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Recent Posts
1
General A/V / Re: Audacity - file formats i can edit ?
Last post by HA-User -
hi ab,

i end up going to audacity for quite a few things, such as clipping the song, and making adjustments to various things.  so i am just gonna call up each song once.  do whatever, and then amplify it to the max.  an album at a time.  and then use foobar to convert to apple lossless mono.

this is a project/hobby that will last a long time.  so i am just gonna have fun doing it a little bit at a time -  thanks for your help
2
General Audio / Re: Getting and running fdkaac and opusenc encoders for dbpoweramp and foobar.
Last post by Octocontrabass -
Is there any improvement in quality by converting 16 bit depth MP3
There is no "improvement in quality" by converting MP3 to anything else.

Is it possible to use intel HD audio and realtek 897 codec during encoding process.
No. Those are only used for input and output. They have no effect on encoding or decoding.
4
General A/V / Re: Audacity - file formats i can edit ?
Last post by HA-User -
on the convert screen, it allows me to do a replay gain.  in the scan mode, it wanted to reduce everything.  but at least it wanted to do SOMETHING - LOL.

and that worked for playing in foobar.  but that is about it, as far as i could tell.  i loaded it into audacity - no change.  i played it on vlc media player - no change.  i then placed it in my itunes library, so that itunes would play it - no change.  no change in volume, except for in foobar.
5
Off-Topic / Re: Mogi Grumbles butchered the album End of Line
Last post by kode54 -
Ah, my bad, they did sort of notice that. Apparently they're making a remaster of the album to fix issues like this, and will release it for free on Bandcamp, like this one was already. I'll grab the remaster for free, because I already gave them some money for the original now.
6
General A/V / Re: Audacity - file formats i can edit ?
Last post by HA-User -
hi ab,

i read thru his explanation.  but it is confusing to me.

first, when you go to preferences, context.  you can click on the replay gain box.  then if you double click that, there are 12 more boxes to either click on or click off

what i want to be able to do is load some files, and apply enough gain to each track, so that it is at the same level.

i think i am at least headed in the right direction.

the first thing i need to do is select and de-select the correct boxes on these first 12 choices.
7
3rd Party Plugins - (fb2k) / Re: Georgia-ReBORN - A Clean foobar2000 Theme
Last post by TT -
@bmorgan,

you could try open your config file "georgia-reborn-config.jsonc" located in your foobar2000/profile/georgia-reborn folder 
and head to the "title_format_strings": { section.

Replace the original date syntax:
Code: [Select]
$if3(%original release date%,%originaldate%,%date%,%fy_upload_date%,)

with this modified one:
Code: [Select]
$if3($cut(%original release date%,4),%originaldate%,%date%,%fy_upload_date%,)

Save the file and restart foobar.

-TT
8
General A/V / Re: Audacity - file formats i can edit ?
Last post by HA-User -
i definitely want the track option.  i have no desire to try and blend an album together.  for me, each song is a song in and of itself.  if i am forced to edit a song for some reason, i will very often raise quiet portions of a song.  i like the loudness level to be very consistent throughout my listening process.

i very seldom would take every song of a cd or album to even listen to, much less try and put some sort of "album" view to a collection of songs.

are there any programs that can raise the gain on a lossless file ?
10
Lossless / Other Codecs / Re: Monkey's Audio - Observations and the need to update some statements on the net?
Last post by kode54 -
PS. I tested Monkey's Audio with FFmpeg ages ago. It required me to add things to my decoder. Basically, the seeking function is inaccurate, I have to read the timestamp/sample of the first decoded packet after seeking, then discard samples.

Recently, I added more code to my FFmpeg handling input, this time, disabling avcodec's start of stream discard feature with (codecCtx->flags2 |= AV_CODEC_FLAG2_SKIP_MANUAL;), then discarding stream->start_time worth of samples at the start, and setting all seek operations to discard that many more samples than the target position. This seems to fix gapless decoding and seeking.

(The built-in skipping only supports skipping up to a single packet worth of audio, and often discards more than that, returning EAGAIN errors, making you think you lost a whole packet. Also, it doesn't adjust the stream/packet PTS according to the skip, so seeking is again inaccurate if you don't skip manually.)