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Author SHA1 Message Date
Austin Orion
0bb354c540 Add and refactor functionality into rtc_base/win 
This change moves ScopedComInitializer out of core_audio_utility and
into rtc_base/win so it can be reused elsewhere more easily.

It also adds HSTRING and GetActivationFactory functionality to
rtc_base/win. These two were heavily based on what is already present
base/win.

All of these are necessary for the new window capturer based on the
Windows.Graphics.Capture API. You can see how these APIs will be
used in this CL: 186603: Implement WgcCaptureSession |
https://webrtc-review.googlesource.com/c/src/+/186603

Bug: webrtc:9273
Change-Id: I0a36373aac98be779ccbabe1053bb8d6e234f6a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188523
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32522}
 2020-10-29 20:39:10 +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
Danil Chapovalov
41559a2b46 In modules/audio_device replace mock macros with unified MOCK_METHOD macro 
Bug: webrtc:11564
Change-Id: Ic93bc8272da9d7cd3f4adde5a24c07fd05b894bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175643
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31317}
 2020-05-19 09:11:48 +00:00
Kiran Thind
d5d0a2b546 Fix: rename ms_per_buffer to buffer_duration 
Buffer duration is in seconds, not milliseconds.

No-Try: True
Bug: webrtc:11430
Change-Id: Ib03c2002f2dc6c43e01e50d745d709c2644c8b1e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170500
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30798}
 2020-03-16 11:04:20 +00:00
Fabian Bergmark
9a4eb32477 Change the AudioDeiviceDataObserver to be passed as a unique_ptr. 
Bug: webrtc:11356
Change-Id: If89305f257fd966d83f37dbd03922c4d030b6d8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168771
Commit-Queue: Fabian Bergmark <fabianbergmark@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30575}
 2020-02-20 14:45:15 +00:00
Fabian Bergmark
575c2ad8c5 Support passing the ADM to the ADMWrapper. 
Bug: webrtc:11356
Change-Id: Ie68de35908e80cf395b6558d0725c0462412f333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168482
Commit-Queue: Fabian Bergmark <fabianbergmark@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30543}
 2020-02-18 14:13:46 +00:00
Danil Chapovalov
5528402ef8 Use newer version of TimeDelta and TimeStamp factories in modules/ 
This change generated with following commands:
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I117d64a54950be040d996035c54bc0043310943a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30489}
 2020-02-10 11:49:57 +00:00
Steve Anton
760fd52494 Replace MockAudioDeviceModule mock refcounting with real refcounting 
Bug: webrtc:11308
Change-Id: Ic55ec2c4b45f8fc709fe1348556bdeea6202e7a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166580
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30366}
 2020-01-23 19:04:58 +00:00
Anders Klemets
eb8c4ca608 Remove unnecessary checks from AudioDeviceWindowsCore::CoreAudioIsSupported 
This removes some code in the AudioDeviceWindowsCore::CoreAudioIsSupported function that was checking that every audio input and output device was functional. There are legitimate cases where some, or all, audio devices may not be accessible, and that was causing CoreAudioIsSupported to return false.

If CoreAudioIsSupported returns false, a subsequent RTC_CHECK call fails, which causes the entire app to exit.

After this change, the CoreAudioIsSupported() function simply checks if the Core Audio APIs are supported and no longer tries to do extra stuff unrelated to checking if the APIs are supported.

Note that Core Audio is actually supported in all versions of Windows after Windows XP. There were log messages in the code saying that if CoreAudioIsSupported() returns false, WebRTC will use the Wave Audio APIs instead. But this is no longer the case. The Wave Audio APIs would only be needed for Windows XP, and this code appears to have already been removed from WebRTC.
It is tempting to simply make CoreAudioIsSupported() do a "return true;" but for now I only removed the part of the logging messages that mentioned the Wave Audio APIs.

I understand that there is a new Audio Device Module (ADM) called WindowsCoreAudio2, which is now recommended for use by apps. Apps are supposed to instantiate WindowsCoreAudio2 and pass it in to WebRTC. When the app supplies its own ADM, CoreAudioIsSupported() does not get invoked, which avoids the bug. To help make it clearer that using WindowsCoreAudio2 is an acceptable solution, I am removing a comment that says that kWindowsCoreAudio2 is "experimental".

Bug: webrtc:11081
Change-Id: I7ed1684a276799f4c83006b45629e48814f0b18b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161463
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30025}
 2019-12-06 10:09:03 +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
bbeb10925e Reporting audio device underrun counter 
Bug: webrtc:10884
Change-Id: I35636fcbc1e2a19a89242379cdff6ec5c12fd21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28874}
 2019-08-16 11:49:55 +00:00
henrika
d8c6ec4d2f Adds support for disabling autostart in ADM2 for Windows 
Landing with TBR given vacation times and the fact that none of this
code is active "in production". The ADM2 implementation can be seen
as experimental (non-default) code and it takes some work to enable it
and replace the existing ADM. Hence, extremely low risk to break
anything.

