On Mon, 2006-10-23 at 15:09 +0200, Takashi Iwai wrote:
> At Fri, 20 Oct 2006 16:12:48 -0400,
> Lee Revell wrote:
> > ~ # aplay /usr/share/sounds/test/phone.wav
> > Playing WAVE '/usr/share/sounds/test/phone.wav': Signed 16 bit Little
> > Endian, Rate 44100 Hz, Stereo
> > hw_params: stereo yes, format 16bit, rate 44100
> > dream_pcm_open: channel 0x0 eight_bit 0x0 stereo 0x1 rate 0xac44
> > dream_playback_ack
> > dream_playback_copy: not running!
> > dream_pcm_start: channel is 0
> > dream_work_transfer: polling status byte returned 0
> > calling snd_pcm_indirect_playback_transfer
> > in dream_playback_copy: 0x4000 bytes DMA area 0xcdfb0000 sw_data 0x0
> > dream_playback_ack
> > dream_playback_ack
> > dream_work_transfer: polling status byte returned 0
> > calling snd_pcm_indirect_playback_transfer
> > dream_playback_ack
> > playback size is 0x1005
> > dream_playback_ack
> > playback size is 0x100a
> > dream_playback_ack
> > playback size is 0x100f
> > dream_playback_ack
> > playback size is 0x1015
> > dream_playback_ack
> > dream_playback_ack
>
> The problem looks like the calculation of dream->playback_hw_ptr
> passed to snd_pcm_indirect_playback_pointer(). This must be a _byte_
> offset in the "hardware" ring buffer. That is, it's between 0 and
> (rec->hw_buffer_size - 1). You set hw_buffer_size = 0x4000, but
> calculate playback_hw_ptr as [0, runtime->buffer_size] and in frames.
>
With your patches I still get a similar result:
snd_pcm_indirect_playback_transfer is only called once or twice and the
first chunk of sound repeats endlessly.
~ # aplay /usr/share/sounds/test/phone.wav
Playing WAVE '/uhw_params: stereo yes, format 16bit, rate 44100
sr/share/sounds/dream_pcm_open: channel 0x0 eight_bit 0x0 stereo 0x1
rate 0xac44
test/phone.wav' dream_playback_ack
dream_playback_copy: not running!
dream_pcm_start: channel is 0
: Signed 16 bit calling snd_pcm_indirect_playback_transfer
in dream_playback_copy: 0x4000 bytes DMA area 0xcdfb0000 sw_data 0x0
Little Endian, Rdream_playback_copy: polling status byte returned 0
ate 44100 Hz, Stereo
dream_playback_copy done
dream_playback_ack
dream_playback_ack
dream_playback_ack
calling snd_pcm_indirect_playback_transfer
in dream_playback_copy: 0x4000 bytes DMA area 0xcdfb0000 sw_data 0x16384
dream_playback_ack
dream_playback_pointer: 0x134
At this point playback hangs.
I just can't understand why the version with the copy callback worked
perfectly, but this PCM indirect stuff has been nothing but trouble.
Did I mention I am using ALSA 1.0.9b? Does that version have a buggy
PCM indirect implementation?
Is it really necessary to make up a fake pointer like this? Can't I
just advance the pointer by 0x2000 words after writing each chunk?
Here is my userspace PCM playback implementation that works *perfectly*.
I just need a way to make ALSA do the same thing when a .wav file is
played.
dream_pcm_open(chip, port, channel, audio8bits, stereo, sample_rate);
dream_pcm_set_vol_fl(chip, port, channel, 0xFFFF);
dream_pcm_set_vol_fr(chip, port, channel, 0xFFFF);
dream_pcm_start(chip, port, channel);
bptr = & buffer[DREAM_MS_WAV_HEADER_SIZE/2];
rest = bufsize-DREAM_MS_WAV_HEADER_SIZE/2;
if(audio8bits) /* Endianness doesn't make sence for 8bits WAV files */
endianness = DREAM_BIG_ENDIAN;
for (i = 0; rest > 0; i++, bptr+=size)
{
retval = dream_get_mpu_data_poll_status_byte(chip, port, MPU401_IT_MASK,
MPU401_IT_VALUE,
100, & tout);
if(retval < 0)
pRes += sprintf(pRes, "%d:%d[%u];", i, retval, tout/1000);
size = AKA_DIAG_MIN(rest, samples);
/* Send more half buffer of dummy samples pausing after each */
for (sample = 0; sample < size; sample++)
dream_mpu_oper16(MPU_WRITE_DATA16, port, endianness,
&bptr[sample], 0);
dream_pcm_end_xfer(chip, port, channel);
rest -= size;
if(rest == 0)
{
if(size == samples) /* last half buffer was full */
{
rest = 2; /* send 2 additional words */
bptr -= 2;
}
}
}
rsp = dream_pcm_close(chip, port, channel);
Lee
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