Hi ,
snd_usb_hardware is defined in sound/usb/pcm.c as :
static struct snd_pcm_hardware snd_usb_hardware =
{
.
.buffer_bytes_max = 1024 * 1024,
.period_bytes_min = 64,
.period_bytes_max = 512 * 1024,
.periods_min = 2,
.perio
On 06/26/12 13:24, Bryan Ischo wrote:
>
> I do still get pops, which I don't understand, but they only occur
> after very "large" underruns, where I have allowed a significant
> amount of silence to play, instead of "small" underruns, which happen
> when my thread delivering audio data is very s
On 06/26/12 12:39, Bryan Ischo wrote:
>
> So what I need is a way to tell ALSA to re-sync the application write
> position with the hardware read position without having to reset the
> device.
>
> Before I start digging into the ALSA code to understand this better,
> is there any obvious way to
On 06/26/12 11:03, Bryan Ischo wrote:
> On 06/26/12 10:15, Bryan Ischo wrote:
>>
>> snd_pcm_sw_params_get_boundary(sw_params, &boundary);
>>
>> // Never underrun; allow the sound card to just keep playing whatever
>> is in the ring
>> // buffer, including re-playing old audio if new audio is not a
OK, solved: I have to specify "hw:2,3" as passthrough-parameter for xine, then
it works.
Original-Nachricht
> Datum: Mon, 25 Jun 2012 08:52:13 +0200
> Von: Clemens Ladisch
> An: q...@gmx.de
> CC: alsa-user@lists.sourceforge.net
> Betreff: Re: [Alsa-user] Also ignores passthroug
Le Tue, 26 Jun 2012 08:56:01 -0700 (PDT),
Bill Unruh a écrit :
> On Tue, 26 Jun 2012, Dominique Michel wrote:
>
> > Le Mon, 25 Jun 2012 14:08:07 -0400,
> > Jerry Geis a écrit :
> >
> >>> The best things to do would be to use something like a dbx, a
> >>> feedback suppressor. The problem is than
On 06/26/12 10:15, Bryan Ischo wrote:
>
> snd_pcm_sw_params_get_boundary(sw_params, &boundary);
>
> // Never underrun; allow the sound card to just keep playing whatever
> is in the ring
> // buffer, including re-playing old audio if new audio is not available
> snd_pcm_sw_params_set_stop_threshol
On 06/23/12 05:59, Clemens Ladisch wrote:
> Bryan Ischo wrote:
>> Maybe I don't understand what the boundary is supposed to represent;
>> I thought it was supposed to represent the largest number of audio
>> frames that could be buffered for playback by the driver.
> That would be the buffer size.
I didn't catch the beginning of this thread? Just thought I'd add a comment:
some microphones (e.g. cardioid) are designed to shape the gain/response, by
angle, which helps to suppress feedback, e.g. from the back of the mic.
Depending on how you hold the mic, it is possible to block off some of
On Tue, 26 Jun 2012, Dominique Michel wrote:
Le Mon, 25 Jun 2012 14:08:07 -0400,
Jerry Geis a écrit :
The best things to do would be to use something like a dbx, a
feedback suppressor. The problem is than they are expensive pieces
of hardware. The principle is simple: above a given (adjustabl
Le Mon, 25 Jun 2012 14:08:07 -0400,
Jerry Geis a écrit :
> > The best things to do would be to use something like a dbx, a
> > feedback suppressor. The problem is than they are expensive pieces
> > of hardware. The principle is simple: above a given (adjustable)
> > input level, you get an output
On Tue, Jun 26, 2012 at 14:24:03, Mark Brown wrote:
> On Tue, Jun 26, 2012 at 11:33:43AM +0530, Hebbar, Gururaja wrote:
> > In sound/soc/codecs/tlv320aic3x.c
> >
> > data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
> > snd_soc_write(codec, AIC3X_PLL_PROGA_REG,
> >
On Tue, Jun 26, 2012 at 14:31:42, Prchal Jiří wrote:
> Hi Gururaja,
> shouldn't be better to use:
>
> snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p);
>
> instead of "read mask write" ?
Sure will resend the patch. Thanks for the review
> Even at this place you don't need to ke
I ran into this:
http://arunraghavan.net/2011/08/hello-hello-hello/
Perhaps I will give pulseaudio a try with the module-echo-cancel.
Thanks all.
Jerry
--
Live Security Virtual Conference
Exclusive live event will cove
14 matches
Mail list logo
