On 10/09/2012 06:25 PM, Atte André Jensen wrote:
> On 10/09/2012 03:27 AM, Roger wrote:
>> On 10/09/2012 05:01 AM, Atte André Jensen wrote:
>>> "mp3gain optionally writes gain adjustments directly into the encoded
>>> data. In this case, the adjustment works with all mp3 players, i.e. no
>>> support for a special tag is required. This mode is activated by any
>>> of the options -r, -a, -g, or -l."
>>>
>> From http://www.replaygain.org/ :
>> Player requirements
>> "Loudness normalization, pre-amplification and clipping prevention are
>> the operations performed by a ReplayGain player."
>
> Mind that replaygain and mp3gain are two different things.
>
> Replaygain is "a proposed standard ... to measure the perceived
> loudness of audio".
>
> mp3gain "automatically adjusts mp3s so that they all have the same
> volume" and it "uses David Robinson's Replay Gain algorithm to
> calculate how loud the file actually sounds to a human's ears"
>
>> These two sources are confusing. It would be good to get a definitive
>> explanation of what mp3gain does to the audio part of the file. I guess
>> mp3gain uses replaygain method to determine levels and then writes the
>> audio file? This would not require support in the player.
>
> By default mp3gain uses tags, but with the above options (-r, -a, -g,
> or -l), the audio stream is directly modified instead. The great thing
> is that mp3gain is able to change the gain of the audio without
> decoding and re-encoding the mp3-file, which besides being *alot*
> faster, also means it doesn't introduce further loss of quality.
>
>> Vorbisgain does require player support (
>> http://www.sjeng.org/vorbisgain.html ). I assumed that mp3gain worked
>> similarly, but maybe not.
>
> See above. I guess the reason mp3gain have been developed to be able
> to modify the audio and not rely on tags, is the fact that alot of
> players are confused about how to figure out the tags, to the point
> that few actually work with tags, at least with a single tagging method.
>
Thanks for that explanation, Atte. Seems like mp3gain should do what you
want.
I use replay gain applied by SoundKonverter (and previously
foobar2000) and it seems to give good results across a mixture of mp3,
vorbis, flac and m4a files played with Amarok, DeadBeef, Qmmp and
Clementine all of which have Replay Gain support.
Could someone point me in the right direction in setting this up perhaps you know some good tutorials somewhere? I'm not sure exactly whats required do i need to patch the kernel or recompile the kernel is it worth compiling the lastest kernel from kernel.org 3.6.1 or do i modify my current kernel 3.2.0-23-lowlatency? I'm running ubuntu studio 12.04. Sorry i have nil experience with this sort of thing but i'm dead keen to get it working just need a few pointers to get started. Many thanks to Clemens for scripting this hopefully we can make it work.
Please try adding the following entry somewhere in sound/usb/quirks-table.h:
{
USB_DEVICE(0x0499, 0x1507),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
/* .vendor_name = "Yamaha", */
/* .product_name = "THR10", */
.ifnum = QUIRK_ANY_INTERFACE,
.type = QUIRK_COMPOSITE,
.data = (const struct snd_usb_audio_quirk[]) {
{
.ifnum = 1,
.type = QUIRK_AUDIO_STANDARD_INTERFACE
},
{
.ifnum = 2,
.type = QUIRK_AUDIO_STANDARD_INTERFACE
},
{
.ifnum = 3,
.type = QUIRK_MIDI_YAMAHA
},
{
.ifnum = -1
}
}
}
},
Regards,
Clemens
I don't seem to be able to get involved in the music side of things much these
days - too much work and not enough free time :(
However, here is a revision of a very old, lazy chill-out tune of mine. It is
now 100% yoshimi.
http://www.musically.me.uk/music/Dreamer.ogg
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
Hi
I play keyboards in a couple of bands. When setting up my sounds I often
find that on some songs my keyboard is much louder than on others. I'm
pretty sure it's because to the extremely mixed level of the mp3 files I
use for rehearsing and programming the synth sounds.
