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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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
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