Commit Graph

340 Commits

Author SHA1 Message Date
e9d2b4efdd Revert "Remove old audio device implementation."
This reverts commit 0cfa4cba5cae5e942f5d8e0e4e93b94982d0bfc3.

Reason for revert: audio_device_ios_objc target is removed, but still referenced by iPhone Meetins:Meeting_build_test, which now fails to build

Original change's description:
> Remove old audio device implementation.
> 
> The iOS ADM implementation now lives in sdk/objc/native/api/audio_device_module.{h,mm}.
> 
> Bug: webrtc:10514
> Change-Id: Ib0b162027b5680ebc40d621a57f1155f08e7a057
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131326
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27488}

TBR=henrika@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org

Change-Id: I5be10b3d17403a79ea30afc255cde01171bc9f5b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10514
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131960
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27492}
2019-04-08 15:00:10 +00:00
0cfa4cba5c Remove old audio device implementation.
The iOS ADM implementation now lives in sdk/objc/native/api/audio_device_module.{h,mm}.

Bug: webrtc:10514
Change-Id: Ib0b162027b5680ebc40d621a57f1155f08e7a057
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131326
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27488}
2019-04-08 14:08:48 +00:00
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
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
673f7e56e4 Do not assume /DUNICODE and /D_UNICODE.
As a library, WebRTC should not assume UNICODE and _UNICODE to be
defined globally.

This CL explicitly selects wide character functions and types in
order to build WebRTC with /UUNICODE and /U_UNICODE.

Bug: None
Change-Id: Ie4e2bcb4c5c34aee6f68dc7b5b54b76f088ee3e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128904
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Cr-Commit-Position: refs/heads/master@{#27313}
2019-03-27 14:18:41 +00:00
c771c805f1 Use scoped_refptr to share the instance of OpenSLEngineManager
Use rtc::scoped_refptr instead of std::unique_ptr to hold the instance
of OpenSLEngineManager; this makes it safe to share it between
OpenSLESRecorder and OpenSLESPlayer.

Bug: webrtc:10436
Change-Id: Ibd0717e5410020c89a40bfdb05953a02378a6a4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128651
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27253}
2019-03-22 20:15:42 +00:00
56325786d4 Mark MediaBufferImpl as final.
Without being marked final, calling "delete this" produces a warning
because IMediaBuffer/IUnknown don't have a virtual destructor when
-Wdelete-non-virtual-dtor is set.

Bug: None
Change-Id: I1905e4e5b0beab695cc7a4b901542b3aeb789b20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128821
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Noah Richards <noahric@chromium.org>
Cr-Commit-Position: refs/heads/master@{#27251}
2019-03-22 19:23:51 +00:00
94b57c044e Cleanup BUILD.gn files from imports like foo:foo
Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn
files.

Done with Sublime regex replace.
Find: \b([-a-zA-Z0-9_]+):+\1\b
In: *.gn
Replace with: \1

Bug: None
Change-Id: I40aba1b14face687a595b852ffe443cb20197611
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:05:28 +00:00
77efcd82db Reland "Replacing rtc::Thread with task queue for TestAudioDeviceModule."
This is a reland of 1b871d07532c25d2f27e4db192cb9ce2229b1cee

Original change's description:
> Replacing rtc::Thread with task queue for TestAudioDeviceModule.
>
> This prepares for running it in simulated time.
>
> TBR=henrika@webrtc.org
>
> Bug: webrtc:10465
> Change-Id: I9b29b8af9aeba7f0c8209ca77294a63d8068ff1a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126481
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27083}

TBR=henrika@webrtc.org

Bug: webrtc:10465
Change-Id: Icda8043fb5b1156129bc3b706bf8f190782b0921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127520
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27093}
2019-03-13 09:01:05 +00:00
fa852efb73 Revert "Replacing rtc::Thread with task queue for TestAudioDeviceModule."
This reverts commit 1b871d07532c25d2f27e4db192cb9ce2229b1cee.

