10 Oct
2003
10 Oct
'03
12:50 p.m.
Hello !
I've read somewhere that it is not possible that threads write different data to
/dev/dsp.
But how does Cheesetracker, for example, make it then?
Sascha Retzki
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10 Oct
10 Oct
1:36 p.m.
Hello !
I've read somewhere that it is not possible that threads write different data to /
dev/dsp.
But how does Cheesetracker, for example, make it then?
i don't how cheesetracker does it, but it requires user-space
synchronization (e.g. pthread_mutex_(un)lock). with ALSA, you don't
need to to do this - any thread can write to the device at any
time. that's the theory, at least. in general, its probably a bad
design to be doing that, much like it is for GUI programming, where
its better to have a single thread mediate between the "device" and
the rest of the program.
11 Oct
11 Oct
2:48 p.m.
On Friday 10 October 2003 11:31, hexe_2003(a)directbox.com wrote:
Hello !
I've read somewhere that it is not possible that threads write different
data to /dev/dsp. But how does Cheesetracker, for example, make it then?
In cheesetracker, only one thread writes to the DSP and manages audio
.. the main one is for the GUI. Intercomunication between both is done
via shared memory, unless on destructive operations, where locking is
performed. (lock on the gui, and trylock on the audio thread). This is to
ensure that jack will not kill the app :)
Juan Linietsky
3:24 p.m.
New subject: [linux-audio-dev] ladcca
Juan Linietsky
hi
by some no doubt fated coincidence i was just re-reading your
seminal email (!) which inspired ladcca and myself. Things look
a little quiet on the web site so i was wondering how you felt
about it a year later. I do hope you havnt lost interest!
best regards
--
Tim Orford
8:43 p.m.
New subject: [linux-audio-dev] ladcca
On Saturday 11 October 2003 13:50, Tim Orford wrote:
Hi ! I am not the author of ladcca, Bob Ham is.
I personally am happy about it! :)
Cheers
Juan Linietsky
Juan Linietsky
hi
by some no doubt fated coincidence i was just re-reading your
seminal email (!) which inspired ladcca and myself. Things look
a little quiet on the web site so i was wondering how you felt
about it a year later. I do hope you havnt lost interest!
best regards
12 Oct
12 Oct
2:33 p.m.
New subject: [linux-audio-dev] ladcca
Hi ! I am not the author of ladcca, Bob Ham is.
I personally am happy about it! :)
woops! :-)
I guess Bob is far too modest to put his name on the web page:-)
anyway, having recomended ladcca the other day i was a
little disturbed to see the web page gives the impression
of a dead project. There is no activity since march and in the
middle of the page it says "LADCCA is not particularly stable or usable
at the moment. It is not recommended for use with production systems".
Perhaps a minor cosmetic update would be in order?
as i think ladcca is crucial to the furtherance of a Free modular
audio system, any comments from people on how it is going,
especially with regards to client support would be most welcome.
Bob, i'm assuming you're reading this list, so what help if any is needed?
regards to all
--
Tim Orford
8:28 p.m.
New subject: [linux-audio-dev] ladcca
On Sun, 2003-10-12 at 16:55, Tim Orford wrote:
anyway, having recomended ladcca the other day i was
a
little disturbed to see the web page gives the impression
of a dead project.
It's not really dead, it's just been a while since the last release.
I've been focused (more) on finding a job recently, so lad stuff has
taken a bit of a back seat. I am still plugging away at it tho.
There is no activity since march and in the
middle of the page it says "LADCCA is not particularly stable or usable
at the moment. It is not recommended for use with production systems".
Perhaps a minor cosmetic update would be in order?
Well.. at the moment, it's not really usable for production systems, and
the last release was in March :) I'm not sure what would be updated.
as i think ladcca is crucial to the furtherance of a
Free modular
audio system, any comments from people on how it is going,
especially with regards to client support would be most welcome.
Bob, i'm assuming you're reading this list, so what help if any is needed?
There's a lot of stuff that's changed/changing from the last release;
the server interface protocol is more usable and I'm intending to change
the project file format to XML and do some other cleaning up. The
conditions for the next release are these being finished, and being able
to actually use the system to make some music without any problems. I
wouldn't want to guess at when that'll be though.
I'll put the code in CVS soon and mail the list.
Bob
--
Bob Ham <rah(a)bash.sh>
"At some point, keystroke recorders got installed on several machines at
Valve. Our speculation is that these were done via a buffer overflow in
Outlook's preview pane." -- Gabe Newell on the Half-Life 2 source leak
13 Oct
13 Oct
9:46 p.m.
New subject: [linux-audio-dev] ladcca
On Sun, 2003-10-12 at 23:12, Bob Ham wrote:
I'll put the code in CVS soon and mail the list.
Done.
root: :pserver:anoncvs@rose.clear.bash.sh:/home/cvs/ladcca
password: anoncvs
module: ladcca
Bob
--
Bob Ham <rah(a)bash.sh>
"At some point, keystroke recorders got installed on several machines at
Valve. Our speculation is that these were done via a buffer overflow in
Outlook's preview pane." -- Gabe Newell on the Half-Life 2 source leak
13 Nov
13 Nov
6:50 p.m.
