Pavel Hofman napsal(a):
Jaroslav Kysela napsal(a):
On Tue, 1 Mar 2011, Pavel Hofman wrote:
Jaroslav Kysela napsal(a):
It's correct, pcm:plug does not support buffer modifiers. Look to "Automatic conversion plugin".
I checked the source code and the keywords seem to be ok
pcm.c: static const char *const names[SND_PCM_HW_PARAM_LAST_INTERVAL + 1] = { [SND_PCM_HW_PARAM_FORMAT] = "format", [SND_PCM_HW_PARAM_CHANNELS] = "channels", [SND_PCM_HW_PARAM_RATE] = "rate", [SND_PCM_HW_PARAM_PERIOD_TIME] = "period_time", [SND_PCM_HW_PARAM_PERIOD_SIZE] = "period_size", [SND_PCM_HW_PARAM_BUFFER_TIME] = "buffer_time", [SND_PCM_HW_PARAM_BUFFER_SIZE] = "buffer_size", [SND_PCM_HW_PARAM_PERIODS] = "periods" };
It's just a table to translate all possibilites between index and name. The plug plugins allow only this:
err = snd_pcm_slave_conf(root, slave, &sconf, 3, SND_PCM_HW_PARAM_FORMAT, SCONF_UNCHANGED, &sformat, SND_PCM_HW_PARAM_CHANNELS, SCONF_UNCHANGED, &schannels, SND_PCM_HW_PARAM_RATE, SCONF_UNCHANGED, &srate); JaroslavWell, still I do not understand why we are talking about the plug plugin, when I am trying to redefine the slave via pcm_slave. The documentation at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html says the pcm_slave can define period_time and buffer_time, while a slightly modified example from the wiki page fails:
pcm_slave.slave_rate44100Hz { pcm "hw:0,0" rate 44100 ---> period_time 1024 }
pcm.rate44100Hz { type plug slave slave_rate44100Hz }
How does the slave definition relate to the plug plugin? Sorry for these basics but I am really trying to understand that, so far no success.
The above description is an universal description for all slaves. Unfortunately, not all plugins accept the whole range of parameters. You have to look to the snd_pcm_slave_conf() caller which parameters are accepted.
Thanks a lot for your clarification. Would it make sense to add these parameters to the plug plugin? I guess it would not be as simple as in dshare.c:
params.period_size = psize; params.buffer_size = bsize;
when opening the slave.
One more question regarding the defaults. Older version of audacious (e.g. in ubuntu karmic) uses a very simple code:
snd_pcm_hw_params_alloca(&hwparams); snd_pcm_hw_params_any(pcm_handle, hwparams); snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(pcm_handle, hwparams, afmt); snd_pcm_hw_params_set_channels(pcm_handle, hwparams, nch); snd_pcm_hw_params_set_rate(pcm_handle, hwparams, rate, 0);
err = snd_pcm_hw_params(pcm_handle, hwparams); if (err < 0) { _ERROR("snd_pcm_hw_params failed: %s", snd_strerror(err)); return -1; }
There is no buffer or period setup. I assume some alsa-lib default value is used. When playing, the period size is set very low:
access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 96000 (96000/1) period_size: 8 buffer_size: 8192
As a result, the sound card (ICH5) keeps throwing 12.000 interrupts per second and CPU load is rather high. Does this "default" period_size value make sense? This is the reason I am trying to specify it outside of the application.
Thanks a lot for help.
Pavel.