[alsa-devel] Questions about writing a new ALSA driver for a very limitted device

[Tim Harvey]

--- Takashi Iwai <tiwai at suse.de> wrote:

> The "period" defines the frequency to update the status, usually via
> the invokation of interrupts.  The "period_size" defines the frame
> sizes corresponding to the "period time".  This term corresponds to
> the "fragment size" on OSS.  On major sound hardwares, a ring-buffer
> is divided to several parts and an irq is issued on each boundary.
> The period_size defines the size of this chunk.
> 
> On some hardwares, the irq is controlled on the basis of a timer.  In
> this case, the period is defined as the timer frequency to invoke an
> irq.
> 
> 

My system has a hardware buffer of 4096 bytes (so I will use an intermediate
buffer ) and is fixed at 8KHz 16bit stereo.  The 4K hw buffer would accommodate
128ms of data which is pretty small.  Therefore I'll choose to use a 32K
intermediate buffer which will accommodate about 1sec of audio data.  My IRQ is
timer based so I will need to decide upon a reasonable period.  As the period
seems to dictate the ability for the middle layer to see the status of the
buffer I'll use 4 periods allowing a 250ms granularity in my intermediate
buffer.

I'm assuming that the snd_pcm_hardware struct has a min AND max period_bytes
and periods because the framesize can vary if your hardware supports multiple
configurations.  As mine only supports 8Khz 16bit stereo, my min/max values
should all be the same correct?

So using the above info my struct should look like:

static snd_pcm_hardware_t snd_my_playback_hw = {
        .info = (SNDRV_PCM_INFO_MMAP |
                 SNDRV_PCM_INFO_INTERLEAVED |
                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
                 SNDRV_PCM_INFO_MMAP_VALID),
        .formats =          SNDRV_PCM_FMTBIT_U16_BE,
        .rates =            SNDRV_PCM_RATE_8000,
        .rate_min =         8000,
        .rate_max =         8000,
        .channels_min =     2,
        .channels_max =     2,
        .buffer_bytes_max = 32768,
        .period_bytes_min = 8192,
        .period_bytes_max = 8192,
        .periods_min =      4,
        .periods_max =      4,
};

Does this seem reasonable?

Because I'm using a 32K intermediate buffer I'll preallocate it with
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
  snd_dma_continuous_data(GFP_KERNEL), 0, 32768);

I want to make sure I'm implementing things properly for full duplex
capability.  I'm creating the pcm device via:

snd_pcm_new(card, "my dev", 0, 1 /* play streams */, 1 /* capt streams */,
&pcm)

If I understand this preallocation a 32K DMAable buffer will be allocated by
ALSA for each stream my device can play/capture (= 2 streams, 1 playback and 1
capture).  I can obtain the buffer pre-allocated from the runtime data in the
prepare callback and use that buffer as my intermediate buffer correct?  I
don't have a great grasp of the 'prepare' callback yet, but because thats the
first time you've got access to valid format/buffer info I'll assume that each
time 'prepare' is called your PCM buffer/hardware/stream should be initialized.

If I understand correctly the pointer callback should return the num of frames
processed in the 'intermediate buffer'.  Because this is a virtual ring buffer,
I guess I should simply return the offset in the buffer (divided by my
bytes_per_frame which is 4) that I last pulled data from?

I've looked at the example dealing with the hw interrupt and period_elapsed in
http://www.alsa-project.org/~iwai/writing-an-alsa-driver/x773.htm.  The example
doesn't specify where it was getting runtime pointer from so I am assuming that
I'll need to keep a pointer to each (playback and capture) substream so that I
can call period_elapsed appropriately for each if they are concurrent?

Thanks for all the help.  I've got rudimentary playback working but I'm pretty
sure I'm not implementing the period_elapsed and the pointer callback correctly
as the relatively simple playback app I'm using sits for a few seconds after
playback of the 2 sec test sample I'm using completes.

Tim