Reason for revert: Breaks webrtc downstream projects.

Original change's description:
> Replacing rtc::Thread with task queue for TestAudioDeviceModule.
> 
> This prepares for running it in simulated time.
> 
> TBR=henrika@webrtc.org
> 
> Bug: webrtc:10465
> Change-Id: I9b29b8af9aeba7f0c8209ca77294a63d8068ff1a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126481
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27083}

TBR=henrika@webrtc.org,ossu@webrtc.org,srte@webrtc.org

Change-Id: I16d7c2a46d38c9aaf82cc3ab7bd7b9c5e10f5a5e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127341
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27086}
2019-03-12 20:05:53 +00:00
1b871d0753 Replacing rtc::Thread with task queue for TestAudioDeviceModule.
This prepares for running it in simulated time.

TBR=henrika@webrtc.org

Bug: webrtc:10465
Change-Id: I9b29b8af9aeba7f0c8209ca77294a63d8068ff1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126481
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27083}
2019-03-12 18:25:31 +00:00
7bc331f664 Android: Use android_deps directly
This is preparing for upstream removing the alias java_groups for the
individual support library targets: https://crrev.com/c/1500780

Bug: chromium:937987
Change-Id: I1c9efd83f6997288b334f3dc2f41233fa4e2ab61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125961
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Peter Wen <wnwen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#26995}
2019-03-06 15:14:42 +00:00
fc52b912a3 Implicitly suppress //build/config/clang:find_bad_constructs.
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).

The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.

Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.

[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html

Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
2019-03-01 10:18:17 +00:00
c4dd730765 Fix -Wextra-semi warnings.
Starting from https://chromium-review.googlesource.com/c/1485012,
-Wextra-semi is enabled and WebRTC has some violations to fix.

This is a follow-up of https://webrtc-review.googlesource.com/c/123560.

Bug: webrtc:10355
Change-Id: I012b7497fc8991037fd77aa98f1579c22e08206f
Reviewed-on: https://webrtc-review.googlesource.com/c/124126
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26831}
2019-02-25 09:22:51 +00:00
1c54605e77 [clang-tidy] Apply performance-move-const-arg fixes (misc).
This CL is a manual spin-off of [1], which tried to apply clang-tidy's
performance-move-const-arg [1] to the WebRTC codebase.

Since there were some wrong fixes to correct, this CL lands a few
different fixes, like adding a constructor overload to take an rvalue
reference or remove 'const' to make std::move effective.

[1] - https://webrtc-review.googlesource.com/c/src/+/120350
[2] - https://clang.llvm.org/extra/clang-tidy/checks/performance-move-const-arg.html

Bug: webrtc:10252
Change-Id: I42a777247fee2cb788efcd7c2035148330056b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/120928
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26553}
2019-02-05 15:12:20 +00:00
c84f661b10 Stop using Googletest legacy APIs.
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.

This CL moves WebRTC to the new set of APIs.

More info in [1].

This CL has been generated with this script:

declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
  git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format

[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature

Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
2019-01-31 13:23:33 +00:00
fe055c197a [clang-tidy] Apply modernize-use-override fixes.
This CL applies clang-tidy's modernize-use-override [1] to the
WebRTC codebase.

All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html

Bug: webrtc:10252
Change-Id: I2bb8bd90fa8adb90aa33861fe7c788132a819a20
Reviewed-on: https://webrtc-review.googlesource.com/c/120412
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26461}
2019-01-30 09:26:17 +00:00
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
d3a5aaa521 Check "rtc_include_internal_audio_device" before creating unittest for audio_device_ios_objc
Bug: webrtc:10241
Change-Id: I335718c81436502cc492c9142220cd023b7da80c
Reviewed-on: https://webrtc-review.googlesource.com/c/119860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#26412}
2019-01-25 18:51:07 +00:00
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
5a6ae02e90 Reland "Trim down FileWrapper class to be merely a wrapper owning a FILE*"
This is a reland of 80b95de7651caa0cfeb684ffc200860989f667dc

