a7aaa7 Use WebRTC's Java VM initialization in tests.
WebRTC should not depend on chromium's //base.

Bug: webrtc:13662
Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-20 08:41:48 +00:00
Xavier Lepaul
0f50cc2849 Remove checks for SDK <= 21
WebRTC’s minSdk is 21, so all those checks are dead code.

Change-Id: I26497fd92259b66d9e5ac6afbb393adf4d904c77
Bug: webrtc:13780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253124
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36140}
2022-03-07 09:56:42 +00:00
Olov Brändström
b732bd5fb5 Add timestamps to AudioDeviceBuffer::SetRecordedBuffer
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will
be used to store audio timestaps in future changes.

This is a part of the A/V sync metric metric feature for mobile. The metric
have already launched for web clients.

Bug: webrtc:13609
Change-Id: I0031843476ff1b573b262308fca52d587fae30b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35851}
2022-01-31 12:32:58 +00:00
Harald Alvestrand
ef5b21e637 Deprecate and remove usage for WARNING log level
Bug: webrtc:13362
Change-Id: Ida112158e4ac5f667e533a0ebfedb400c84df4d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239124
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35425}
2021-11-27 22:21:54 +00:00
Harald Alvestrand
97597c0f51 Remove usage of INFO alias for LS_INFO in log messages
Bug: webrtc:13362
Change-Id: Ifda893861a036a85c045cd366f9eab33c62ebde0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35310}
2021-11-04 13:46:17 +00:00
Byoungchan Lee
02334e07c5 Replace the android support annotation library with androidx's one.
This change does not affect downstream dependencies as androidx.annotation
is fully compatible with android.support.annotation.

Bug: webrtc:11962
Change-Id: I714b473df8d0fee8000ddf3a9beca7c5613db5ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226881
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34839}
2021-08-24 16:02:17 +00:00
Artem Titov
0146a34b3f Use backticks not vertical bars to denote variables in comments for /modules/audio_device
Bug: webrtc:12338
Change-Id: I27ad3a5fe6e765379e4e4f42783558c5522bab38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227091
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34620}
2021-08-02 10:24:10 +00:00
Artem Titov
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
Artem Titov
c8421c4c3e Replace rtc::ThreadChecker with webrtc::SequenceChecker
Bug: webrtc:12419
Change-Id: I825c014cc1c4b1dcba5ef300409d44859e971144
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33136}
2021-02-02 14:56:27 +00:00
Mirko Bonadei
3b68aa346a Move some RTC_LOG to RTC_DLOG.
Some locations in the WebRTC codebase RTC_LOG the value of the
__FUNCTION__ macro which probably is useful in debug mode. Moving
these instances to RTC_DLOG saves ~10 KiB on arm64.

Bug: webrtc:11986
Change-Id: I5d81cc459d2850657a712b9aed80c187edf49a3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203981
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33086}
2021-01-28 10:05:00 +00:00
Karl Wiberg
c95b939667 Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.

Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
2020-11-09 10:47:55 +00:00
Niels Möller
7c85d395d7 Delete unneeded includes of system_wrappers/include/sleep.h
Non-test usage is in modules/audio_device and modules/desktop_capture.

Bug: None
Change-Id: Ie7dd89aa40e6dcfa9e49e1956b87b50fd9f1c227
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190140
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32484}
2020-10-26 09:55:26 +00:00
Markus Handell
5f61282687 Migrate modules/audio_device to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I6d1a7145aaaae2e4cd0c8658fa31a673f857dbd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178814
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31664}
2020-07-08 09:32:12 +00:00
Ivo Creusen
f1393e23a2 Add UMA histogram for actual Android buffer size
Previously a histogram was added to track the requested buffer size,
this CL adds a histogram for the actually used buffer size.

Bug: b/157429867
Change-Id: I04016760982a4c43b8ba8f0e095fe1171b705258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176227
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31385}
2020-05-29 11:14:55 +00:00
Ivo Creusen
bdb5830d69 Add UMA histogram for native audio buffer size in ms
The Android native audio code asks the OS to provide an appropriate
buffer size for real-time audio playout. We should add logging for this
value so we can see what values are used in practice.

Bug: b/157429867
Change-Id: I111a74faefc0e77b5c98921804d6625cba1b84af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176126
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreasson <henrika@chromium.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31368}
2020-05-27 14:33:50 +00:00
Jonas Olsson
b2b2031457 Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
Mirko Bonadei
9f9e20a3dc Fix errorprone issues preventing Chromium Roll.
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889

Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
2019-11-27 12:52:48 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
Alex Narest
44dc241ae8 Allows configuration of playout audio buffer
Playout audio buffer length in Java audio device configuration with fieldtrial.