New subject: [linux-audio-dev] question about time and compressed formats
Hi all,
I' ve done some tests on a bunch of songs in different compressed formats (
samplerate = 44100 ): Mp3 and Ogg. For the Mp3 format I tested various
bitrates and I find out that on the playback phase this format has a value of
26.12 milliseconds/frame ( meaning that every frame cover 26.12 ms! ).
For the Ogg format I tested only a nominal bitrate of 192 Kb and I find out a
value of 23.22 milliseconds/frame. These values are taken with an error of
about 10^(-6) seconds.
My ask: is there a general algorythm to calculate the ms/frame value for all
conpressed formats? Could someone confirm my values? Are this values affected
from some parameters ( I think surely samplerate... )? Why different values
for Ogg and Mp3? Ogg will be affected by bitrate?
Thanks to all people!
J_Zar
Gianluca Romanin
14 Nov
14 Nov
10:21 a.m.
New subject: [linux-audio-dev] question about time and compressed formats
On Thursday 13 November 2003 16:48, J_Zar wrote:
I' ve done some tests on a bunch of songs in
different compressed formats
( samplerate = 44100 ): Mp3 and Ogg. For the Mp3 format I tested various
bitrates and I find out that on the playback phase this format has a value
of 26.12 milliseconds/frame ( meaning that every frame cover 26.12 ms! ).
My ask: is there a general algorythm to calculate the
ms/frame value for
all conpressed formats? Could someone confirm my values? Are this values
affected from some parameters ( I think surely samplerate... )? Why
different values for Ogg and Mp3? Ogg will be affected by bitrate?
An MPEG frame always contains 1152 PCM frames, as per the standard. This is
true of Layers One, Two and Three. Thus, the time length of an MPEG frame
would be:
l=1152/fs
where
l = Length of frame in mS
fs = Sample rate in kHz
Thus, for your example case, the calculated length would be:
1152/44.1 = 26.1 mS
thus yielding good agreement with your measured results.
I don't know how it is with Vorbis, but I'd suspect something similar. See:
http://www.xiph.org/
Cheers!
|-------------------------------------------------------------------------|
| Frederick F. Gleason, Jr. | Director of Broadcast Software Development |
| | Salem Radio Labs |
|-------------------------------------------------------------------------|
| Logic is a way to go wrong with confidence. |
| --Robert Heinlein |
| "The Notebooks of Lazarus Long" |
|-------------------------------------------------------------------------|
11:05 a.m.
New subject: [linux-audio-dev] question about time and compressed formats
On Fri, Nov 14, 2003 at 09:20:47AM -0500, Fred Gleason wrote:
FYI, from the vorbis docs:
Vorbis provides none of its own framing, synchronization or protection
against errors; it is solely a method of accepting input audio,
dividing it into individual frames and compressing these frames into
raw, unformatted 'packets'. The decoder then accepts these raw packets
in sequence, decodes them, synthesizes audio frames from them, and
reassembles the frames into a facsimile of the original audio stream.
Vorbis is a free-form variable bit rate (VBR) codec and packets have
no minimum size, maximum size, or fixed/expected size. Packets are
designed that they may be truncated (or padded) and remain decodable;
this is not to be considered an error condition and is used
extensively in bitrate management in peeling. Both the transport
mechanism and decoder must allow that a packet may be any size, or end
before or after packet decode expects.
Vorbis packets are thus intended to be used with a transport mechanism
that provides free-form framing, sync, positioning and error
correction in accordance with these design assumptions, such as Ogg
(for file transport) or RTP (for network multicast).
So, if I'm reading this correctly, the vorbis framesize is not fixed
within the general spec, so you'd have to be prepared for all sorts of
frame sizes.
784 - Michael C. Piantedosi - drclaw(a)dogsolitude.org
On Thursday 13 November 2003 16:48, J_Zar wrote:
> I' ve done some tests on a bunch of songs in different compressed formats
- snip -
different
values for Ogg and Mp3? Ogg will be affected by bitrate?
An MPEG frame always contains 1152 PCM frames, as per the standard. This is
true of Layers One, Two and Three. Thus, the time length of an MPEG frame
would be:
l=1152/fs
where
l = Length of frame in mS
fs = Sample rate in kHz
Thus, for your example case, the calculated length would be:
1152/44.1 = 26.1 mS
thus yielding good agreement with your measured results.
I don't know how it is with Vorbis, but I'd suspect something similar. See:
http://www.xiph.org/
7470
days inactive
7505
days old
linux-audio-dev@lists.linuxaudio.org
10 comments
8 participants
participants (8)
-
Bob Ham -
drclaw＠dogsolitude.org -
Fred Gleason -
hexe_2003＠directbox.com -
J_Zar -
Juan Linietsky -
Paul Davis -
Tim Orford