| 1b40a9a8afe0d7b2244ad8dea19e8222fec3c207 |
|
12-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
RefCountInterface: Make AddRef() and Release() const
This CL makes AddRef() and Release() const member methods and the refcount integer mutable. This is reasonable, because they only manage the lifetime of the object, and this is also how it's done in Chromium.
The purpose is to be able to capture a const pointer in a scoped_refptr, which is currenty impossible. The practial problem this CL solves is this:
void Foo::Bar() const {}
rtc::Callback0<void> Foo::MakeClosure() const {
return rtc::Bind(&Foo::Bar, this);
}
We currently capture |this| as const Foo*. With this CL, |this| will be captured as scoped_refptr<const Foo>.
A test is also added in bind_unittest to check this behaviour.
BUG=webrtc:5065
R=perkj@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1403683004 .
Cr-Commit-Position: refs/heads/master@{#10253}
/external/webrtc/webrtc/modules/audio_device/include/fake_audio_device.h
|
| c14f5ff60fb0c42c97702de112a9e8f1eccba574 |
|
23-Sep-2015 |
henrika <henrika@webrtc.org> |
Improving support for Android Audio Effects in WebRTC.
Now also supports AGC and NS effects and adds the possibility
to override default settings.
R=magjed@webrtc.org, pbos@webrtc.org, sophiechang@chromium.org
TBR=perkj
BUG=NONE
Review URL: https://codereview.webrtc.org/1344563002 .
Cr-Commit-Position: refs/heads/master@{#10030}
/external/webrtc/webrtc/modules/audio_device/include/fake_audio_device.h
|
| 0b1534c52eab79372557a6d81aaf4dd9407f55d3 |
|
15-Dec-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/fake_audio_device.h
|
| a954c07ee1c93175e6ebbeb20517b347474362ae |
|
09-Dec-2014 |
henrika@webrtc.org <henrika@webrtc.org> |
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/fake_audio_device.h
|
| c7c432aa9b8c9f9ba6d41554917784a27b21426a |
|
02-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove AudioDevice::{Microphone,Speaker}IsAvailable.
This was only used for logging, except on Mac, where the methods are
now private.
BUG=3132
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5831 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/fake_audio_device.h
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| b082ade3db118113bba284d0f8fd32901371a2a0 |
|
18-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Hook up audio/video sync to Call.
Adds an end-to-end audio/video sync test.
BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/fake_audio_device.h
|
| f0a90c37c4b8a2581268f0054cc9d977e7beee8e |
|
05-Mar-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Expose the capture-side AudioProcessing object and allow it to be injected.
* Clean up the configuration code, including removing most of the weird defines.
* Add a unit test.
Review URL: https://webrtc-codereview.appspot.com/1152005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3605 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/include/fake_audio_device.h
|