Bug: webrtc:10928
Change-Id: I79286f09591f4b2c6a6146f23d3dce92a29f6b21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150657
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#29005}
2019-08-29 12:57:14 +00:00
Oleh Prypin
b1686786e8 Add RTC_ prefix to non-standard format specifier macro "PRIdNS"
Some of the macros in format_macros.h follow the C standard and try to fill holes in it (on Windows). But this one has no direct equivalent in the standard and is just mimicking the naming convention. That's not nice.

References:
https://devblogs.microsoft.com/cppblog/c99-library-support-in-visual-studio-2013/
https://stackoverflow.com/a/2524673

Change-Id: I53f3faca2976a5b5d4b04a67ffb56ae0f4e930b2
Bug: webrtc:10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147862
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28794}
2019-08-07 13:36:05 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Sebastian Jansson
c01367db40 Deprecating ThreadChecker specific interface.
All changes outside thread_checker.h are by:
s/CalledOnValidThread/IsCurrent/
s/DetachFromThread/Detach/

Bug: webrtc:9883
Change-Id: Idbb1086bff0817db58e770116acf4c9d60fae8b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27494}
2019-04-08 16:58:07 +00:00
Danil Chapovalov
1c41be6e05 Propagate TaskQueueFactory to AudioDeviceBuffer
keep using GlobalTaskQueueFactory in android/ios bindings.
Switch to DefaultTaskQueueFactory in tests.

Bug: webrtc:10284
Change-Id: I034c70542be5eeb830be86527830d51204fb2855
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130223
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27380}
2019-04-01 08:00:49 +00:00
Mirko Bonadei
185e802971 Prefix AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO with WEBRTC_.
Since it is a WebRTC-only macro, let's prefix it with WEBRTC_.

Bug: None
Change-Id: I309666858ea898dc7cd1a68c21be190f98c87b11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129935
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27327}
2019-03-28 08:44:27 +00:00
Mirko Bonadei
d970807e0c Remove rtc_base/scoped_ref_ptr.h.
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.

More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.

Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:29:58 +00:00
Mirko Bonadei
2fd09a40af Remove deprecated code from audio device.
Bug: webrtc:7306, webrtc:10198
Change-Id: Iaeef4d7449c18325511f1763eba510b385959bfe
Reviewed-on: https://webrtc-review.googlesource.com/c/118446
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26383}
2019-01-24 11:27:38 +00:00
Mirko Bonadei
977c82020c Rename AttachCurrentThreadIfNeeded to avoid clash with function.
A function with the same name exists here [1]. If the two headers are included
together this causes compilation errors.

[1] - https://cs.chromium.org/chromium/src/third_party/webrtc/sdk/android/src/jni/jvm.h?l=27&rcl=82f96e6a56e6230e98ee70de5178d7de69795c26

Bug: None
Change-Id: Icbc680f24a02ec66ea2b5e2b6584a53042cf45c7
Reviewed-on: https://webrtc-review.googlesource.com/c/116662
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26229}
2019-01-11 19:09:23 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Artem Titarenko
69540f4419 Use android Nullable instead of javax Nullable
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.

Original comment from upstream change:

> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.

Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
2018-12-10 15:03:58 +00:00
Niels Möller
140b1d94dc Eliminate use of EventWrapper from android audio device tests
Bug: webrtc:3380
Change-Id: I746d2245966afe89065472d4a6a7447f8c63f9f9
Reviewed-on: https://webrtc-review.googlesource.com/c/110163
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25598}
2018-11-12 13:22:46 +00:00
Paulina Hensman
6c966eaf17 Remove @SuppressLint(NewApi) and guard @TargetApi methods
Also rename runningOnLollipopOrHigher() etc in WebRtcAudioUtils
to runningOnApi21OrHigher() etc since mapping API numbers to
names is error prone.

Bug: webrtc:9818
Change-Id: I4a71de72e3891ca2b6fc2341db9131bb2db4cce7
Reviewed-on: https://webrtc-review.googlesource.com/c/103820
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25009}
2018-10-05 10:36:14 +00:00
henrika
36b3179312 Removes flaky thread checker in AudioDeviceBuffer.
This CL removes a set of DCHECKs in AudioDeviceBuffer (ADB) where the goal has been
to ensure that some methods are called on one and the same native I/O thread.
The implementation of the ADB is platform independent but the underlying (driving)
audio components differ between platforms. This combination has shown to generate complex
corner cases such as:

- OS dependent I/O-thread(s) changes while audio is active
- OS dependent audio device changes and it leads to restart of native I/O threads
- Start/Stop of audio has different timing depending on platform and possibly also usage of
JNI and/or emulators.

To summarize: the gain of maintaining the current strict thread checking (in Debug mode)
is not worth all the efforts trying to resolve complex dynamic cases where the native
I/O threads changes ID.

