8 Nov
2010
8 Nov
'10
6:32 p.m.
Le Mon, 8 Nov 2010 12:23:12 +1100,
Erik de Castro Lopo <mle+la(a)mega-nerd.com> a écrit :
Again, for what I understand, libsoundfile is used into
input/sndfile/sndfile_engine.c
It is an input plugin. They are other input plugins:
audiofile, cdda, flac, mad, mikmod, mpg123, vorbis and wav.
Well, it is no active developer at that time. It was written by Andy Lo A Foe,
his last working commit is from December 2003. After that, this project has
been dead until I take over its administration in January 2007.
I applied important patches they was sleeping on the mailing list, and done a
new release the 1 February.
After this release, madej was doing a very great job with a new gtk2
interface in 2007, but he was not willing to incorporate the team.
It is some other developers, but now one is active for now. It was a discussion
into 2007 about sockets for AP, but no work was done in that direction. Now, as
I see the current state of AP, it seam more important to me to first fix the
current issues, like the resampling quality and a more up-to-date jack code.
After that, I am open to anything that will be a good move.
The core of AP was not modified from 2003, at the exception of bugfixes. For
this release, it is more bugfixes that peoples external to the AP team send
me. That can explain why the internal audio format is still using integer
instead of float.
It will be very great if someone was willing to share its knowledge and improve
this player. It have some outstanding features like the ability to do varispeed
in a linear fashion from 10 times backwards to 10 time forward. Mplayer do have
a similar function.
I was talking about libsamplerate because, according to Fons, this lib can do
the resampling and the varispeed. I don't know with libsndfile.
Dominique
Dominique Michel wrote:
I am not sure, I can only understand very obvious things in c and c++.
It look like to be into app/AlsaNode.cpp and the different output plugins (into
output/*). For what I understand, the output plugins are just sending the
sampling rate to AlsaNode.cpp, but after that, I am sure of anything.
Fons seam to know the internal of the AlsaPlayer.
While it's a nice player, it has some serious
audio quality
issues.
- Resampling 44.1 -> 48 kHz (for jack) sounds horrible...
Which resampler are you using? - The sndfile input plugin reduces everything to
16 bits.
This is really absurd, even if your files and your
sound card are 24 bit you only get 16.
Floating point wav files apparently aren't read at all
(they load but produce silence when played).
This is not a problem with libsndfile itself, but rather
with the way you are using libsndfile. All of this could be solved by using a good
resampler
lib, and making the internal format floating point
rather than short.
There is little reason not to use 32 bit float as an
internal format, even on things like netbooks and
tablets.
I am just an admin and don't have the
knowledge to make the needed code.
Anybody that can contribute such a great feature for AP will be welcomed. If
you are interested, please take contact with me, on this list or privately.
Who are the developers of this project or has it been
abandoned?
Erik
--
"We have the heroes we deserve."