[FFmpeg-user] volumedetect audio filter result
jacky
renaux.jacky at orange.fr
Mon Jan 7 15:15:43 CET 2013
Hi
Greetings to all of you
I am using from few days the volumedetect filter which help to normalize
sounds
based on one short FLV sample file
my present command line is
**** ffmpeg -f lavfi -i amovie=%dest_file%,volumedetect -f null -y
%gain_log% 2> %gain_log%
the respons is
ffmpeg version N-47062-g26c531c Copyright (c) 2000-2012 the FFmpeg
developers
built on Nov 25 2012 12:21:26 with gcc 4.7.2 (GCC)
configuration: --enable-gpl --enable-version3 --disable-pthreads
--enable-runtime-cpudetect --enable-avisynth --enable-bzlib
--enable-frei0r --enable-libass --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libfreetype --enable-libgsm
--enable-libmp3lame --enable-libnut --enable-libopenjpeg
--enable-libopus --enable-librtmp --enable-libschroedinger
--enable-libspeex --enable-libtheora --enable-libutvideo
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid
--enable-zlib
libavutil 52. 9.100 / 52. 9.100
libavcodec 54. 77.100 / 54. 77.100
libavformat 54. 37.100 / 54. 37.100
libavdevice 54. 3.100 / 54. 3.100
libavfilter 3. 23.102 / 3. 23.102
libswscale 2. 1.102 / 2. 1.102
libswresample 0. 17.101 / 0. 17.101
libpostproc 52. 2.100 / 52. 2.100
[lavfi @ 023dcdc0] Estimating duration from bitrate, this may be inaccurate
Input #0, lavfi, from 'amovie=h264_aac_main_48k.flv,volumedetect':
Duration: N/A, start: 0.007000, bitrate: 352 kb/s
Stream #0:0: Audio: pcm_s16le, 22050 Hz, mono, s16, 352 kb/s
Output #0, null, to 'gain_263_aac_48k.txt':
Metadata:
encoder : Lavf54.37.100
Stream #0:0: Audio: pcm_s16le, 22050 Hz, mono, s16, 352 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
Press [q] to stop, [?] for help
size= 0kB time=00:00:50.10 bitrate= 0.0kbits/s
**********************many others lines without more details
**********************end of report file
video:0kB audio:2158kB subtitle:0 global headers:0kB muxing overhead
-100.000000%
[Parsed_volumedetect_1 @ 023df480] n_samples: 1104896
[Parsed_volumedetect_1 @ 023df480] mean_volume: -30.4 dB
[Parsed_volumedetect_1 @ 023df480] max_volume: -10.0 dB
[Parsed_volumedetect_1 @ 023df480] histogram_9db: 1
[Parsed_volumedetect_1 @ 023df480] histogram_10db: 9
[Parsed_volumedetect_1 @ 023df480] histogram_11db: 19
[Parsed_volumedetect_1 @ 023df480] histogram_12db: 85
[Parsed_volumedetect_1 @ 023df480] histogram_13db: 175
[Parsed_volumedetect_1 @ 023df480] histogram_14db: 436
[Parsed_volumedetect_1 @ 023df480] histogram_15db: 810
questions
1- is there a way to find out where these peaks are located (time) ?
2- there are different volume references between lame (replygain value)
and volumedetect
what is the reference level 24dB or 27dB other one value ?
3- is this reference level fonction of the -tune option fixed during
encode ?
many thanks
jacky
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