| 30166cb1a8d7ca2d67981927d056e60aa58cb1ae |
|
07-Jan-2016 |
henrika <henrika@webrtc.org> |
iOS stability improvement for device switching, including BT devices
BUG=webrtc:5058
Review URL: https://codereview.webrtc.org/1554163002
Cr-Commit-Position: refs/heads/master@{#11168}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 46ad5426b025eddac8e9232014d347e73d27180e |
|
07-Dec-2015 |
pbos <pbos@webrtc.org> |
Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
Reason for revert:
Broke downstream compile step, possibly relandable when using a MSVC version that has constexpr, other than that I'm out of ideas.
.../webrtc/base/atomicops.h:71:8: note: no known conversion for argument 1 from '<brace-enclosed initializer list>' to 'const rtc::AtomicInt&'
Original issue's description:
> Reland of "Create rtc::AtomicInt POD struct."
>
> Relands https://codereview.webrtc.org/1420043008/ with brace initializers
> instead of constructors hoping that they won't introduce static
> initializers.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/84f0970d100e67a1dc4fe9a1b16b7d293302044e
> Cr-Commit-Position: refs/heads/master@{#10920}
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1505053002
Cr-Commit-Position: refs/heads/master@{#10922}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 84f0970d100e67a1dc4fe9a1b16b7d293302044e |
|
07-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Reland of "Create rtc::AtomicInt POD struct."
Relands https://codereview.webrtc.org/1420043008/ with brace initializers
instead of constructors hoping that they won't introduce static
initializers.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1498953002 .
Cr-Commit-Position: refs/heads/master@{#10920}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| c729032b1b31e9064a37d3f862bcf60e3651bdff |
|
02-Dec-2015 |
henrika <henrika@webrtc.org> |
Resolves issue with multiple calls to audio unit initialization
BUG=webrtc:5166
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1472833002 .
Cr-Commit-Position: refs/heads/master@{#10865}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 34911ad55c4c4c549fe60e1b4cc127420b15666b |
|
20-Nov-2015 |
henrika <henrika@webrtc.org> |
Improved error handling in iOS ADM to avoid race during init
BUG=webrtc:5166
R=pbos@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1435293003 .
Cr-Commit-Position: refs/heads/master@{#10728}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 5a71f03f8bb0bd1691729717ab7ef22b1c3f94a0 |
|
17-Nov-2015 |
henrika <henrika@webrtc.org> |
Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant
since it is recommended for VoIP apps.
BUG=b/23356406
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1418483004 .
Cr-Commit-Position: refs/heads/master@{#10673}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 3c12f4dadb969afb97468aeb1548777bf860085a |
|
17-Nov-2015 |
pbos <pbos@webrtc.org> |
Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ )
Reason for revert:
Caused static initializers.
BUG=chromium:556866
TBR=tommi@webrtc.org
Original issue's description:
> Create rtc::AtomicInt POD struct.
>
> Prevents accidental non-atomic reads, increments and stores since
> "volatile int" doesn't enforce atomic usage.
>
> BUG=
> R=kwiberg@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/b27f590ece487819c3d1fda400315e582fb975b6
> Cr-Commit-Position: refs/heads/master@{#10657}
TBR=kwiberg@webrtc.org,tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1453093002
Cr-Commit-Position: refs/heads/master@{#10669}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| b27f590ece487819c3d1fda400315e582fb975b6 |
|
16-Nov-2015 |
pbos <pbos@webrtc.org> |
Create rtc::AtomicInt POD struct.
Prevents accidental non-atomic reads, increments and stores since
"volatile int" doesn't enforce atomic usage.
BUG=
R=kwiberg@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1420043008
Cr-Commit-Position: refs/heads/master@{#10657}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 96839648a0eb09476f07791fdf3ff2facffa9a8a |
|
12-Nov-2015 |
henrika <henrika@webrtc.org> |
Trivial initialization fix in AudioDeviceIOS
NOTRY=TRUE
TBR=tkchin
BUG=webrtc:5058
Review URL: https://codereview.webrtc.org/1435003002
Cr-Commit-Position: refs/heads/master@{#10616}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 45c136b579c04c6cbf5f3ea77e7f8171d48a5890 |
|
21-Oct-2015 |
henrika <henrika@webrtc.org> |
Adds support for Bluetooth headsets to the iOS audio layer.
This patch also also ensures that audio is restored after an incoming
GSM call.
BUG=webrtc:5058, webrtc:5012
TEST=Manual tests using modified AppRTCDemo and three different BT headsets
Review URL: https://codereview.webrtc.org/1401963002
Cr-Commit-Position: refs/heads/master@{#10354}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 8c471e7bdfc3bd420d19af118b2bdf8fd716288e |
|
01-Oct-2015 |
henrika <henrika@webrtc.org> |
Objective-C++ style guide changes for iOS ADM
BUG=NONE
Review URL: https://codereview.webrtc.org/1379583002
Cr-Commit-Position: refs/heads/master@{#10135}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 86d907cffda803ee34ee68f9833c1980d1b9f7a6 |
|
07-Sep-2015 |
henrika <henrika@webrtc.org> |
Refactor the AudioDevice for iOS and improve the performance and stability
This CL contains major modifications of the audio output parts for WebRTC on iOS:
- general code cleanup
- improves thread handling (added thread checks, remove critical section, atomic ops etc.)
