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9 Commits

Author SHA1 Message Date
henrika
918b554789 Adds support for OpenSL ES based audio capture on Android. 
NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that
OpenSL ES is not accidentally activated in existing clients. There are still some
unresolved issues to sort out before it can be utilized.

Enables possibility to use OpenSL ES based audio in both directions for WebRTC.
All unit tests and demo clients have been tested with the new implementation but
the new support is behind a flag (see above).

More testing is needed before it can be used in the field and additional support for
hardware effects is still missing.

BUG=webrtc:5925
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2119633004 .

Cr-Commit-Position: refs/heads/master@{#14290}
 2016-09-19 13:44:22 +00:00
henrika
521f7a8db7 Moves ownership of OpenSL engine object to Android audio manager with the goal of adding support for OpenSL ES based audio capture. 
BUG=webrtc:5925

Review-Url: https://codereview.webrtc.org/2019223004
Cr-Commit-Position: refs/heads/master@{#12975}
 2016-05-31 14:03:26 +00:00
henrika
1f0ad1085d Adds support for detection of pro-audio support on Android. 
A new API is added which enables detection of support of pro-audio on
Android. This is part of a larger change and the new API is not used yet.
Most likely it will only be used for logging purposes.

BUG=webrtc:5925

Review-Url: https://codereview.webrtc.org/2015483002
Cr-Commit-Position: refs/heads/master@{#12890}
 2016-05-25 12:15:19 +00:00
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