TBR: henrik.lundin
Bug: webrtc:9265
Change-Id: Ia5cfb2aaa8eaf9537b916b3375f55d8df6287071
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145921
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28600}
 2019-07-18 13:48:15 +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
Niels Möller
44bc19b0f8 Delete TestAudioDeviceModule methods using rtc::PlatformFile 
Bug: webrtc:6463
Change-Id: I5d1d9e9036b5e745d5b37c971de91b1b38fdd368
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141666
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28255}
 2019-06-12 15:28:41 +00:00
Danil Chapovalov
08fa953711 Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" 
This reverts commit fd5166c305068772d00ad7edf50151bba215400b.

Reason for revert: Stop using CreateTestAudioDeviceModule in downstream

Original change's description:
> Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
> 
> This reverts commit fc961357a721cd87dcd45ed409c66cb8cda6f4a2.
> 
> Reason for revert: Breaks downstream importer.
> 
> Original change's description:
> > Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
> > 
> > Bug: webrtc:10284
> > Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28227}
> 
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10284
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28228}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,philipel@webrtc.org

Change-Id: I42bc19793d48350ca45b751d7e1b26124ac7fbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141670
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28254}
 2019-06-12 14:44:01 +00:00
Philip Eliasson
fd5166c305 Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" 
This reverts commit fc961357a721cd87dcd45ed409c66cb8cda6f4a2.

Reason for revert: Breaks downstream importer.

Original change's description:
> Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
> 
> Bug: webrtc:10284
> Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28227}

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

Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28228}
 2019-06-11 12:32:23 +00:00
Danil Chapovalov
fc961357a7 Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory 
Bug: webrtc:10284
Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28227}
 2019-06-11 12:15:44 +00:00
Danil Chapovalov
48edc9224c Delete deprecated AudioDeviceWithDataObserver factory 
Bug: webrtc:10284
Change-Id: I00ccba2c84e47f2b97bdd9c841467ccc0c6f900f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140281
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28162}
 2019-06-05 09:01:25 +00:00
Danil Chapovalov
98499d5a20 Remove deprecated AudioDeviceModule factory 
Bug: webrtc:10284
Change-Id: If1c732b113c5d340dfc800f55f4d567576e82ce3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132222
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27801}
 2019-04-29 11:22:11 +00:00
Artem Titov
153056b059 Add ability to play audio in circle for TestAudioDevice wav file capturer 
Also use this ability in PC smoke test.

Bug: webrtc:10138
Change-Id: I83d526344f203082a19377d9642c9e453454f7ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133163
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27649}
 2019-04-16 15:33:03 +00:00
Artem Titov
dd1c16f00c Use absl::make_unique in TestAudioDeviceModule factory methods 
Bug: webrtc:10138
Change-Id: Ibe9f4b4343b8e5c9a5e1a6d41bd06b24d69db878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133166
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27646}
 2019-04-16 14:43:55 +00:00
Mirko Bonadei
6a489f22c7 Fully qualify googletest symbols. 
Semi-automatically created with:

git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format

After this, two .cc files failed to compile and I have fixed them
manually.

Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
 2019-04-09 17:18:20 +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
Sebastian Jansson
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
Seth Hampson
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
Sebastian Jansson
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
Mirko Bonadei
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
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
Artem Titov
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
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
Steve Anton
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
Niels Möller
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
Niels Möller
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
Niels Möller
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
Yves Gerey
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
Niels Möller
1c9d7bbeaf Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper." 
This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a

Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
> 
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
> 
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}

Bug: webrtc:3380
Change-Id: I39c6b35d24182475b33a7a321cdf3b3ac9b8979a
Reviewed-on: https://webrtc-review.googlesource.com/97861
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24602}
 2018-09-06 11:31:57 +00:00
henrika
5b6afc0ce6 Adds stream-switch support in new Windows ADM. 
Second round of the new Windows ADM is now ready for review. Main
changes are:

Supports internal (automatic) restart of audio streams when an active
audio stream disconnects (happens when a device is removed).

Adds support for IAudioClient3 and IAudioClient2 for platforms which
supports it (>Win8 and >Win10).

Modifies the threading model to support restart "from the inside" on
the native audio thread.

Adds two new test methods for the ADM to emulate restart events or
stream-switch events.

Adds two new test methods to support rate conversion to ensure that
audio can be tested in loopback even if devices runs at different
sample rates.

Added initial components for low-latency support. Verified that it works
but disabled it with a flag for now.

Bug: webrtc:9265
Change-Id: Ia8e577daabea6b433f2c2eabab4e46ce8added6a
Reviewed-on: https://webrtc-review.googlesource.com/86020
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24578}
 2018-09-05 13:04:01 +00:00
Niels Moller
4c6747c9db Revert "Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper."" 
This reverts commit cd87e014f34069fd5a73c1ed5b74ddf251a95c2d.

Reason for revert: Somehow introduces a race where rtc::Thread auto-wrapping may be applied to the TestAudioDevice thread rather than the main thread. This causes failures when running video_engine_tests without any test filter.