TBR=glaznev

Bug: b/115385789
Change-Id: I681c89adec497a18b97d2a40421c04ea218fd919
Reviewed-on: https://webrtc-review.googlesource.com/100200
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24723}
2018-09-13 11:41:52 +00:00
Sami Kalliomäki
3d50a31aad Remove redundant initializers from WebRTC Java code.
Removes redundant field initializers such as null, 0 and false.

Bug: webrtc:9742
Change-Id: I1e54f6c6000885cf95f7af8e2701875a78445497
Reviewed-on: https://webrtc-review.googlesource.com/99481
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24676}
2018-09-11 09:58:10 +00:00
henrika
cfbd26df1e Relands Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC
First version was reverted in https://webrtc-review.googlesource.com/c/src/+/97941.
The issue is now fixed.

TBR=ivoc

Bug: b/113648245
Change-Id: If631fdea95aa963952f15e48e9d2d678797dc225
Reviewed-on: https://webrtc-review.googlesource.com/97942
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24573}
2018-09-05 10:24:35 +00:00
Patrik Höglund
e2924d555d Revert "Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC."
This reverts commit f217903a67.

Reason for revert: Breaks downstream tests

Original change's description:
> Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC.
> 
> Also ensures that audio parameters are accessed atomically.
> 
> Bug: b/113648245
> Change-Id: Ic812bfe2b2c4cfb3b00d9d411bb4986dfeda1028
> Reviewed-on: https://webrtc-review.googlesource.com/97331
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24550}

TBR=henrika@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org

Change-Id: I620406f25762cf76db0470b3b29b50bc146935c7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/113648245
Reviewed-on: https://webrtc-review.googlesource.com/97941
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24569}
2018-09-05 08:52:51 +00:00
henrika
f217903a67 Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC.
Also ensures that audio parameters are accessed atomically.

Bug: b/113648245
Change-Id: Ic812bfe2b2c4cfb3b00d9d411bb4986dfeda1028
Reviewed-on: https://webrtc-review.googlesource.com/97331
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24550}
2018-09-04 11:22:53 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
henrika
29e865a5d8 Adds stereo support to FineAudioBuffer for mobile platforms.
...continuation of review in https://webrtc-review.googlesource.com/c/src/+/70781

This CL ensures that the FineAudioBuffer can support stereo and also adapts
all classes which uses the FineAudioBuffer.

Note that, this CL does NOT enable stereo on mobile platforms by default. All it does is to ensure
that we *can*. As is, the only functional change is that all clients
will now use a FineAudioBuffer implementation which supports stereo (see
separate unittest).

The FineAudioBuffer constructor has been modified since it is better to
utilize the information provided in the injected AudioDeviceBuffer pointer
instead of forcing the user to supply redundant parameters.

The capacity parameter was also removed since it adds no value now when the
more flexible rtc::BufferT is used.

I have also done local changes (not included in the CL) where I switch
all affected audio backends to stereo and verified that it works in real-time
on all affected platforms (Androiod:OpenSL ES, Android:AAudio and iOS).

Also note that, changes in:

sdk/android/src/jni/audio_device/aaudio_player.cc
sdk/android/src/jni/audio_device/aaudio_recorder.cc
sdk/android/src/jni/audio_device/opensles_player.cc
sdk/android/src/jni/audio_device/opensles_recorder.cc

are simply copies of the changes done under modules/audio_device/android since we currently
have two versions of the ADM for Android.

Bug: webrtc:9172
Change-Id: I1ed3798bd1925381d68f0f9492af921f515b9053
Reviewed-on: https://webrtc-review.googlesource.com/71201
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22998}
2018-04-24 11:58:54 +00:00
Artem Titov
3d19009c56 Temporary suppress bytebuffer warnings.
Currently this warnings prevernt chromium roll into webrtc, because we
consider them as errors. So to unblock roll all warning are suppressed.
All places are documented into bug and will be fixed later.

TBR=henrika@webrtc.org

Bug: webrtc:9175
Change-Id: I0bf5a4b65eb49308e28f71a92d42b5fad6a99b74
Reviewed-on: https://webrtc-review.googlesource.com/71420
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22956}
2018-04-20 11:45:28 +00:00
henrika
8d7393bb28 FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer.
This work is also done as a preparation for adding stereo support to the
FineAudioBuffer.

Review hints:

Actual changes are in modules/audio_device/fine_audio_buffer.h,cc, the rest is
just adaptations to match these changes.

We do have a forked ADM today, hence, some changes are duplicated.

The changes have been verified on all affected platforms.

Bug: webrtc:6560
Change-Id: I413af41c43809f61455c45ad383fc4b1c65e1fa1
Reviewed-on: https://webrtc-review.googlesource.com/70781
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22938}
2018-04-19 12:20:28 +00:00