| 82ccfcf5cae798d21881e41a7123e9ca3016988a |
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14-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Remove unused and rarely used LOG_ macros.
BUG=
Review URL: https://codereview.webrtc.org/1522053002
Cr-Commit-Position: refs/heads/master@{#11014}
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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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_processing/agc/agc_manager_direct.cc
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| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
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28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| dce40cf804019a9898b6ab8d8262466b697c56e0 |
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24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| ecf6b81644af9823dbff5c24a3d5b9bb596c0d5b |
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25-Jun-2015 |
aluebs <aluebs@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| 51c7cbb86ae13bc0c005fc8d14973f3e58d6ff4c |
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25-Jun-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| 518c683f3e413523a458a94b533274bd7f29992d |
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25-Jun-2015 |
aluebs <aluebs@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| f260fc21360ac5f599e64ce9ea2ea616909eca44 |
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19-Jun-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| 34be126c1b3ee60ecdb86b1de41a0648347450b2 |
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18-Jun-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| adc46c4cf7dd8a015b0757c99787d80525c123ab |
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15-Apr-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| 2ebfac5649a5e48fbbc501b42a4336ff979c03e6 |
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14-Jan-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| 3df38b442f6ba29722049b4c4d7121053003a1f8 |
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13-Jan-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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| 788acd17adf6b3d605b5ea66cf394eb81fc086a9 |
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15-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
|