Last version come with Alsa support (configure with --enable-mp4live-alsa)
ALSA audio_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 <unruh@xxxxxxxxxxxxxx> wrote:
> 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
>
> List of Media Sources
>
> V4L video_v4l_source.cpp
> Acquires YUV12 images from a V4L (Video For Linux) device
> V4L2 video_v4l2_source.cpp
> Acquires YUV12 images from a V4L2 (Video For Linux) device (recommended)
> OSS audio_oss_source.cpp
> Acquires PCM16 audio samples from an OSS (Open Sound System) device
> self loop_source.cpp
> Acquires raw YUV or PCM from main process
>
>
>
>
>
> >
> > 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 run mp4live it says:
> >
> > mp4live: pcm_params.c:2351: sndrv_pcm_hw_params: Assertion `err >= 0' failed.
> >
> > I dont know what to do.
> >
> >
> > Thanks
> >
> >
> >
> >
> > On 22/12/2007, Bill Unruh <unruh@xxxxxxxxxxxxxx> wrote:
> >> 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" the audio... where? .. I
> >>> suppose this should be in asound.conf (or .asoundrc)... setting in the
> >>> plug parameters "format", "channels" and "rate"... this way :
> >>
> >> arecord -f cd >outfile
> >>
> >>>
> >>> pcm.plugfile{
> >>> type plug
> >>> slave {
> >>> pcm infile
> >>> format S16_LE
> >>> channels 2
> >>> rate 44100
> >>> }
> >>> }
> >>>
> >>> but this does not work...
> >>>
> >>>
> >>> thanks anyway
> >>>
> >>>
> >>>
> >>> On 22/12/2007, Lee Revell <rlrevell@xxxxxxxxxxx> wrote:
> >>>> On Dec 21, 2007 9:39 PM, Carlos Hernandez <dspcarlos@xxxxxxxxx> 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:
> >>>>>
> >>>>> carlos@chaos:~$ 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 case, the default
> >>>> of 8Khz 8-bit mono)
> >>>>
> >>>> Lee
> >>>>
> >>>
> >>>
> >>>
> >>
> >> --
> >> William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273
> >> Physics&Astronomy | Advanced Research | Fax: +1(604)822-5324
> >> UBC, Vancouver,BC | Program in Cosmology | unruh@xxxxxxxxxxxxxx
> >> Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/
> >>
> >
> >
> >
>
> --
> William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273
> Physics&Astronomy | Advanced Research | Fax: +1(604)822-5324
> UBC, Vancouver,BC | Program in Cosmology | unruh@xxxxxxxxxxxxxx
> Canada V6T 1Z1 | and Gravity | www.theory.physics.ubc.ca/
>
--
--------------------------------------
Universidad de Talca
Facultad de Ingeniería
Ingeniería Civil en Computación
--------------------------------------
Movil +56 9 92624541
Skype dspcarlos
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