problems with wavefiles in amarok/xmms/audacious

[fenk]

hi!

i would really like to be able to play wavefiles without problems... but:

amarok crashes when reading waves created by cubase sx
xmms does load waves correctly in the playlist, but skips them instead of playing
audacious plays waves, but it takes an eternity until the playlist is loaded, and when fastforwarding/backwarding it crashes sometimes.

is there something wrong with libsndfile??? or what is it? these wavefiles were no problem in windows...

[i92guboj]

[fenk]

sorry i have to disappoint you: all the wavefiles i want to play are carefully mixed in 16bit44khz to be playable with standard apps. i indeed use higher resolutions while recording and mixing (which i for the near future still intend to do in cubase... to learn not too much new software at a time). the files we're talking about were also used live on stage, so they had to be waterproof.

[fenk]

p.s.

aplay works without problems, but aplay is not very useful for me...

[i92guboj]

Well, then that is not the problem I had to ask it hehe, since I have seen that happen many times.

So we can start looking into other things. Even wav files can be encoded on many different ways, you could try to change the mix down options to another encoding scheme. I don't know what version of cubase you are using, and even if I knew I really don't remember the exact output formats that each version can handle. There are many, and I don't know what you are using, so I can't tell any more details.

You said that aplay works, then maybe we can dig a bit more about the thing with the output of aplay -v [file], not that I am an expert in sound formats hehe, but it is just an idea.

I assume the file size is not insane either.

[fenk]

well, i wouldn't like to change all the wavefiles... the only way of changing the problem is to open them with a soundeditor and just save them, without any changes. but for a total of several hours wavefiles this is not the best solution.
aplay -v says strange things (what is it now?! 44khz as it should, or 48khz?)

Playing WAVE '/mnt/e/ultrabert4-2/ultrabert4-22-darkness.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Plug PCM: Rate conversion PCM (48000, sformat=S16_LE)
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 44100
exact rate : 44100 (44100/1)
msbits : 16
buffer_size : 15052
period_size : 940
period_time : 21333
tick_time : 0
tstamp_mode : NONE
period_step : 1
sleep_min : 0
avail_min : 940
xfer_align : 940
start_threshold : 15040
stop_threshold : 15052
silence_threshold: 0
silence_size : 0
boundary : 986447872
Slave: Soft volume PCM
Control: PCM Playback Volume
min_dB: -51
resolution: 256
Its setup is:
stream : PLAYBACK
access : MMAP_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
tick_time : 0
tstamp_mode : NONE
period_step : 1
sleep_min : 0
avail_min : 1024
xfer_align : 1024
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Slave: Direct Stream Mixing PCM
Its setup is:
stream : PLAYBACK
access : MMAP_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
tick_time : 0
tstamp_mode : NONE
period_step : 1
sleep_min : 0
avail_min : 1024
xfer_align : 1024
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Hardware PCM card 0 'Intel 82801DB-ICH4' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : MMAP_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
tick_time : 1000
tstamp_mode : NONE
period_step : 1
sleep_min : 0
avail_min : 1024
xfer_align : 1024
start_threshold : 1
stop_threshold : 1073741824
silence_threshold: 0
silence_size : 1073741824
boundary : 1073741824

[i92guboj]

The wave format seems ok. I dont really know what the problem can be.

My old VIA sound chip worked also with a sampling rate of 48 khz, but that never was an obstacle to play any sound file that I can remember. The message seems a bit scary though. I wonder why aplay can play that ok while the others do not. If audacity fails, well, then there is no mistery, it is known to be a bit unstable still. But amarok should be ok.

I don't know why certain chips tune themselves to this sampling rate, I know that in via it caused problems like noise on all the programs. But yours seems to be another problem, since aplay can handle it ok, so I doubt the problem is at driver or hardware level. I am out of ideas for now, sorry. If I think of something more I will let you know.

Just one thing, are all the apps you tried using the same output plugin? Xine for example?

If so, you could try mplayer or something like that, if that works, kaffeine can use mplayer as backend, and that could be a solution.

[fenk]

i'm not at home right now, but i remember, that amarok uses xine. in audacious i can not select anything but alsa and arts, i chose alsa. same with xmms. maybe there might be the way to a solution... didn't check, what mplayer says. thanks so far, i hope that maybe also others will contribute ideas about this.

edit:
no change about the problem, but i found out that audacious ignores the "pause" button, at least when playing waves... quite annoying feature.