- reduces loopback latency of iPhone 6 from ~90ms to ~60ms ;-)
- improves selection of audio parameters on iOS
- reduces complexity by removing complex and redundant delay estimates
- now instead uses fixed delay estimates if for some reason the SW EAC must be used
- adds AudioFineBuffer to compensate for differences in native output buffer size and
the 10ms size used by WebRTC. Same class as is used today on Android and we have unit tests for
this class (the old code was buggy and we have several issue reports of crashes related to it)
Similar improvements will be done for the recording sid as well in a separate CL.
I will also add support for 48kHz in an upcoming CL since that will improve Opus performance.
BUG=webrtc:4796,webrtc:4817,webrtc:4954, webrtc:4212
TEST=AppRTC demo and iOS modules_unittests using --gtest_filter=AudioDevice*
R=pbos@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1254883002 .
Cr-Commit-Position: refs/heads/master@{#9875}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 1380e266ff48be9718ce0867cfd65058cb09c5fc |
|
29-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Convert some more things to size_t.
These changes stem from requests by Andrew on https://codereview.webrtc.org/1228823002/ to eliminate some "return -1"s and change to using asserts plus returning size_ts. I then also converted the relevant connected bits.
This also cleans up a bunch of style issues, e.g. no spaces around operators.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, henrik.lundin@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://codereview.webrtc.org/1305983003 .
Cr-Commit-Position: refs/heads/master@{#9813}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 4de6622bccb0e8ba3dff9de26ca69e3adf7a4eba |
|
10-Aug-2015 |
Jiawei Ou <jiawei.ou@gmail.com> |
Fix a bug in computing audio delay on ios device. Converts seconds to
milliseconds by multiplying 1000 instead of dividing 1000.
BUG=
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1265823003 .
Patch from Jiawei Ou <jiawei.ou@gmail.com>.
Cr-Commit-Position: refs/heads/master@{#9693}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 324d9c9a86e2fa331234eb0fa227845fde9f0317 |
|
20-Jul-2015 |
henrika <henrika@webrtc.org> |
Avoids error message about unknown selected data source for Port iPhone Microphone
TBR=tkchin
BUG=webrtc:4845
TEST=modules_unittests
Review URL: https://codereview.webrtc.org/1237233003 .
Cr-Commit-Position: refs/heads/master@{#9602}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| ba35d05a4918b3efa7ab88674781aadb48017ff8 |
|
14-Jul-2015 |
henrika <henrika@webrtc.org> |
Cleanup of iOS AudioDevice implementation
TBR=tkchin
BUG=webrtc:4789
TEST=modules_unittests --gtest_filter=AudioDeviceTest* and AppRTCDemo
Review URL: https://codereview.webrtc.org/1206783002 .
Cr-Commit-Position: refs/heads/master@{#9578}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 9303eaf512dde59d97298f74afdd250f4bc0c347 |
|
27-May-2015 |
Noah Richards <noahric@chromium.org> |
Don't unnecessarily set mode/category on AVAudioSession.
Doing so clears transient properties on the session back to defaults.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52589004
Cr-Commit-Position: refs/heads/master@{#9297}
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 38492c5b6fbb615159fa32b9cc24cd887295573b |
|
22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Re-land 8810 "- Add a SetPriority method to ThreadWr..."
> Revert 8810 "- Add a SetPriority method to ThreadWrapper"
> Seeing if this is causing roll issues.
>
> > - Add a SetPriority method to ThreadWrapper
> > - Remove 'priority' from CreateThread and related member variables from implementations
> > - Make supplying a name for threads, non-optional
> >
> > BUG=
> > R=magjed@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/44729004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/48609004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50459005
Cr-Commit-Position: refs/heads/master@{#8819}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8819 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 90a1cb463092c5189b1a69837731a3395d79f61c |
|
22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8810 "- Add a SetPriority method to ThreadWrapper"
Seeing if this is causing roll issues.
> - Add a SetPriority method to ThreadWrapper
> - Remove 'priority' from CreateThread and related member variables from implementations
> - Make supplying a name for threads, non-optional
>
> BUG=
> R=magjed@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/44729004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48609004
Cr-Commit-Position: refs/heads/master@{#8818}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| b6817d793fa647ec77aaaaf74df82a94e46632bb |
|
20-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
- Add a SetPriority method to ThreadWrapper
- Remove 'priority' from CreateThread and related member variables from implementations
- Make supplying a name for threads, non-optional
BUG=
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44729004
Cr-Commit-Position: refs/heads/master@{#8810}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 361981faa86668cd9b20a2837d0b166fc024cd9b |
|
19-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Use scoped_ptr for ThreadWrapper::CreateThread.
BUG=
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45799004
Cr-Commit-Position: refs/heads/master@{#8794}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8794 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 86639737b83d8877abc4810100e30a8af863189d |
|
13-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove thread id from ThreadWrapper::Start().
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.
BUG=4413
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43699004
Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|
| 122caa51b145f0d28f7b260cdb044631df395eee |
|
15-Jul-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue.
CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.
BUG=3487
R=glaznev@webrtc.org, noahric@chromium.org
Review URL: https://webrtc-codereview.appspot.com/21769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/ios/audio_device_ios.mm
|