Commit Graph

27 Commits

Author SHA1 Message Date
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
574e69ace1 Removing -Wno-reorder from audio_device.
Bug: webrtc:9251
Change-Id: I5717af1056d432413d448cea0fb562856b364d46
Reviewed-on: https://webrtc-review.googlesource.com/88360
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23948}
2018-07-12 08:37:45 +00:00
5a61967bd0 Removing -Wno-microsoft-extra-qualification.
Bug: webrtc:9251
Change-Id: I67d8c108d6d8b8457d83dffec9d29050e9ef9f62
Reviewed-on: https://webrtc-review.googlesource.com/87846
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23907}
2018-07-10 13:31:48 +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
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
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
cefc46517e RTC_LOG_* macros: Implement argument passing with a single variadic call
Instead of making multiple calls to the std::stringstream << operator,
collect all the arguments and make a single printf-like variadic call
under the hood.

Besides reducing our reliance on iostreams, this makes each RTC_LOG_*
call site smaller; in aggregate, this reduces the size of
libjingle_peerconnection_so.so by 28-32 kB.

A quick benchmark indicates that this change makes log statements
a few percent slower.

Bug: webrtc:8982, webrtc:9185
Change-Id: I3137a4dd8ac510e8d910acccb0c97ce4fffb61c9
Reviewed-on: https://webrtc-review.googlesource.com/75440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23375}
2018-05-23 23:15:04 +00:00
434327376b Don't assume that RTC_LOG's << operator is std::ostream
Bug: webrtc:8982, webrtc:9185
Change-Id: I8a88c10725508f7ea8a7f46e8bcdac4afdb2c617
Reviewed-on: https://webrtc-review.googlesource.com/77681
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23359}
2018-05-23 10:07:20 +00:00
9ab6eb738a Minor namespace change for CoreAudioUtility
NOTRY=TRUE

TBR: kwiberg@webrtc.org
Bug: webrtc:9265
Change-Id: Ic40634eb5258739ef06becd5db7a70a1e31d29e3
Reviewed-on: https://webrtc-review.googlesource.com/78020
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23323}
2018-05-21 12:28:25 +00:00
7e6fcea7de Ports CoreAudioUtil from Chrome to WebRTC.
See https://cs.chromium.org/chromium/src/media/audio/win/core_audio_util_win.h?q=coreaudio&sq=package:chromium&g=0&l=34
for details.

Bug: webrtc:9265
Change-Id: I0fd26620d94a81ccced68d81021c39723a5be2cb
Reviewed-on: https://webrtc-review.googlesource.com/76900
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23298}
2018-05-18 10:25:26 +00:00
52cab50c22 Fixes thread-safety-analysis warnings for Windows ADM.
Now using attribute to ensure that we avoid error like these when bulding with -Wthread-safety-analysis:
error: mutex '_critSect' is still held at the end of function [-Werror,-Wthread-safety-analysis]

RTC_NO_THREAD_SAFETY_ANALYSIS is an attribute on functions or methods, which turns off thread safety
checking for that method. It provides an escape hatch for functions which are either
(1) deliberately thread-unsafe, or
(2) are thread-safe, but too complicated for the analysis to understand.

Bug: webrtc:9202
Change-Id: Ie332bca7eb7eb535ed965de5ddc42872c4f30602
Reviewed-on: https://webrtc-review.googlesource.com/76562
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23221}
2018-05-14 16:56:29 +00:00
7eca805ce3 Removing -Wno-unused-private-field.
This CL is part of the effort to remove warning suppression flags from
the WebRTC build.

Bug: webrtc:9251
Change-Id: I45ece25e897a14a6d4ce8a90ba59688f8fc6fe32
Reviewed-on: https://webrtc-review.googlesource.com/75503
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23187}
2018-05-09 10:55:56 +00:00
03b41483c1 Removing warning suppression -Wno-missing-braces.
Bug: webrtc:9251
Change-Id: Ie32a052738d260364a7543e83e8b46ee3d34df59
Reviewed-on: https://webrtc-review.googlesource.com/75200
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23175}
2018-05-08 13:39:52 +00:00
7149136e72 Lowercase windows includes in win audio device.
Also include rtc_base/win32.h, as windows.h needs to be included before
any other header.

