kwiberg
f01633e667
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/
...
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1722083002
Cr-Commit-Position: refs/heads/master@{#11740}
2016-02-24 13:00:45 +00:00
Peter Kasting
6955870806
Convert channel counts to size_t.
...
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org , henrika@webrtc.org , kjellander@webrtc.org , minyue@webrtc.org , perkj@webrtc.org , solenberg@webrtc.org , stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
henrika
c14f5ff60f
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}
2015-09-23 12:09:40 +00:00
Peter Kasting
1380e266ff
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}
2015-08-29 00:31:15 +00:00
Peter Kasting
dce40cf804
Update a ton of audio code to use size_t more correctly and in general reduce
...
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org , pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00
henrika
b26198972c
Adding support for OpenSL ES output in native WebRTC
...
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org , phoglund@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 14:49:04 +00:00
