[FFmpeg-user] mov [ProRes/pcm_s24le] -> mp4 same quality

[HallMarc Websites]

> The this is at least _very_ unusual and AFAIK the first time someone has
> reported this to this list. My personal tests support what carl eugen and
> countless other competent people have said on this list. So either you go
to
> the trouble of building a reproducible test case that you can share
without an
> NDA or you are probably on your own and you should at least stop being
> rude to people who to the best of their (undisputed) knowledge are trying
to
> help you with a patience I don't think anyone could ask them to have. You
> are a beginner and it's fine to ask questions but you attitude is like
pissing on
> the table of the family who has just invited you for a free lunch and then
> insulting the cook without ever having cooked yourself in your entire
life.
> 
> Peace
> 
And now for the final say, I never argued with anyone about what the current
release of ffmpeg is supposed to do, what I have asked for through this list
is to point me in the right direction, tell me why my audio is sounding so
muddy when it shouldn't. That's it. A very simple and basic question and it
took a Multimedia Engineer from another company to show me the missing flag
from my command line!!!
 
I'm done beating my head against this brick wall of a list. Maybe the people
like Carl are just incapable of helping when it comes to the basics. Maybe
they are only truly helpful if it is a complicated query. I don't know. What
I do know is that I now have my answer that NO ONE HERE provided and that
what I did get from this list was reprimands and hints that I should hand
over private client files or get lost. The person who finally helped me
didn't need the file. They looked into the #faac --long-help (no one
mentioned this, either) and from there was able to see the flag that needed
to be included. -aq 128. I know it doesn't need to be set quite this high
but he prefers powers of 2. w/e 

This person not only provided the answer but gave me a decent explanation as
well. Funny how he could do what the devs that actually wrote ffmpeg just
simply didn't. 

"other competent people" While I have no doubt that the folks that frequent
this list may be competent and I know that I have a good amount of respect
for anyone that can work with compression algorithms it still bewilders me
that I couldn't get the answer I was looking for. You can claim anything you
wish, actions do the speaking.

" but you attitude is like pissing on the table of the family who has just
invited you for a free lunch and then insulting the cook without ever having
cooked yourself in your entire life" - really? My frustration at not being
able to get a meaningful answer from this list is fuel for an attack on me?
Good grief... 

If any of you feel I was being rude, I apologize. Was never my intent. My
intent has been to learn from this list not to make enemies. 

Now, for the knowledge dump:
# faac --long-help 

Freeware Advanced Audio Coder 
FAAC 1.28 

Usage: faac [options] infiles ... 

Quality-related options: 
-q <quality> Set default variable bitrate (VBR) quantizer quality in
percent. 
(default: 100, averages at approx. 120 kbps VBR for a normal 
stereo input file with 16 bit and 44.1 kHz sample rate; max. 
value 500, min. 10). 
-b <bitrate> Set average bitrate (ABR) to approximately <bitrate> kbps. 
(max. value 152 kbps/stereo with a 16 kHz cutoff, can be raised 
with a higher -c setting). 
-c <freq> Set the bandwidth in Hz (default: automatic, i.e. adapts 
maximum value to input sample rate). 

Input/output options: 
- <stdin/stdout>: If you simply use a hyphen/minus sign instead 
of an input file name, FAAC can encode directly from stdin, 
thus enabling piping from other applications and utilities. The 
same works for stdout as well, so FAAC can pipe its output to 
other apps such as a server. 
-o X Set output file to X (only for one input file) 
only for one input file; you can use *.aac, *.mp4, *.m4a or 
*.m4b as file extension, and the file format will be set 
automatically to ADTS or MP4). 
-P Raw PCM input mode (default: off, i.e. expecting a WAV header; 
necessary for input files or bitstreams without a header; using 
only -P assumes the default values for -R, -B and -C in the 
input file). 
-R Raw PCM input sample rate in Hz (default: 44100 Hz, max. 96 kHz) 
-B Raw PCM input sample size (default: 16, also possible 8, 24, 32 
bit fixed or float input). 
-C Raw PCM input channels (default: 2, max. 33 + 1 LFE). 
-X Raw PCM swap input bytes (default: bigendian). 
-I <C[,LFE]> Input multichannel configuration (default: 3,4 which means 
Center is third and LFE is fourth like in 5.1 WAV, so you only 
have to specify a different position of these two mono channels 
in your multichannel input files if they haven't been reordered 
already). 

MP4 specific options: 
MP4 support unavailable. 

Expert options, only for testing purposes: 
--tns Enable coding of TNS, temporal noise shaping. 
--no-midside Don't use mid/side coding. 
--mpeg-vers X Force AAC MPEG version, X can be 2 or 4 
--obj-type X AAC object type. (LC (Low Complexity, default), Main or LTP 
(Long Term Prediction) 
--shortctl X Enforce block type (0 = both (default); 1 = no short; 2 = no 
long). 
-r Generate raw AAC bitstream (i.e. without any headers). 
Not advised!!!, RAW AAC files are practically useless!!! 

Documentation: 
--license Show the FAAC license. 
--help Show this abbreviated help. 
--long-help Show complete help. 

More tips can be found in the audiocoding.com Knowledge Base at 
<http://www.audiocoding.com/wiki/>

As for the -aq setting; FAAC is a variable bit rate codec so the 128 doesn't
translate directly. The range is 0-100 with 100 reported as capable of
getting 120kbps average.