| 6955870806624479723addfae6dcf5d13968796c |
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13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| 949028fbf1e9a01fb96b186b95606c0096e7d13f |
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15-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Make LevelEstimation not a ProcessingComponent.
BUG=webrtc:5355
Review URL: https://codereview.webrtc.org/1523483002
Cr-Commit-Position: refs/heads/master@{#11033}
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| df3efa8c079294857a8b8e0a02634d06a6d6b6d6 |
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28-Nov-2015 |
peah <peah@webrtc.org> |
Introduced the new locking scheme
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1424663003
Cr-Commit-Position: refs/heads/master@{#10836}
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.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/level_estimator_impl.cc
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| d35a5c350617cc9d60ce45201764a99229b7299a |
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10-Feb-2015 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Make ChannelBuffer aware of frequency bands
Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer.
This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample].
All the files using the ChannelBuffer needed to be re-factored.
Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test.
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36999004
Cr-Commit-Position: refs/heads/master@{#8318}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| a7384a1126cda7ce726f73b023bad997627fc138 |
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03-Dec-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Simplify audio_buffer APIs
Now there is only one API to get the data or the channels (one const and one no const) merged or by band.
The band is passed in as a parameter, instead of calling different methods.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7790 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| 21299d4e00781e199a53ba33ec192cdce920acec |
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14-May-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.
Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc
Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.
BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| 382c0c209d323c1e6972d988a7b26f08fc2e8a6b |
|
05-May-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Allow the RTP level indicator computation to work at any sample rate.
Break out the computation to a separate class, and call directly into
this from channel.cc rather than going through AudioProcessing. This
circumvents AudioProcessing's sample rate limitations.
We now compute the RMS over all samples rather than downmixing to a
single channel. This makes the call point in channel.cc easier, is
more "correct" and should have similar (negligible) complexity.
This caused slight changes in the RMS output, so the ApmTest.Process
reference has been updated. Snippet of the failing output:
[ RUN ] ApmTest.Process
Running test 4 of 12...
Value of: rms_level
Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 5 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 6 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 10 of 12...
Value of: rms_level
Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 11 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 12 of 12...
Value of: rms_level
Actual: 26
Expected: test->rms_level()
Which is: 27
BUG=3290
TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc
produce reasonable printed out results from RMS().
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| 65f933899b815b6c09f8ca7beefeace09ee3ae70 |
|
30-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix constness of AudioBuffer accessors.
Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.
Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| 5964fe0f86a4f33831d1f4994dbde1b42c93bd81 |
|
22-Apr-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
audio_processing: DestroyHandle() now returns void
The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.
BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org, aluebs@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| 56e4a05053d6addc7dbbe2b4d07271305fdbea75 |
|
27-Feb-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove ProcessingComponent's dependence on AudioProcessingImpl.
- Move needed accessors to AudioProcessing.
- Inject the crit directly as a dependency.
- Remove the now unneeded EchoCancellationImplWrapper.
BUG=2894
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5620 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| 7fad4b8c9f1e9a6e3de9962fb74d4953b4f1bb03 |
|
28-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in audio_processing/
BUG=1662
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4116 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
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| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/level_estimator_impl.cc
|