Vladimir Mosgalin wrote:
> Hi everybody.
>
> I have Audigy 2 ZS card and alsa driver 1.0.11 (from fc 2.6.16 kernel).
> I was trying to experiment with p16v, and noticed a few things.
> I'm using analog output.
>
> 1) Distiortion / clipping. Regular playback sound fine at master/pcm @
> 100%. However, output from p16v sounds really distorted when its only
> mixer, "HD Analog Front" is at 100%. I read that 0dB is at 100% mixer
> values for all modern creative cards, is this true? Because according
> to my hearing, clipping disappears at about 80 mixer value. 78-82
> sound the best. This isn't really a problem, just like to clarify
> this issue.
81 is 0dB, 100 is +12dB for that control.
Fortunately, the lastest alsa from the hg repository displays all the dB
gain values for you in alsamixer.
> 2) The correct way of using p16v is plughw 0:4 with padded 32-bit s32le
> 48/96/192 khz data, correct? What about hw 0:4, can it be used? When
> trying to use it in aplay/mplayer/gstreamer, it doesn't work good
> with 48khz data, and with 96/192 plays very short fragment of sound
> very fast. Can it ever be used directly, or absolutely all user
> application can use only plughw?
hw:0,4 playback is a 8 channel output. The plughw:0,4 lets alsa-lib
convert whatever format the user wants into the format the sound card
expects.
hw:0,4 expects
format: S32_LE
subformat: STD
channels: 8
rate: 48000 (48000/1)
period_size: 2046
buffer_size: 16368
> 3) p16v can be opened at other rates like 44.1khz, but sound quality
> isn't good - no wonder. However, the following thing bugs me.
>
> from p16v.h, sorry for long lines:
> #define SRCSel 0x60 /* SRCSel. Default 0x4. Bypass P16V 0x14 */
> /* [0] 0 = 10K2 audio, 1 = SRC48 mixer output.
> * [2] 0 = 10K2 audio, 1 = SRCMulti SPDIF mixer output.
> * [4] 0 = 10K2 audio, 1 = SRCMulti I2S mixer output.
> */
> /* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */
> /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
> /* SRC48 and SRCMULTI sample rate select and output select. */
>
>
> #define SRCMULTI_ENABLE 0x6e /* SRCMulti input audio enable. Default 0xffffffff */
> /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
> /* [7:0] The corresponding P16V channel to SRCMulti_I2S enabled if == 1.
> * [15:8] The corresponding E10K2 channel to SRCMulti I2S enabled.
> * [23:16] The corresponding P16V channel to SRCMulti SPDIF enabled.
> * [31:24] The corresponding E10K2 channel to SRCMulti SPDIF enabled.
> */
>
> Is it my understanding that what I send to p16v in 96khz is converted to
> 48khz after that by default? Can I2S output be used with rates other
> than 48khz? If yes, how do I do this? And what's this stuff about
> converting 48khz to 44.1?
>
> Can I tweak these settings in runtime? Actually, should I? I don't
> really need 96khz output right now, only 24-bit 48khz one. But according
> to this, if I try to send 48khz signal, it gets converted to 44.1 and
> then back to 48khz.. What's with all this stuff? I'm completely confused
> now. Why must it be that complex?
>
The hardware resampler on the card can only handle a single stereo
channel, so I have currently disabled the card's on board resampler, and
alsa-lib resampler is then used instead.
The p16v can handle both 48 and 96kHz.
.formats = SNDRV_PCM_FMTBIT_S32_LE, /* Only
supports 24-bit samples padded to 32 bits. */
.rates = SNDRV_PCM_RATE_192000 |
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100,
The 44100 only works if using the SPDIF output.
James
-------------------------------------------------------------------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the chance to share your
opinions on IT & business topics through brief surveys -- and earn cash
http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Alsa-user mailing list
Alsa-user@xxxxxxxxxxxxxxxxxxxxx
https://lists.sourceforge.net/lists/listinfo/alsa-user
