Re: intel8x0 dual codec issue

[Ron Cococcia <ron.cococcia@request.com>]

I had been looking through that block of code, and had made some of my 
own "tweaks".  What I did was make "reg_ok" an array (size 4, all 
initialized to 0).  I used the codec index (cidx) to index into that 
array.  The idea is that there would be a reg_ok for each codec, instead 
of just 1 for all codecs.  This correctly set the output on all codecs 
after making the change.

Unfortunately, I was still noticing some audio distortions on the 
secondary codec.  Originally, some tones would sound "tinny".  After the 
change, the tinny sound went away, but I could hear distinct "clicks" in 
the audio.  What causes this is unknown to me, but I thought it might be 
some kind of problem where both codecs are producing some information 
(maybe interrupts) and causing some weirdness with loading the data.  If 
I knew more about what was going on I would have looked into this more, 
but I am still just an novice driver "tweaker".

On a hunch, I figured that if the primary codec was doing something 
weird, I might need to make it "forget" about the primary codec.  I went 
into intel8x0.c, and started to take a look, and was able to add a quick 
test hack.  If the number of codecs detected in snd_intel8x0_mixer was 
2, set it to 1.  In snd_intel8x0_codec_{read/write/read_test}, it would 
add 1 to ac97->num, so it would access the secondary codecs values and 
skip the primary.

After compiling and installing, alsamixer only knows of 1 codec 
(CS4299), and the audio playback appears to be greatly improved with no 
clicks or tinny sounds.

For now, I will be leaving my hack in, unless there is a better fix 
available.  A few hours after getting it working, I managed to hose my 
development setup anyway, so it will be a little while before I can play 
around again.

Thanks,
Ron


Takashi Iwai wrote:
> At Mon, 07 Feb 2005 23:43:39 -0500,
> Ron Cococcia wrote:
> 
>>Hello,
>>
>>I'm having a little trouble with a dual codec setup that I am using. 
>>The motherboard has an Intel 815 chip (82801BAICH2).  The motherboard 
>>has the primary (00) codec, which is an AD1885.  A secondary (01) codec 
>>has been added over a CNR connector.  This one is a CS4299.  The 
>>secondary codec uses the timing values generated by the primary.
>>
>>I had been hearing some distortion in the audio.  It was quite 
>>noticeable.  Using a wav file of sine waves that has different 
>>frequencies (440, 880, 1760, 3520) at 44100Hz, I could see that there 
>>was some jitter on a scope.  As the frequencies went up, it got worse. 
>>I then created another test file with the same frequencies, but sampled 
>>at 48000Hz.  Playback of that audio was extremely clean on the scope.
>>
>>Looking into it more, I noticed that the PCM front DAC (ac97#1-1) was 
>>not adjusting itself to the appropriate frequency.  The DAC on the 
>>primary codec was switching between 44100Hz and 48000Hz when I played 
>>the different samples.
>>
>>I'd like to have the secondary codec reflect the output rate that is 
>>being sent to it.  How might I get it to do this?  Is it a bug (should 
>>the codec front DAC on the secondary codec reflect the appropriate 
>>frequency?), or how is it intended to work?
> 
> 
> I guess it's a bug.  Could you try the attached patch?
> 
> 
> Takashi
> 
> 
> ------------------------------------------------------------------------
> 
> Index: alsa-kernel/pci/ac97/ac97_pcm.c
> ===================================================================
> RCS file: /suse/tiwai/cvs/alsa/alsa-kernel/pci/ac97/ac97_pcm.c,v
> retrieving revision 1.19
> diff -u -r1.19 ac97_pcm.c
> --- alsa-kernel/pci/ac97/ac97_pcm.c	27 Dec 2004 13:19:32 -0000	1.19
> +++ alsa-kernel/pci/ac97/ac97_pcm.c	10 Feb 2005 17:19:45 -0000
> @@ -602,8 +602,10 @@
>  					snd_printk(KERN_ERR "invalid AC97 slot %i?\n", i);
>  					continue;
>  				}
> +#if 0
>  				if (reg_ok & (1 << (reg - AC97_PCM_FRONT_DAC_RATE)))
>  					continue;
> +#endif
>  				//printk(KERN_DEBUG "setting ac97 reg 0x%x to rate %d\n", reg, rate);
>  				err = snd_ac97_set_rate(pcm->r[r].codec[cidx], reg, rate);
>  				if (err < 0)



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