Original change's description:
> Trim down FileWrapper class to be merely a wrapper owning a FILE*
> 
> Bug: webrtc:6463
> Change-Id: If71e2f3a75dc1863bc805ab71de1e2d33294f805
> Reviewed-on: https://webrtc-review.googlesource.com/c/117881
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26311}

Bug: webrtc:6463
Change-Id: I12154ef65744c1b7811974a1d871e05ed3fbbc27
Reviewed-on: https://webrtc-review.googlesource.com/c/118660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26337}
2019-01-21 12:46:25 +00:00
466472796c Revert "Trim down FileWrapper class to be merely a wrapper owning a FILE*"
This reverts commit 80b95de7651caa0cfeb684ffc200860989f667dc.

Reason for revert: Speculative revert for downstream breakage. Possibly FileAudioDevice broken?

Original change's description:
> Trim down FileWrapper class to be merely a wrapper owning a FILE*
> 
> Bug: webrtc:6463
> Change-Id: If71e2f3a75dc1863bc805ab71de1e2d33294f805
> Reviewed-on: https://webrtc-review.googlesource.com/c/117881
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26311}

TBR=aleloi@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,tommi@webrtc.org

Change-Id: I46d37afbf9acb5f62f04e09d944114c1da96eb36
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6463
Reviewed-on: https://webrtc-review.googlesource.com/c/118380
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26318}
2019-01-18 12:04:55 +00:00
80b95de765 Trim down FileWrapper class to be merely a wrapper owning a FILE*
Bug: webrtc:6463
Change-Id: If71e2f3a75dc1863bc805ab71de1e2d33294f805
Reviewed-on: https://webrtc-review.googlesource.com/c/117881
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26311}
2019-01-18 07:25:30 +00:00
66a29b9953 Introduce CopyToFileAudioCapturer.
It will be used to dump generated audio from TestAudioDeviceModule into
user defuned file in peer connection level test framework.

Bug: webrtc:10138
Change-Id: I6e3db36aaf1303ab148e8812937c4f9cd1b49315
Reviewed-on: https://webrtc-review.googlesource.com/c/117220
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26267}
2019-01-15 15:06:55 +00:00
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
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
40d55331d7 Include absl/memory/memory.h if absl::make_unique is used
Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Iaf4533d2ce0e80b351a8a664ef8cf7ba0e5ec583
Reviewed-on: https://webrtc-review.googlesource.com/c/115746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26168}
2019-01-08 20:08:32 +00:00
ff98f4b1d8 Fix stop logging errors for stereo mode when it is not used
When using WebRTC in iOS this Warning is shown for every single call even if stereo is not being used at all.

Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26

Bug: webrtc:10146
Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26
Reviewed-on: https://webrtc-review.googlesource.com/c/85283
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26133}
2019-01-04 09:36:52 +00:00
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
ebad1770ab Include event_wrapper.h only where used.
It's currently used only by the VCMJitterBuffer and VCMReceiver
classes. Injection is needed by the VCMReceiverTimingTest test, which
defines a subclass(!) of EventWrapper.

Bug: webrtc:3380
Change-Id: I765be0ceac58e941928319cc426ba49f1cbdc5fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25893}
2018-12-04 14:50:18 +00:00
2222a80e79 Delete unneeded includes of common_types.h and gn deps on webrtc_common.
Bug: webrtc:5876
Change-Id: Iae14e5f1679067a5a5e0584ca830aee0870c8807
Reviewed-on: https://webrtc-review.googlesource.com/c/111463
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25715}
2018-11-20 16:28:39 +00:00
7dbb7c311f Adding missing build target for audio_device_default.
The header modules/audio_device/include/audio_device_default.h was not
owned by any build target.

