Commit graph

28 commits

Author SHA1 Message Date
Niels Möller
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
henrika
36b3179312 Removes flaky thread checker in AudioDeviceBuffer.
This CL removes a set of DCHECKs in AudioDeviceBuffer (ADB) where the goal has been
to ensure that some methods are called on one and the same native I/O thread.
The implementation of the ADB is platform independent but the underlying (driving)
audio components differ between platforms. This combination has shown to generate complex
corner cases such as:

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

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

TBR=glaznev

Bug: b/115385789
Change-Id: I681c89adec497a18b97d2a40421c04ea218fd919
Reviewed-on: https://webrtc-review.googlesource.com/100200
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24723}
2018-09-13 11:41:52 +00:00
Niels Möller
8909a63aca Reland "Explicitly wrap main thread in test_main.cc."
This is a reland of 711a31aead

Changes since original landing:

Rename methods only used by tests, mainly via FakeClock,

  MessageQueueManager::ProcessAllMessageQueues
     --> ProcessAllMessageQueuesForTesting

  MessageQueue::IsProcessingMessages
     --> IsProcessingMessagesForTesting

Fix the handling of null rtc::Thread::Current() in
ProcessAllMessageQueuesInternal().

Add override Thread::IsProcessingMessagesForTesting() to return false
for the wrapped main thread, unless it's also the current thread. In
tests, the main thread is typically not processing any messages,
but blocked in an Event::Wait().

Original change's description:
> Explicitly wrap main thread in test_main.cc.
>
> Bug: webrtc:9714
> Change-Id: I6ee234f9a0b88b3656a683f2455c3e4b2acf0d54
> Reviewed-on: https://webrtc-review.googlesource.com/97683
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24560}

Bug: webrtc:9714
Change-Id: I6f022d46aaf1e28f86f09f2d68c1803b69770126
Reviewed-on: https://webrtc-review.googlesource.com/98060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24596}
2018-09-06 08:38:18 +00:00
Anders Carlsson
7bca8ca4e2 Obj-C SDK Cleanup
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.

A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.

The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.

The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.

Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
2018-08-30 10:42:41 +00:00
Jiawei Ou
5f7d00eb3d Release audio unit when ios audio device failed to initialize playout and recording.
TBR=henrika@webrtc.org

Bug: webrtc:9552
Change-Id: I7c3e0c1c2126603e7b1cc412cb37cac57eb3cdbf
Reviewed-on: https://webrtc-review.googlesource.com/90085
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24209}
2018-08-07 14:34:12 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

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

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

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
henrika
79445eadcc Thread checker fails when switching to/from bluetooth headset.
Made some minor changes to resolve the issue. Only affects Debug builds.

NOTRY=TRUE

Bug: webrtc:9310
Change-Id: Ieeeb57d24b559282b2eefd4d8785f7cfe4f44e40
Reviewed-on: https://webrtc-review.googlesource.com/79624
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23434}
2018-05-29 14:50:04 +00:00
henrika
29e865a5d8 Adds stereo support to FineAudioBuffer for mobile platforms.
...continuation of review in https://webrtc-review.googlesource.com/c/src/+/70781

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

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

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

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

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

Also note that, changes in:

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

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

Bug: webrtc:9172
Change-Id: I1ed3798bd1925381d68f0f9492af921f515b9053
Reviewed-on: https://webrtc-review.googlesource.com/71201
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22998}
2018-04-24 11:58:54 +00:00
henrika
8d7393bb28 FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer.
This work is also done as a preparation for adding stereo support to the
FineAudioBuffer.

Review hints:

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

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

The changes have been verified on all affected platforms.

Bug: webrtc:6560
Change-Id: I413af41c43809f61455c45ad383fc4b1c65e1fa1
Reviewed-on: https://webrtc-review.googlesource.com/70781
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22938}
2018-04-19 12:20:28 +00:00
henrika
883d00f7d1 Add support of AAudio in native WebRTC on Android O and above
Bug: webrtc:8914
Change-Id: I016dd8fcebba1644c0a83e5f1460520545d4cdde
Reviewed-on: https://webrtc-review.googlesource.com/56180
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22467}
2018-03-16 10:20:27 +00:00
Niels Möller
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
Karl Wiberg
80ba333fc5 Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix
Bug: chromium:805946
Change-Id: Ibb5dce9af27d0e48c9aee6b0a860b6f62b3c76a0
Reviewed-on: https://webrtc-review.googlesource.com/46145
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21889}
2018-02-05 11:24:59 +00:00
henrika
e6aca637ce Avoids audio crash in combination with invalid audio session on iOS.
Bug: b/70899226
Change-Id: Ie4f92bb1477a29d6b18647e7667f760837a8f1c0
Reviewed-on: https://webrtc-review.googlesource.com/37201
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21522}
2018-01-08 16:29:52 +00:00
henrika
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
Kári Tristan Helgason
86f8047cb7 Remove all code for iOS 8 and below.
Bug: webrtc:8455
Change-Id: I59ae663cea3d734090baa21843e84b8e0ad04c59
Reviewed-on: https://webrtc-review.googlesource.com/16080
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20977}
2017-12-01 14:25:46 +00:00
henrika
fb09eeb8f1 Attempt to resolve crash in AudioDeviceIOS::UpdateAudioDeviceBuffer
Bug: b/69547732
Change-Id: I078175f96d55351ab0318aa2de96f4b859e752ea
Reviewed-on: https://webrtc-review.googlesource.com/24864
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20903}
2017-11-28 09:13:18 +00:00
henrika
32026c3078 Removes Set/GetLoudspeakerStatus APIs from the ADM.
int32_t SetLoudspeakerStatus(bool enable)
int32_t GetLoudspeakerStatus(bool* enabled) const

These APIs are only implemented on iOS and they do not belong in the
native audio layer since the client can achieve the same functionality
by using the shared audio session in sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h.
It also gives the client a better flexibility in how the audio routing is done.

Bug: webrtc:7306
Change-Id: I853e2f57e0f5ae0a0f9fc4729ce961d81f92588b
Reviewed-on: https://webrtc-review.googlesource.com/23740
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20721}
2017-11-16 19:44:24 +00:00
Mirko Bonadei
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
Mirko Bonadei
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
Kári Tristan Helgason
47d3a0197f Reenable some supressed warnings for the objc SDK.
Bug: webrtc:8441
Change-Id: I6b427dfc1fe275e274d042766e0850628cf19994
Reviewed-on: https://webrtc-review.googlesource.com/15000
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20425}
2017-10-25 11:17:36 +00:00
henrika
a642efb6ee Ensures that iOS audio parameters are valid.
Bug: b/62909493
Change-Id: I0f7621f884f7cb9ae9262fb99d2cf33770b31344
Reviewed-on: https://webrtc-review.googlesource.com/14554
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20395}
2017-10-23 14:57:57 +00:00
henrika
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
Fredrik Solenberg
a32dd018eb Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
This is a reland of 34cdd2d402
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
Fredrik Solenberg
d4404c232d Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
This reverts commit 34cdd2d402.

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
Fredrik Solenberg
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
henrika
a86ac6d198 Improves UMA stat for built-in AGC monitoring on iOS
Bug: b/33617347
Change-Id: I27674c1aec7bfe15c2ccaa4b0dd1a0387e7d168a
Reviewed-on: https://webrtc-review.googlesource.com/4063
Reviewed-by: Per Åhgren <peah@google.com>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20046}
2017-09-29 14:05:17 +00:00
Mirko Bonadei
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
Mirko Bonadei
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