Re: [Alsa-devel] [PATCH] Audigy LS support.

2004-05-28 Thread James Courtier-Dutton
Takashi Iwai wrote:
At Thu, 27 May 2004 20:17:17 +0100,
James Courtier-Dutton wrote:
Here is my first go at Audigy LS support.
It can play sound to the front speakers.

great!

/* hardware definition */
static snd_pcm_hardware_t snd_audigyls_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_S16_LE,
	.rates =		SNDRV_PCM_RATE_48000,
	.rate_min =		48000,
	.rate_max =		48000,
	.channels_min =		2,
	.channels_max =		2,
	.buffer_bytes_max =	(32*1024),
	.period_bytes_min =	64,
	.period_bytes_max =	(16*1024),
	.periods_min =		2,
	.periods_max =		16,
^^
are you sure this is ok?
in the case of emu10k1, it generates irq twice per buffer (HALF and
BUFFER).  then periods_max must be 2.
With periods_max set to 2, sound plays, but has artifacts. With periods 
 2, the sound is smooth, without any clicks etc.


anyway, the driver code is quite similar with emu10k1x.c.
i checked the diff between them and the amount of difference is really
small.
how about merging them?  then we can understand / maintain the codes
better.
Takashi

I would like to keep the two drivers separate for a bit. That way I can 
keep changing stuff without caring if the emu10k1x.c stays working.
Once I get all the features working, we could then look at merging the two.
There is still a lot to go.
1) 24bit 192khz sound out.
2) 5.1 sound.
3) Digital SPDIF output.
4) Capture.
5) MIDI.
It seems that the changing of the sample rate etc. from the default is 
done via some sort of UART.

Also, I can change how often the interrupts happen, so I will need to 
implement that.

Cheers
James
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Re: [Alsa-devel] [PATCH] Audigy LS support.

2004-05-28 Thread Takashi Iwai
At Fri, 28 May 2004 12:16:09 +0100,
James Courtier-Dutton wrote:
 
 Takashi Iwai wrote:
  At Thu, 27 May 2004 20:17:17 +0100,
  James Courtier-Dutton wrote:
  
 Here is my first go at Audigy LS support.
 It can play sound to the front speakers.
  
  
  great!
  
  
  
 /* hardware definition */
 static snd_pcm_hardware_t snd_audigyls_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_S16_LE,
 .rates =SNDRV_PCM_RATE_48000,
 .rate_min = 48000,
 .rate_max = 48000,
 .channels_min = 2,
 .channels_max = 2,
 .buffer_bytes_max = (32*1024),
 .period_bytes_min = 64,
 .period_bytes_max = (16*1024),
 .periods_min =  2,
 .periods_max =  16,
  
  ^^
  are you sure this is ok?
  in the case of emu10k1, it generates irq twice per buffer (HALF and
  BUFFER).  then periods_max must be 2.
 
 With periods_max set to 2, sound plays, but has artifacts. With periods 
   2, the sound is smooth, without any clicks etc.

well, then the question is how the period size is controlled.
you have never set the period size/bytes on the hardware.


  anyway, the driver code is quite similar with emu10k1x.c.
  i checked the diff between them and the amount of difference is really
  small.
  how about merging them?  then we can understand / maintain the codes
  better.
  
  
  Takashi
  
  
 
 I would like to keep the two drivers separate for a bit. That way I can 
 keep changing stuff without caring if the emu10k1x.c stays working.

ok.  but it's also true that changing one of them can help to reveal
features of these chips.  anyway, keep tracking of cvs changes...


Takashi


 Once I get all the features working, we could then look at merging the two.
 There is still a lot to go.
 1) 24bit 192khz sound out.
 2) 5.1 sound.
 3) Digital SPDIF output.
 4) Capture.
 5) MIDI.
 It seems that the changing of the sample rate etc. from the default is 
 done via some sort of UART.
 
 Also, I can change how often the interrupts happen, so I will need to 
 implement that.
 
 Cheers
 James
 


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