Re: Fast(er) transcoding from aac to mp3.

[James Cook]

On Sat, 19 Mar 2011 00:06:23 +, you wrote:

Looking back at my script I wrote back at Christmas when the iphone 
option stopped working and my programs started coming down in aac I seem 
to have a faster solution which works for me;

time( ffmpeg -i You_and_Yours_-_16_03_2011_b00zf33w_default.aac 
-map_meta_data 
You_and_Yours_-_16_03_2011_b00zf33w_default.mp3:You_and_Yours_-_16_03_2011_b00zf33w_default.aac
 
You_and_Yours_-_16_03_2011_b00zf33w_default.mp3)

real2m4.772s
user2m2.440s
sys 0m1.580s


it is faster than std ffmpeg + lame b 128
but the output file is half the size (bitrate is 64 instead of 128
kb/s)

using lame -b 64 is 6 secs slower than your method - 8%?
(encoding a 21 minute aac)

JC


Std:
--
FFmpeg version git-c9e16a9-Sherpya, Copyright (c) 2000-2011 the FFmpeg
developer
LAME 3.98.2 32bits (http://www.mp3dev.org/)
Using polyphase lowpass filter, transition band: 16538 Hz - 17071 Hz
Encoding stdin
  to
Adam_Dalgliesh_-_A_Taste_for_Death_-_Episode_5_b00tb9n1_default.partial
.aac.ffmpeg.new.mp3
Encoding as 44.1 kHz j-stereo MPEG-1 Layer III (11x) 128 kbps qval=3

real1m49.862s
user1m47.266s
sys 0m0.497s
--

Without lame pipe:
--
FFmpeg version git-c9e16a9-Sherpya, Copyright (c) 2000-2011 the FFmpeg
developer
:
Output #0, mp3, to
'Adam_Dalgliesh_-_A_Taste_for_Death_-_Episode_5_b00tb9n1_defa
ult.partial.ffmpeg.new.wolame.mp3':
  Metadata:
TSSE: Lavf52.95.0
Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, s16, 64 kb/s
Stream mapping:
  Stream #0.0 - #0.0
Press [q] to stop encoding
size=9845kB time=1260.09 bitrate=  64.0kbits/s
video:0kB audio:9844kB global headers:0kB muxing overhead 0.000327%

real1m19.481s
user0m0.000s
sys 0m0.062s
--

With lame -b 64 i get 
:
LAME 3.98.2 32bits (http://www.mp3dev.org/)
Resampling:  input 44.1 kHz  output 24 kHz
Using polyphase lowpass filter, transition band: 10935 Hz - 11226 Hz
Encoding stdin
  to
Adam_Dalgliesh_-_A_Taste_for_Death_-_Episode_5_b00tb9n1_default.partial
.aac.ffmpeg.new.mp3
Encoding as 24 kHz j-stereo MPEG-2 Layer III (12x)  64 kbps qval=3

real1m25.357s
user1m19.810s
sys 0m0.622s

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Re: Audo test mp4 file

[Andy Bircumshaw]

On 18/3/2011, at 5:56pm, richard wrote:
 bat guano wrote:
 
 This is a file in aac format:-
 http://www.mediafire.com/?hr5lecihn3e1vij
 
 This is the file in m4a format using Nick's experimental update:-
 http://www.mediafire.com/?zdjl0drgowzncf2
 
 Neither of those media files work on my hi-fi cd player (Marantz CD6003
 - released 2009) due to header errors. The cd player supports playback
 of mp3, wma, wav, and aac files( aac files with a m4a extension) ...

Have you played any other m4a files on the Marantz?

If you've ripped these yourself, can you tell us what application you used to 
produce them, please.

Do you have a sample aac m4a file that works on this player that we can have a 
look at, please.

TIA,

aB.
 
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Re: Fast(er) transcoding from aac to mp3.

[Charles Johnson]

dinkypumpkin wrote:

I don't remember if it's the same in Lucid, but in Ubuntu Maverick I just 
needed to install libavcodec-extra-52 (which replaces libavcodec-52 installed 
by default with ffmpeg) for MP3 support.
  

Hmm, i'll look into that. Maybe i can shoehorn it into Lenny. Thanks

CJ

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Re: Fast(er) transcoding from aac to mp3.

[Charles Johnson]

dinkypumpkin wrote:
Oops, forget that.  I see that you're looking for a statically-linked ffmpeg.  

  
Well that's only due to Debian problems ('oldness' and licensing issues 
with mp3).
I notice that i've got libavcodec51 installed, but it's done me no good 
at all.


CJ

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