On Wed, 21.01.09 01:39, Takashi Iwai (tiwai@suse.de) wrote:
The function should look like this:
snd_pcm_sframes_t snd_pcm_busy_for(snd_pcm_t *pcm);I called the prototype "busy for" since effectively the value I am looking for is the time the card will be busy with the data it already has, and doesn't need any new data.
Isn't it snd_pcm_delay() that was originally designed for?
No. Let me summarize the meaning of snd_pcm_update_avail(), snd_pcm_delay() and my snd_pcm_busy_for() to opefully make clear where the differences are:
snd_pcm_update_avail() -- returns how many samples can be written right now without blocking.
snd_pcm_delay() -- returns how many samples will be played before the samples that are written now can be heard.
snd_pcm_busy_for() -- returns how many samples will be played before ALSA would enter an underrun situation if no further samples are written.
snd_pcm_update_avail() and snd_pcm_busy_for() return metrics that are solely dependant on the size and metrics of the hardware buffer and its current indexes. snd_pcm_delay() also includes information about any extra latency that comes after the playback buffer.
Onle snd_pcm_update_avail()/snd_pcm_busy_for() are influenced by "fast starts" as done by the USB driver's double buffering and by block-based transfer.
Hmm, I am trying my best to explain why I want this function and what exactly it should do. Any chance I can convince you guys that this function really matters for timer-based audio scheduling?
Did you check my previous patch?
You mean the one that makes snd_pcm_delay() for USB devices actually include the extra latency that comes after the playback buffer? No, I didn't check that one yet. It takes so much time to patch the kernel and test things... I'll try too finally do it this WE.
Lennart