On Dec 21, 2007 10:58 PM, Carlos Hernandez <[EMAIL PROTECTED]> wrote:
> I know... and that's the reason why I can specify that, using the asound.conf.
>
aplay and arecord are not aware of those parameters in asound.conf.
They just see a stream of 1s and 0s. You have to tell them how to
interpret
On Sat, 2007-12-22 at 16:27 +1000, Mark Constable wrote:
> On Saturday 22 December 2007 14:41:39 Bill Unruh wrote:
> > > Why couldn't the arecord defaults be what would
> > > most likely be the most common settings of 16 bit,
> > > Rate 44,100 Hz, Stereo ?
> >
> > Because then your counterpart wou
On Dec 21, 2007 10:54 PM, Mark Constable <[EMAIL PROTECTED]> wrote:
> On 22 December 2007 13:24, Lee Revell wrote:
> > > [EMAIL PROTECTED]:~$ arecord -D plugfile out.wav
> > > Recording WAVE 'out.wav' :
> > > Unsigned 8 bit, Rate 8000 Hz, Mono
> >
> > arecord does not do any kind of format detectio
On Saturday 22 December 2007 14:41:39 Bill Unruh wrote:
> > Why couldn't the arecord defaults be what would
> > most likely be the most common settings of 16 bit,
> > Rate 44,100 Hz, Stereo ?
>
> Because then your counterpart would complain that since his sound card was
> 48KHz, 24 bit, why did th
Last version come with Alsa support (configure with --enable-mp4live-alsa)
ALSAaudio_alsa_source.cpp
in fact I've used Alsa, and it works perfect using "default" device.
But file device doesnt :(
On 22/12/2007, Bill Unruh <[EMAIL PROTECTED]> wrote:
> On Sat, 22 Dec 2007, Carlos Hernandez wr
On Sat, 22 Dec 2007, Carlos Hernandez wrote:
> well... I really want to use another application(mp4live)... not arecord.
>
> So this application just ask me for : "sample rate", "channels" and
> what Alsa device to use.
Except that mp4live does not use alsa, it uses oss if I read the data right
well... I really want to use another application(mp4live)... not arecord.
So this application just ask me for : "sample rate", "channels" and
what Alsa device to use.
I've tell it "plugfile" device, 44100 as sample rate, and stereo(2
channels). As you can see I cannot specify the format.
When I
On Sat, 22 Dec 2007, Carlos Hernandez wrote:
> I know... and that's the reason why I can specify that, using the asound.conf.
>
> The source audio file is in 16 bit, 44100, stereo... but when I
> record with arecord, it does with 8bit, 8k, mono.
>
> So I must change this before arecord "record" t
On Sat, 22 Dec 2007, Mark Constable wrote:
> On 22 December 2007 13:24, Lee Revell wrote:
>>> [EMAIL PROTECTED]:~$ arecord -D plugfile out.wav
>>> Recording WAVE 'out.wav' :
>>> Unsigned 8 bit, Rate 8000 Hz, Mono
>>
>> arecord does not do any kind of format detection
>> on its input. It records w
I know... and that's the reason why I can specify that, using the asound.conf.
The source audio file is in 16 bit, 44100, stereo... but when I
record with arecord, it does with 8bit, 8k, mono.
So I must change this before arecord "record" the audio... where? .. I
suppose this should be in asound
On 22 December 2007 13:24, Lee Revell wrote:
> > [EMAIL PROTECTED]:~$ arecord -D plugfile out.wav
> > Recording WAVE 'out.wav' :
> > Unsigned 8 bit, Rate 8000 Hz, Mono
>
> arecord does not do any kind of format detection
> on its input. It records with whatever format
> you tell it to (in this c
On Dec 21, 2007 9:39 PM, Carlos Hernandez <[EMAIL PROTECTED]> wrote:
> Here the "format", "rate" and "channels" are set up, and these are the
> same as the original audio file. Then I suppose I would not have to
> specify these parameters with arecord.
> running arecord:
>
> [EMAIL PROTECTED]:~$ a
Hi list
I've asked help before about file plugin, and it does work fine.
But I need something more. The raw file from which I'm reading audio is:
Signed 16 bit Little Endian, 44100 hz, 2 channels.
I've configured asound.conf this way:
pcm.plugfile{
type plug
slave {
> For the older kernel, init_utsname() is defined for compatibility (see
> alsa-driver/acore/info_oss.c). Maybe that's the problem. RH likely
> defines an incompatible version of init_utsname() and it conflicts.
ok, I edited acore/info_oss.c and changed
#if LINUX_VERSION_CODE < KERNEL_VERSIO
At Thu, 20 Dec 2007 12:43:21 +,
Neil Bird wrote:
>
>
>Of late on my main [Fedora 7] box (I don't know for how long), pretty
> much any ALSA-based app. (audacious, xmms, etc.) gives the following error:
>
> ALSA lib pcm.c:2106:(snd_pcm_open_conf) Cannot open shared library
> /usr/lib/al
At Fri, 21 Dec 2007 00:04:59 -0500,
Lee Revell wrote:
>
> On Dec 20, 2007 9:31 PM, Bill Unruh <[EMAIL PROTECTED]> wrote:
> > On Fri, 21 Dec 2007, Ismael Farfán Estrada wrote:
> > In file included from
> /tmp/alsa-driver-1.0.15/acore/../alsa-kernel/core/info_oss.c:30,
> > > from /t
Well, that's working correctly then. (Sometimes you have to change
modes to make the receiver re-synch)
On Wed, 2007-12-19 at 22:02 +1030, Tom Lanyon wrote:
> On 19/12/2007, at 9:41 PM, Brian L Scipioni wrote:
>
> > Is your receiver's DTS indicator light still lit after you decode with
> > MPlay
Around about 20/12/07 17:34, John Haxby typed ...
> On my properly functioning F7 box here I don't have that file -- in fact
> the only thing in /usr/lib/alsa-lib or /usr/lib64/alsa-lib is an
> "smixer" directory containing some libraries.
Ditto :-/
> I would guess that there's something od
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