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i have a problem with this command( FFMPEG PHP ):

Command:

ffmpeg -i 62f76f050494f0ed6a5997967c00c0c0.wmv -ss 0 -t 99 -y -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 -f flv 62f76f050494f0ed6a5997967c00c0c0.flv

Output:

FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
  configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
  libavutil     50.15. 1 / 50.15. 1
  libavcodec    52.72. 2 / 52.72. 2
  libavformat   52.64. 2 / 52.64. 2
  libavdevice   52. 2. 0 / 52. 2. 0
  libavfilter    1.19. 0 /  1.19. 0
  libswscale     0.11. 0 /  0.11. 0
  libpostproc   51. 2. 0 / 51. 2. 0
[asf @ 0xe81670]max_analyze_duration reached
Input #0, asf, from '/var/www/resources/tmp/62f76f050494f0ed6a5997967c00c0c0.wmv':
  Metadata:
    WMFSDKVersion   : 12.0.7601.17514
    WMFSDKNeeded    : 0.0.0.0000
    IsVBR           : 0
  Duration: 00:00:50.87, bitrate: 2467 kb/s
    Stream #0.0: Audio: wmapro, 44100 Hz, stereo, flt, 256 kb/s
    Stream #0.1: Video: vc1, yuv420p, 950x460 [PAR 1:1 DAR 95:46], 25 fps, 25 tbr, 1k tbn, 25 tbc
Output #0, flv, to '/var/www/resources/media/62f76f050494f0ed6a5997967c00c0c0.flv':
  Metadata:
    encoder         : Lavf52.64.2
    Stream #0.0: Video: flv, yuv420p, 950x460 [PAR 1:1 DAR 95:46], q=2-31, 200 kb/s, 1k tbn, 29.97 tbc
    Stream #0.1: Audio: libmp3lame, 11025 Hz, stereo, s16, 64 kb/s
Stream mapping:
  Stream #0.1 -> #0.0
  Stream #0.0 -> #0.1
Press [q] to stop encoding
frame=   72 fps=  0 q=5.0 size=       0kB time=10.91 bitrate=   0.0kbits/s    
Multiple frames in a packet from stream 0
Warning, using s16 intermediate sample format for resampling
frame=  141 fps=139 q=5.0 size=     103kB time=8.15 bitrate= 103.2kbits/s    
frame=  220 fps=144 q=5.0 size=     875kB time=10.92 bitrate= 656.6kbits/s    
frame=  290 fps=143 q=5.0 size=    1525kB time=13.74 bitrate= 909.1kbits/s    
frame=  356 fps=141 q=5.0 size=    2153kB time=15.99 bitrate=1103.1kbits/s    
frame=  427 fps=141 q=5.0 size=    2847kB time=18.70 bitrate=1247.0kbits/s    
frame=  497 fps=141 q=5.0 size=    3771kB time=21.16 bitrate=1460.0kbits/s    
frame=  575 fps=142 q=5.0 size=    4695kB time=24.61 bitrate=1563.0kbits/s    
frame=  639 fps=141 q=5.0 size=    5301kB time=26.80 bitrate=1620.2kbits/s    
frame=  703 fps=139 q=5.0 size=    5829kB time=29.36 bitrate=1626.2kbits/s    
frame=  774 fps=139 q=5.0 size=    6659kB time=32.39 bitrate=1684.0kbits/s    
frame=  842 fps=139 q=5.0 size=    7915kB time=35.27 bitrate=1838.6kbits/s    
frame=  911 fps=139 q=5.0 size=    9011kB time=37.98 bitrate=1943.4kbits/s    
frame=  975 fps=138 q=5.0 size=    9788kB time=40.59 bitrate=1975.3kbits/s    
frame= 1041 fps=138 q=5.0 size=   10904kB time=43.83 bitrate=2037.9kbits/s    
frame= 1115 fps=138 q=5.0 size=   11795kB time=46.24 bitrate=2089.8kbits/s    
frame= 1183 fps=138 q=5.0 size=   12678kB time=48.74 bitrate=2130.7kbits/s    
frame= 1247 fps=137 q=5.0 size=   13964kB time=51.36 bitrate=2227.5kbits/s    
frame= 1271 fps=136 q=5.0 Lsize=   15865kB time=58.86 bitrate=2208.1kbits/s
video:15366kB audio:462kB global headers:0kB muxing overhead 0.238956%

Problem:

Warning, using s16 intermediate sample format for resampling

I've also tried changing the parameter

From

-ar 44100

to

-ar 11025

Thanks!

Solution:

Read this link:

http://en.wikipedia.org/wiki/MP3#Bit_rate

share|improve this question

migrated from stackoverflow.com Oct 4 '12 at 8:17

This question came from our site for professional and enthusiast programmers.

1  
What is the problem? If you want 11.025kHz from a 44.1kHz original, you need to resample. The warning is not a problem... it is just alerting you that the audio will be decoded to its raw uncompressed form prior to resampling. That is to be expected. –  Brad Oct 4 '12 at 4:53
1  
Next to what Brad comments, I also have a problem to see ffmpeg-php realted to this. You should add some more information to make your question (warning report) some actual question. –  hakre Oct 4 '12 at 7:52
1  
Tip, not necessarily related to the "question": Don't use an FFmpeg version from 2010. Download the latest static build from the homepage or compile it yourself, then try again. –  slhck Oct 4 '12 at 10:56
    
Thanks guys! i read this link: en.wikipedia.org/wiki/MP3#Bit_rate –  Olaf Erlandsen Oct 5 '12 at 3:16
1  
Please post an answer as an answer - with all the essential bits inside.*Not* edit the answer into a question. Its simply not how the SE family of sites works. –  Journeyman Geek Oct 5 '12 at 3:25

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