Commit Graph

10 Commits

Author SHA1 Message Date
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
ecf6b81644 Pull the Voice Activity Detector out from the AGC
This change generates bit-exact values when running through audioproc_f than before.

This change was originally uploaded here:
* https://codereview.webrtc.org/1181933002/
* https://codereview.webrtc.org/1177043017/

And reverted because of an ASAN problem in Chrome here:
* https://codereview.webrtc.org/1192863006/
* https://codereview.webrtc.org/1194963003/

TBR=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1212543002

Cr-Commit-Position: refs/heads/master@{#9505}
2015-06-25 19:28:55 +00:00
51c7cbb86a Revert "Pull the Voice Activity Detector out from the AGC"
This reverts commit 518c683f3e413523a458a94b533274bd7f29992d.

Breaks Linux-Asan bot
https://uberchromegw.corp.google.com/i/client.webrtc/builders/Linux%20Asan/builds/4348/steps/libjingle_peerconnection_unittest/logs/stdio

BUG=
TBR=aluebs@webrtc.org

Review URL: https://codereview.webrtc.org/1208793002.

Cr-Commit-Position: refs/heads/master@{#9503}
2015-06-25 06:46:14 +00:00
518c683f3e Pull the Voice Activity Detector out from the AGC
This change generates bit-exact values when running through audioproc_f than before.

This change was originally uploaded here:
* https://codereview.webrtc.org/1181933002/
* https://codereview.webrtc.org/1177043017/

And reverted because of an ASAN problem in Chrome here:
* https://codereview.webrtc.org/1192863006/
* https://codereview.webrtc.org/1194963003/

TBR=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1211563003

Cr-Commit-Position: refs/heads/master@{#9502}
2015-06-25 01:46:03 +00:00
f260fc2136 Revert "Pull the Voice Activity Detector out from the AGC"
This reverts commit 34be126c1b3ee60ecdb86b1de41a0648347450b2.

It breaks Chromium ASAN.

TBR=niklas.enbom@webrtc.org

Review URL: https://codereview.webrtc.org/1192863006.

Cr-Commit-Position: refs/heads/master@{#9472}
2015-06-19 18:24:01 +00:00
34be126c1b Pull the Voice Activity Detector out from the AGC
This change generates bit-exact values when running through audioproc_f than before.

R=andrew@webrtc.org, bloch@google.com

Review URL: https://codereview.webrtc.org/1181933002.

Cr-Commit-Position: refs/heads/master@{#9465}
2015-06-18 19:34:00 +00:00
adc46c4cf7 audio_processing/agc: Adds config to set minimum microphone volume at startup
The AGC is currently bumping up the mic volume to 33% at startup if it is below that level. This is to avoid getting stuck in a poor state from which the AGC can not move, simply a too low input audio level. For some users, 33% is instead too loud.

This CL gives the user the possibility to set that level at create time.
- Extends the Config ExperimentalAgc with a startup_mic_volume for the user to set if desired. Note that the bump up does not apply to the legacy AGC and the "regular" AGC is controlled by ExperimentalAgc.
- Without any actions, the same default value as previously is used.
- In addition I removed a return value from InitializeExperimentalAgc() and InitializeTransient()

This has been tested by building Chromium on Mac and verify through apprtc that
1) startup_mic_volume = 128 bumps up to 50%.
2) startup_mic_volume = 500 (out of range) bumps up to 100%.
3) startup_mic_volume = 0 bumps up to 4%, the AGC min level.

BUG=4529
TESTED=locally
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43109004

Cr-Commit-Position: refs/heads/master@{#9004}
2015-04-15 09:42:35 +00:00
2ebfac5649 Remove COMPILE_ASSERT and use static_assert everywhere
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.

R=aluebs@webrtc.org, andrew@webrtc.org, hellner@chromium.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 10:51:54 +00:00
3df38b442f Unify the two copies of compile_assert.h
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.

R=aluebs@webrtc.org, andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 11:37:48 +00:00
788acd17ad Merge audio_processing changes.
R=aluebs@webrtc.org, bjornv@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00