Bug: webrtc:8946
Change-Id: I3266a613c10963688c3bea701384e1d1bb68daac
Reviewed-on: https://webrtc-review.googlesource.com/c/111201
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25669}
2018-11-16 11:15:08 +00:00
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
b0550bdf96 Eliminate use of EventWrapper from mac audio device
Bug: webrtc:3380
Change-Id: I9b34588a6a2b035f1787782421e4fc3e6650ef1a
Reviewed-on: https://webrtc-review.googlesource.com/c/110244
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25567}
2018-11-08 15:52:53 +00:00
5bb1ed6144 Eliminate use of EventWrapper from ios audio device tests
Bug: webrtc:3380
Change-Id: I2d2f8a7152212e80600449d49e7f7316dd89bfc2
Reviewed-on: https://webrtc-review.googlesource.com/c/110200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25563}
2018-11-08 15:12:50 +00:00
c572ff3c71 Add default constructor for rtc::Event
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
2018-11-07 08:57:50 +00:00
27f31727d0 Simplify use of events in TestAudioDevice
Create events with |manual_reset| and |initially_signalled| both false
(used to be both true). Delete calls to Set and Reset events from the
{Start,Stop}{Playout,Recording} methods. Then, for each event, there
remains a single call to Set, in the ProcessingAudio loop, and a
single call to Wait, in WaitForPlayoutEnd and WaitForRecordingEnd,
respectively.

Bug: webrtc:9962
Change-Id: Ia358b4a36896e2378ad6166f3786d8d71392bf1b
Reviewed-on: https://webrtc-review.googlesource.com/c/109562
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25518}
2018-11-06 12:18:05 +00:00
2c16cc61c2 Replace some usage of EventWrapper with rtc::Event.
Bug: webrtc:3380
Change-Id: Id33b19bf107273e6f838aa633784db73d02ae2c2
Reviewed-on: https://webrtc-review.googlesource.com/c/107888
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25407}
2018-10-29 09:37:24 +00:00
7d76a31f3d Use MediaTransportInterface, for audio streams.
Bug: webrtc:9719
Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239
Reviewed-on: https://webrtc-review.googlesource.com/c/104642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25385}
2018-10-26 11:40:57 +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
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
433eafe1f5 Delete unused includes of assert.h
Bug: None
Change-Id: Iadc531710dca0ba34a00ac363bfe0784355bb6f9
Reviewed-on: https://webrtc-review.googlesource.com/c/103501
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24995}
2018-10-04 14:01:44 +00:00
4f340fa01e Compile audio_device without -Wno-global-constructors.
This CL removes kNumMicrosecsPerSec and kNumMillisecsPerSec from
modules/audio_device/win/core_audio_utility_win.h.

kNumMillisecsPerSec was unused, while kNumMicrosecsPerSec has been
replaced by rtc::kNumMicrosecsPerSec.

Bug: webrtc:9693
Change-Id: I560aa9dad2bfb94a9bf67d3b9941700f1948086b
Reviewed-on: https://webrtc-review.googlesource.com/102860
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24906}
2018-10-01 08:49:51 +00:00
78e0ac1b39 Improves threading model in AudioDeviceTest.
These changes are based on finding when using Tsan v2. More changes are
needed before usage of the THREAD_SANITIZER build flag can be removed.
Hence, all tests are still ignored when this flag is set. The changes
are still improvements.

See https://bugs.chromium.org/p/webrtc/issues/detail?id=9778#c10
for more details.

Bug: webrtc:9778
Change-Id: I1266cec48165046dcffc16f104ec5b88b41500b2
Reviewed-on: https://webrtc-review.googlesource.com/102440
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24880}
2018-09-28 08:19:47 +00:00
17f4878419 Remove deprecated field_trial_default and metrics_default.
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default

It also refreshes all the dependencies on field_trial.h and metrics.h.

