Re: [Jack-Devel] what could cause: "impossible sample width (1) discovered!"
>> "16bits or 24bits" indeed. I don't know where Jack's one byte
>> comes from, but you might want to correct this.
>
> JACK's ALSA backend does this:
>
> driver->playback_sample_bytes =
> snd_pcm_format_physical_width (driver->playback_sample_format)
> / 8;
>
> and then later checks that value to determine how it is going to
> convert samples into 32 bit floating point.
>
> it is certainly true that it does not handle 8 bit values, and the
> error message could be friendlier. but the value is being reported by
> ALSA.
Hi again,
i did some testing with arecord but i do
not fully understand the outputs. As i
understand it, my hardware is able to
send 16bit. Maybe i do have to make some
alsa configurations?
here is what i did:
$ arecord -vv testrec.wav
Recording WAVE 'testrec.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
Plug PCM: Rate conversion PCM (48000, sformat=U8)
Converter: linear-interpolation
Protocol version: 10002
Its setup is:
stream : CAPTURE
access : RW_INTERLEAVED
format : U8
subformat : STD
channels : 1
rate : 8000
exact rate : 8000 (8000/1)
msbits : 8
...
Slave: Route conversion PCM (sformat=S16_LE)
Transformation table:
0 <- 0*0.5 + 1*0.5
Its setup is:
stream : CAPTURE
access : MMAP_INTERLEAVED
format : U8
subformat : STD
channels : 1
rate : 48000
exact rate : 48000 (48000/1)
msbits : 8
...
Slave: Direct Snoop PCM
Its setup is:
stream : CAPTURE
access : MMAP_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 16384
...
Hardware PCM card 0 'LUFA Audio In Demo' device 0 subdevice 0
Its setup is:
stream : CAPTURE
access : MMAP_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
...
overrun!!! (at least 1276313785.524 ms long)
Status:
state : XRUN
trigger_time: 1300121528.603269000
tstamp : 1300121528.608661000
delay : 2560
avail : 170
avail_max : 117
#+ | 00%^C
Aborted by signal Interrupt...
and later i did:
$ arecord -vv -fdat testrec.wav
Recording WAVE 'testrec.wav' : Signed 16 bit Little Endian, Rate 48000
Hz, Stereo
Plug PCM: Direct Snoop PCM
Its setup is:
stream : CAPTURE
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 16384
period_size : 1024
period_time : 21333
tstamp_mode : NONE
period_step : 1
...
Hardware PCM card 0 'LUFA Audio In Demo' device 0 subdevice 0
Its setup is:
stream : CAPTURE
access : MMAP_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
...
#+ | 00%^C
Aborted by signal Interrupt...
Does this help to identify where the missing
bits get lost?
1300123368.17839_0.ltw:2,a <AANLkTi=1yA_NY4Vv8m5YSQgWraAye2rq55poPYxg9YHe at mail dot gmail dot com>