Original change's description:
> Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper."
>
> This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a
>
> Original change's description:
> > Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
> >
> > In addition, let the processing thread loop explicitly, and not use
> > the deprecated builtin looping in PlatformThread.
> >
> > Bug: webrtc:3380
> > Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> > Reviewed-on: https://webrtc-review.googlesource.com/96544
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24492}
>
> Bug: webrtc:3380
> Change-Id: I671e3a60ace6ade765a8537b7e20e36f1782a60d
> Reviewed-on: https://webrtc-review.googlesource.com/97320
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24528}

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

# Not skipping CQ checks because original CL landed > 1 day ago.
# Skip anyway, needed for a two-step revert.

No-try: True
Bug: webrtc:3380
Change-Id: Ia7c8cfab36b8259f150b5ccd0c28defd0e7237f6
Reviewed-on: https://webrtc-review.googlesource.com/97682
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24552}
 2018-09-04 13:04:11 +00:00
Niels Möller
cd87e014f3 Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper." 
This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a

Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
> 
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
> 
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}

Bug: webrtc:3380
Change-Id: I671e3a60ace6ade765a8537b7e20e36f1782a60d
Reviewed-on: https://webrtc-review.googlesource.com/97320
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24528}
 2018-09-03 08:58:11 +00:00
Niels Moller
6be91eb2f8 Revert "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper." 
This reverts commit 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a.

Reason for revert: Makes the perf test RampUpTest.AudioTransportSequenceNumber fail on windows, almost every time.

Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
> 
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
> 
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}

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

Change-Id: I8867a22d695494bd5abfda6a97f0719cb3ff3d66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3380
Reviewed-on: https://webrtc-review.googlesource.com/96840
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24496}
 2018-08-30 12:59:13 +00:00
Niels Möller
9ea5765f78 Refactor TestAudioDeviceModule to not depend on EventTimerWrapper. 
In addition, let the processing thread loop explicitly, and not use
the deprecated builtin looping in PlatformThread.

Bug: webrtc:3380
Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
Reviewed-on: https://webrtc-review.googlesource.com/96544
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24492}
 2018-08-30 10:40:01 +00:00
Niels Möller
a12c42a6b2 Delete root header file typedef.h. 
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
 2018-07-25 14:59:26 +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
Sami Kalliomäki
a97c931cba Fix a bug where TestAudioDeviceModule crashes if destroyed uninitialized. 
Because thread_ object is created in Init, destructor used to crash when
calling thread_->Stop() because it was referencing a null pointer.

Bug: webrtc:9404
Change-Id: I1c943d0fa50f9341aaa516b32495bb25bf4d664b
Reviewed-on: https://webrtc-review.googlesource.com/84122
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23682}
 2018-06-20 12:27:36 +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
Sami Kalliomäki
e1d617c266 Delay the creation of the platform thread in TestAudioDeviceModule. 
This allows constructing TestAudioDeviceModule on a different thread
than the worker thread and avoids unnecessary invoke. Before,
thread->Start() would fail in a thread check.

Bug: b/79961243
Change-Id: I5c55d8feada2b0ae12bc121f3f795e76a8d04059
Reviewed-on: https://webrtc-review.googlesource.com/82941
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23574}
 2018-06-12 07:36:28 +00:00
henrika
ec9c745228 Adds support for new Windows ADM with limited API support. 
Summary of what this CL does:

Existing users can keep using the old ADM for Windows as before.

A new ADM for Windows is created and a dedicated factory method is used
to create it. The old way (using AudioDeviceImpl) is not utilized.

The new ADM is based on a structure where most of the "action" takes
place in new AudioInput/AudioOutput implementations. This is inline
with our mobile platforms and also makes it easier to break out common
parts into a base class.

The AudioDevice unittest has always mainly focused on the "Start/Stop"-
parts of the ADM and not the complete ADM interface. This new ADM supports
all tests in AudioDeviceTest and is therefore tested in combination with
the old version. A value-parametrized test us added for Windows builds.

Improved readability, threading model and makes the code easier to maintain.

Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility.

Bug: webrtc:9265
Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d
Reviewed-on: https://webrtc-review.googlesource.com/78060
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23554}
 2018-06-08 14:44:38 +00:00
Magnus Jedvert
7dfd5fc3df AudioTransport: Remove PushCaptureData() method 
This CL removes PushCaptureData(), which is unused.

The reason I'm removing it is since this method is cauing chromium-style
violations for all files that includes
modules/audio_device/include/audio_device_defines.h, and it's annoying
to suppress it everywhere.

Bug: webrtc:8659
Change-Id: I9133d05259075d8e8ec89b764be934f37b5fa77e
Reviewed-on: https://webrtc-review.googlesource.com/66404
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22717}
 2018-04-04 08:04:09 +00:00
Artem Titov
ac9365ed64 Set safe values to prevent possible sigsegv while using AudioTransport, add doc 
Bug: webrtc:8946
Change-Id: Ica066a05905894fba6ba24e45af46b0d5951b5d5
Reviewed-on: https://webrtc-review.googlesource.com/65040
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22652}
 2018-03-28 15:05:26 +00:00