On Fri, 2009-11-13 at 14:50 +0000, Mark Brown wrote:
> > + /* Update with actual sent buffer size */
> > + if (prtd->out[idx].used != BUF_INVALID_LEN)
> > + prtd->pcm_irq_pos +=
> prtd->out[idx].used;
> > +
> > + if (prtd->pcm_irq_pos > prtd->pcm_size)
> > + prtd->pcm_irq_pos = prtd->pcm_count;
>
> This looks suspicious - it looks like what's going on is that you're
> getting a notification from the DSP that it has consumed a data buffer
> but this might go out of bounds and get clamped.
The first time that it's called prtd->out[idx].used = BUF_INVALID_LEN so
it's skipped(it's the same first bogus time than in my previous mails).
The second time prtd->out[idx].used is equal to the period_bytes_max
which is equal to the .buffer_bytes_max / 2
prtd->pcm_irq_pos is 0 then is incremented to prtd->out[idx].used
So it's clamped only the first time.
By the way I've used that info to skip it in the alsa_send_buffer
so it looks like this:
in alsa_send_buffer I've tried to add a check for the invalid len before
of after the "if (frame->used && prtd->out_needed) {"
here an example of it beeing before "if (frame->used &&
prtd->out_needed) {"
@@ -411,6 +418,10 @@ ssize_t alsa_send_buffer(struct msm_audio *prtd,
const char __user *buf,
spin_lock_irqsave(&the_locks.write_dsp_lock, flag);
frame = prtd->out + prtd->out_tail;
+ if (frame->used == BUF_INVALID_LEN){
+
spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag);
+ break;
+ }
if (frame->used && prtd->out_needed) {
audio_dsp_send_buffer(prtd, prtd->out_tail,
frame->used);
@@ -418,10 +429,12 @@ ssize_t alsa_send_buffer(struct msm_audio *prtd,
const char __user *buf,
prtd->out_needed--;
}
spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag);
for the try where it was after the spin_unlock was not necessary because
it was put just after the spin_unlock
Then...
mplayer said:
alsa-init: got buffersize=19200
alsa-init: got period size 600
and it played a song for half a second
and then the kernel panicked
here the code of mplayer:
// gets buffersize for control
if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams,
&bufsize)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize,
snd_strerror(err));
return 0;
}
else {
ao_data.buffersize = bufsize * bytes_per_sample;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n",
ao_data.buffersize);
}
if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams,
&chunk_size, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize,
snd_strerror(err));
return 0;
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n",
chunk_size);
}
ao_data.outburst = chunk_size * bytes_per_sample;
a strace shows that:
ioctl(0, TIOCGWINSZ, {ws_row=51, ws_col=202, ws_xpixel=0, ws_ypixel=0})
= 0
) = 401, "A: 0.0 (00.0) of 2.0 (02.0) ??"..., 40A: 0.0 (00.0) of 2.0
(02.0) ??,?%
gettimeofday({1259531032, 572021}, NULL) = 0
select(1, [0], NULL, NULL, {0, 0}) = 0SNDRV_PCM_IOCTL_STATUS
(Timeout)
ioctl(6, 0x806c4120, 0xbee5dac0) = 0
nanosleep({0, 27000000}, NULL) = 0
ioctl(6, 0x806c4120, 0xbee5dac0) = 0
nanosleep({0, 27000000}, Read from remote host 192.168.0.202: Connection
reset by peer
Connection to 192.168.0.202 closed.
0x806c4120 corresponds to A 0x20 that corresponds to
SNDRV_PCM_IOCTL_STATUS = _IOR('A', 0x20, struct snd_pcm_status) in
asound.h
After that I moved the snd_pcm_period_elapsed like this:
@@ -154,7 +158,10 @@ void alsa_dsp_event(void *data, unsigned id,
uint16_t *msg)
wake_up(&the_locks.write_wait);
}
spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag);
+
snd_pcm_period_elapsed(prtd->playback_substream);
break;
and removed it from the old location:
diff --git a/sound/soc/msm/msm7k-pcm.c b/sound/soc/msm/msm7k-pcm.c
index 38e8283..3e54184 100644
--- a/sound/soc/msm/msm7k-pcm.c
+++ b/sound/soc/msm/msm7k-pcm.c
@@ -152,8 +152,6 @@ static struct snd_pcm_hw_constraint_list
constraints_sample_rates = {
static void playback_event_handler(void *data)
{
- struct msm_audio *prtd = data;
- snd_pcm_period_elapsed(prtd->playback_substream);
}
If I remember well aplay doesn't use the SNDRV_PCM_IOCTL_STATUS
Denis.
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