Commit Graph

22 Commits

Author SHA1 Message Date
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
c7d935899a Adds multi-channel support to new ADM2 on Windows.
Now checks the preferred channel configuration and requests implicit channel
upmixing (audio engine extends from 2 to N channels internally) if the
preferred number of channels is larger than two; i.e., initialize the
stream in stereo even if the preferred configuration is multi-channel.

To summarize: with this CL, it is now possible to use e.g. a 7.1 headset
with a native WebRTC client. All internal processing in WebRTC will be in
stereo, and the audio device will be opened up in stereo as well to match
WebRTC. Before this change, we would open up the audio device using 8
channels but that was not supported by WebRTC.

Bug: webrtc:9265
Change-Id: I1530fee28c4b8b5cda29ab6baf8d65fd391d935d
Reviewed-on: https://webrtc-review.googlesource.com/98421
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24743}
2018-09-14 14:16:27 +00:00
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
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
196100efa6 Replace rtc::Optional with absl::optional
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script passing top level directories except rtc_base and api

find $@ -type f \( -name \*.h -o -name \*.cc -o -name \*.mm \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I9465c172e65ba6e6ed4e4fdc35b0b265038d6f71
Reviewed-on: https://webrtc-review.googlesource.com/84584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23697}
2018-06-21 09:32:56 +00:00
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
eb98c727e6 Minor improvements in ADM unittest for Windows.
This CL resolves some minor issues related to running ADM unittests on Windows.
It is rather common on Windows that devices can't be opened up in mono mode and
some tests have been hardcoded to use mono and that leads to crashes and/or error
logs. Now, all tests runs in stereo as well.

NOTRY=TRUE

Bug: None
Change-Id: Iebf11a6ff63c19ff1be45575a8e0a3df4e112bd4
Reviewed-on: https://webrtc-review.googlesource.com/62940
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22510}
2018-03-20 11:57:35 +00:00
1e06289cdb Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.

Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
2018-02-07 10:07:28 +00:00
6ad9f26975 Optional: Use nullopt and implicit construction in /modules/audio_device
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

Bug: None
Change-Id: I9f581b44a43db701300e100a3ff70d90e3dd4f6e
Reviewed-on: https://webrtc-review.googlesource.com/23572
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21581}
2018-01-11 13:55:11 +00:00
5d2bb36b95 Fix linker error when building without internal audio device.
Bug: webrtc:8678
Change-Id: I30f0d1aa7bec6931edb9e5ba00a6813dea1536ff
Reviewed-on: https://webrtc-review.googlesource.com/35480
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21408}
2017-12-21 09:50:00 +00:00
e26456a4ed Removes usage of AGC APIs in the ADM.
Bug: webrtc:8598
Change-Id: I5ebc2e3549eba039797e40d2f8aea48341f3fe46
Reviewed-on: https://webrtc-review.googlesource.com/31520
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21254}
2017-12-13 16:32:21 +00:00
e40468ba3d Move some numeric utility code from rtc_base/ to rtc_base/numerics/
Specifically, I'm moving

  safe_compare.h
  safe_conversions.h
  safe_minmax.h

They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.

BUG=webrtc:8445

Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
2017-11-22 11:21:47 +00:00
616e3138b8 Removes id parameter in ADM factory method (reland)
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/21980

Bug: webrtc:7306
Change-Id: Idbbb6d5371b154247ea3db55c28e5c0618c4fae1
Reviewed-on: https://webrtc-review.googlesource.com/22480
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20653}
2017-11-13 13:28:18 +00:00
cda2562b06 Revert "Removes id parameter in ADM factory method"
This reverts commit cba3d274daaf57b3c6e8bc6cd10b959cf2ec73bf.

Reason for revert: Breaking internal projects

Original change's description:
> Removes id parameter in ADM factory method
> 
> Bug: webrtc:7306
> Change-Id: I1c9def8c9bcd30da245ecf1894670f4b492547fc
> Reviewed-on: https://webrtc-review.googlesource.com/21980
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20630}

TBR=solenberg@webrtc.org,henrika@webrtc.org

Change-Id: Ie60a44d64df28ec8779305f3c7e3b071100c643f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7306
Reviewed-on: https://webrtc-review.googlesource.com/22200
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20637}
2017-11-10 19:32:20 +00:00
cba3d274da Removes id parameter in ADM factory method
Bug: webrtc:7306
Change-Id: I1c9def8c9bcd30da245ecf1894670f4b492547fc
Reviewed-on: https://webrtc-review.googlesource.com/21980
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20630}
2017-11-10 13:11:05 +00:00
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
919dc2e843 Removes fallback from Linux PulseAudio to ALSA.
Unit test now checks that ADM:Init() works before any test runs.
It means that all tests will be skipped on bots that lack Pulse
support which is as how it worked before this CL as well. But then,
we detected the lack of support by checking that the audio layer had
changed from Pulse to Alsa.

As a consequence, I also decided to inject fake/mock ADMs in more
unit tests. One was actually already injected for other reasons
(see https://codereview.webrtc.org/2997383002/) but it had accidentally
been "reverted" later in combination with other changes.

To summarize: before this change we had a set of unit tests where real
audio was tested but it was not the intention of the test or required.
In addition, some Linux bots (VM:s) did not support PulseAudio and on
them the tests relied on a fallback mechanism to ALSA to work, i.e.,
we had a rather complex dependency on hardware. This dependency has now
been removed and it should result in more stable tests.

Bug: webrtc:7306, webrtc:7806
Change-Id: Ia0485658c04a4ef3b3f2dc0d557d73738067304b
Reviewed-on: https://webrtc-review.googlesource.com/8640
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20307}
2017-10-16 11:49:37 +00:00
b6ca72154f Changes test condition in AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex
TBR=henrik.lundin

Bug: webrtc:7744
Change-Id: I8e8adc666e4734a9333dc3eaa273f68d66a8d0af
Reviewed-on: https://webrtc-review.googlesource.com/7260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20183}
2017-10-06 11:47:14 +00:00
6b3e1a2bbd Fixes issue in ADM on Mac OSX when audio is renegotiated
Moved from https://codereview.webrtc.org/3009093002/

TBR=hlundin-webrtc

Bug: webrtc:8041
Change-Id: I33485629a6f1dcb86fd4242468841605e7d8a72a
Reviewed-on: https://webrtc-review.googlesource.com/3440
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19949}
2017-09-25 15:26:33 +00:00
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00