A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm

Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 07:21:07 +00:00
e0c01b9802 Fix global_constructors, exit_time_destructors in audio device pulse.
Bug: webrtc:9693
Change-Id: I05498473be8a86756d65d0b9000d626c966d4ed3
Reviewed-on: https://webrtc-review.googlesource.com/100422
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24865}
2018-09-27 13:19:11 +00:00
a3b9b27412 Also disable AudioDeviceTest constructor for TSAN.
The flaky crash is happening in the constructor.

Bug: webrtc:9778
Change-Id: I9ac9a89e033e17de690e594ef263ff83d14fcc5a
Reviewed-on: https://webrtc-review.googlesource.com/101880
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24829}
2018-09-25 14:44:00 +00:00
f1239b5405 Fixes issue where WebRTC.Audio.RecordSampleRateOffsetInPercent can report 100% (part II)
See https://webrtc-review.googlesource.com/c/src/+/100241 for part I.

It can happen that recording fail to start but playout works. If that happens, we can
log stats like this:

9-25 15:32:02.023 13903 13948 I audio_device_buffer.cc: (line 414): [REC : 10002msec, 48kHz] callbacks: 0, samples: 0, rate: 0, rate diff: 100%, level: 0
09-25 15:32:02.024 13903 13948 I audio_device_buffer.cc: (line 432): [PLAY: 10002msec, 48kHz] callbacks: 1002, samples: 480960, rate: 48086, rate diff: 0%, level: 0
09-25 15:32:12.028 13903 13948 I audio_device_buffer.cc: (line 414): [REC : 10005msec, 48kHz] callbacks: 0, samples: 0, rate: 0, rate diff: 100%, level: 0
09-25 15:32:12.028 13903 13948 I audio_device_buffer.cc: (line 432): [PLAY: 10005msec, 48kHz] callbacks: 1000, samples: 480000, rate: 47976, rate diff: 0%, level: 0

hence, we log invalid UMA stats for a rate offset of 100%. This change fixes the problem, and in the
case above, we now instead log:

09-25 15:35:56.141 14116 14161 I audio_device_buffer.cc: (line 432): [PLAY: 10011msec, 48kHz] callbacks: 1002, samples: 480960, rate: 48043, rate diff: 0%, level: 0
09-25 15:36:06.151 14116 14161 I audio_device_buffer.cc: (line 432): [PLAY: 10010msec, 48kHz] callbacks: 1002, samples: 480960, rate: 48048, rate diff: 0%, level: 0
09-25 15:36:16.162 14116 14161 I audio_device_buffer.cc: (line 432): [PLAY: 10011msec, 48kHz] callbacks: 1001, samples: 480480, rate: 47995, rate diff: 0%, level: 0
09-25 15:36:26.173 14116 14161 I audio_device_buffer.cc: (line 432): [PLAY: 10011msec, 48kHz] callbacks: 1001, samples: 480480, rate: 47995, rate diff: 0%, level: 0

Bug: b/113648245
Change-Id: Ic8cb71ca049ef24bf68963a81f95d4e5c2282518
Reviewed-on: https://webrtc-review.googlesource.com/101881
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24828}
2018-09-25 14:28:10 +00:00
defb717811 Disable AudioDeviceTest on thread sanitizer.
These tests have become flaky on thread sanitizer.

Bug: webrtc:9778
Change-Id: Ife7004acf11ef1b3cb9d5720da25d4efb2395faf
Reviewed-on: https://webrtc-review.googlesource.com/101841
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24824}
2018-09-25 11:45:39 +00:00
5773ad3bc8 Ensures that ADM unittest uses default audio devices for all platforms.
TBR=ossu

Bug: webrtc:9265
Change-Id: Ifc6d3f9c5c4a4e31dcedfd72ed96a2bde5d074e7
Reviewed-on: https://webrtc-review.googlesource.com/101262
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24786}
2018-09-24 08:00:59 +00:00