So my question is: how can I match the levels of the mp3 files so that
they will playback in levels "as close as possible to the level a my
band plays them"? I imagine a ballad would be played back at lower level
than a power rock tune, not sure if there's a way to compensate for
that, though. So at least something that sounds like the same level.
NB: The tunes vary a great deal in audio quality and
loudness-war-compression (some are new dance/pop tunes, some are old
soul tunes, some are something else).
Thanks in advance for any input!
--
Atte
http://atte.dkhttp://modlys.dk
Hi Grekim!
On Thu, Oct 4, 2012 at 5:56 PM, Grekim Jennings
<grekimj(a)acousticrefuge.com> wrote:
> In case you missed it in LAA and have any interest...
>
> I am excited to announce the release of Mixer4 v 1.08 and Waveplayer
> 1.0.
>
> Waveplayer is a lightweight (16 KB in size) console-based .wav player
> using ALSA. It allows you to choose start and stop playback points,
> and repeat playback or editing of those points.
I'm quite interested in waveplayer although I admit I've not tried it
yet as it doesn't sound like it quite fills my needs just yet.
Up until recently my main player has been moc but I've been having
numerous issues with it under Wheezy recently so I've switched to
Audacious but Audacious doesn't auto-detect ALSA/JACK/PA and I don't
really need a X11 UI - curses/text is good enough and preferred by me.
I've tried cmus but didn't like it.
So, is waveplayer going to remain just a wav player or might you
expand it to play wav, FLAC, ogg and MP3 (I need all 4 as a bare
minimum) and hopefully add sound system auto-detection too?
Does anyone know of any existing alternatives to moc that do all this?
I really don't need any other features from a audio player.
Hi all,
Was just wondering if anyone could recommend any resources (preferably
online) for learning about digital filter design? It's something I don't
really know very much about, but I'm interested in learning a little to
improve my understanding of the character of different filters. Thought
this list might be a good place to try asking.
Thanks!
J
Hello,
Is it possible to move tracks up or down in the editor ? Ardour
2.8.12. It is possible to make them thinner or thicker by using the
small handle on the left corner, but is it possible to move them ? I
have tried a few mouse button combinations to no avail. I would use
that for instance, to move down a set of tracks considered as a sketch
while I keep the upper part of the editor for new tracks to be recorded.
Thanks.
I am using an M-Audio Fast Track Pro usb sound card and I am having
some problems using it as a recording device. I suspect it is a
driver issue.
I've been using ALSA's arecord to test audio capture and I've been
issuing the command like so:
arecord -f cd -D hw:1,1 -vv foo.wav
While this seems to do what I want some of the time, I've found that
sometimes the recording will either be heavily distorted or sound as
though it was recorded in 8 bit. Once the problem shows up,
successive takes will always have this distorted quality. It seems
that the only was to temporarily correct the problem is by resetting
the usb device (i.e. turning it off and then back on).
I tried using this device for recording under Linux a couple of years
ago and ran into the exact same problem. I am still eager to get this
working so I figured I would give it another go. I have been looking
online but I haven't found any mention of this problem.
Are there any Fast Track Pro users out there who have used the device
for recording under Linux? Might there be something I'm missing here?
Hi all,
Ubuntu popped up a security update this morning for dbus, which I
installed, then rebooted. Now, audio processing in SuperCollider takes
about twice as much of Jack's duty cycle as it did before. There aren't any
relevant changes in the SC code base between yesterday and today. The only
thing that changed since yesterday was the update.
I'm not sure how to revert, as a test or longer-term measure. I found
"Force version" in synaptic but backing up a version for libdbus-1-3 and
libdbus-1-dev will uninstall a ton of other packages, including jack2
(oops!).
The new dbus version is 1.4.18-ubuntu1.3, previous was 1.1.
It's not catastrophic- I was running well below my CPU's capacity before.
But it's not optimal, so if there's a way to adjust, I'd like to try.
Thanks,
hjh