Bug: None
Change-Id: Ib2189f9aaadcf618264677fb65c041b5e85682c3
Reviewed-on: https://webrtc-review.googlesource.com/64846
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22616}
2018-03-27 05:00:40 +00:00
2c599d663d Allow native aec to be used in peerconnection_client if available on windows.
Change-Id: Ia0e2e8b5f755602e58c6be75b7ff57ab1e0528fb
Bug: webrtc:8891
Reviewed-on: https://webrtc-review.googlesource.com/53740
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22032}
2018-02-15 12:03:24 +00:00
1a50cd5894 Remove unused members from AudioDeviceBuffer
Removes current_mic_level_, new_mic_level_ and clock_drift_, together
with APIs for accessing them.

Bug: webrtc:8598
Change-Id: I8e07396fcafd2a719e204730e2c7d26797bed762
Reviewed-on: https://webrtc-review.googlesource.com/39783
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21632}
2018-01-16 10:20:32 +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
dc1e3e8414 Fix excessive audio device logging on Windows
Reverted the logic introduced in https://codereview.webrtc.org/2933953003
This is because the audio device buffer behavior changed with https://codereview.webrtc.org/3009193002, and the RequestPlayoutData method now returns number of samples in each channel, which creates mismatch the reverted CL.

Bug: webrtc:8548
Change-Id: Id4711ca48437ddd3483327c2a4c7827d09e5b770
Reviewed-on: https://webrtc-review.googlesource.com/24122
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20737}
2017-11-17 20:00:31 +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
72c4250cab Formatting some files with LOG macros usage.
In order to create a clean CL to switch to RTC_ prefixed LOG macros
this CL runs `git cl format --full` on the files with LOG macros in
the following directories:
- modules/audio_device
- modules/media_file
- modules/video_capture

This CL has been automatically generated with:

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
  for d in media_file video_capture audio_device; do
    cd modules/$d
    git grep -l $m | grep -E "\.(cc|h|m|mm)$" | xargs sed -i "1 s/$/ /"
    cd ../..
  done
done
git cl format --full

Bug: webrtc:8452
Change-Id: I2858b6928e6bd79957f2e5e0b07028eb68a304b2
Reviewed-on: https://webrtc-review.googlesource.com/21322
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20613}
2017-11-09 09:49:12 +00:00
6592f2cfd2 Removes more unused ADM APIs:
- RecordingDelay()
- LastError()

Bug: webrtc:7306
Change-Id: I3bb9cd243a1464f0ba612787c854eeb6602c7e38
Reviewed-on: https://webrtc-review.googlesource.com/12060
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20326}
2017-10-17 14:23:50 +00:00
a32dd018eb Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
This is a reland of 34cdd2d402b08aee4e17a6fd38c87e0e5cd7aa30
Original change's description:
> Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
> 
> (Re-upload of https://codereview.webrtc.org/3020493002/)
> 
> Bug: webrtc:4690, webrtc:7306
> Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e
> Reviewed-on: https://webrtc-review.googlesource.com/5360
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20083}

Bug: webrtc:4690, webrtc:7306
Change-Id: Ib019439fe6ab0e6b759819e1e9bd320ba1d983bd
Reviewed-on: https://webrtc-review.googlesource.com/6300
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20137}
2017-10-04 11:31:18 +00:00
d4404c232d Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
This reverts commit 34cdd2d402b08aee4e17a6fd38c87e0e5cd7aa30.

Reason for revert: Breaks Chromium

Original change's description:
> Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
> 
> (Re-upload of https://codereview.webrtc.org/3020493002/)
> 
> Bug: webrtc:4690, webrtc:7306
> Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e
> Reviewed-on: https://webrtc-review.googlesource.com/5360
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20083}

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

Change-Id: Iad03cafb7865f5a22394c3d4d1d3ff3e0fccd4ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:4690, webrtc:7306
Reviewed-on: https://webrtc-review.googlesource.com/5402
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20085}
2017-10-02 15:10:04 +00:00
34cdd2d402 Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
(Re-upload of https://codereview.webrtc.org/3020493002/)

Bug: webrtc:4690, webrtc:7306
Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e
Reviewed-on: https://webrtc-review.googlesource.com/5360
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20083}
2017-10-02 15:01:20 +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