Fast(er) transcoding from aac to mp3.

[James Cook]

On Sat, 19 Mar 2011 00:06:23 +0000, 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)
>
>real    2m4.772s
>user    2m2.440s
>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

real    1m49.862s
user    1m47.266s
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%

real    1m19.481s
user    0m0.000s
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

real    1m25.357s
user    1m19.810s
sys     0m0.622s