Em 07-09-2011 00:20, Devin Heitmueller escreveu: > On Tue, Sep 6, 2011 at 10:58 PM, Devin Heitmueller > <dheitmueller@xxxxxxxxxxxxxx> wrote: >> On Tue, Sep 6, 2011 at 11:29 AM, Mauro Carvalho Chehab >> <mchehab@xxxxxxxxxx> wrote: >>> There are several issues with the original alsa_stream code that got >>> fixed on xawtv3, made by me and by Hans de Goede. Basically, the >>> code were re-written, in order to follow the alsa best practises. >>> >>> Backport the changes from xawtv, in order to make it to work on a >>> wider range of V4L and sound adapters. >>> >>> Signed-off-by: Mauro Carvalho Chehab <mchehab@xxxxxxxxxx> >> >> Mauro, >> >> What tuners did you test this patch with? I went ahead and did a git >> pull of your patch series into my local git tree, and now my DVC-90 >> (an em28xx device) is capturing at 32 KHz instead of 48 (this is one >> of the snd-usb-audio based devices, not em28xx-alsa). >> >> Note I tested immediately before pulling your patch series and the >> audio capture was working fine. >> >> I think this patch series is going in the right direction in general, >> but this patch in particular seems to cause a regression. Is this >> something you want to investigate? I think we need to hold off on >> pulling this series into the new tvtime master until this problem is >> resolved. >> >> Devin >> >> -- >> Devin J. Heitmueller - Kernel Labs >> http://www.kernellabs.com >> > > Spent a few minutes digging into this. Looks like the snd-usb-audio > driver advertises 8-48KHz. However, it seems that it only captures > successfully at 48 KHz. > > I made the following hack and it started working: > > diff --git a/src/alsa_stream.c b/src/alsa_stream.c > index b6a41a5..57e3c3d 100644 > --- a/src/alsa_stream.c > +++ b/src/alsa_stream.c > @@ -261,7 +261,7 @@ static int setparams(snd_pcm_t *phandle, snd_pcm_t *chandle, > fprintf(error_fp, "alsa: Will search a common rate between %u and %u\n", > ratemin, ratemax); > > - for (i = ratemin; i <= ratemax; i+= 100) { > + for (i = ratemax; i >= ratemin; i-= 100) { > err = snd_pcm_hw_params_set_rate_near(chandle, c_hwparams, &i, 0); > if (err) > continue; > > Basically the above starts at the *maximum* capture resolution and > works its way down. One might argue that this heuristic makes more > sense anyway - why *wouldn't* you want the highest quality audio > possible by default (rather than the lowest)? That change makes sense to me. Yet, you should try to disable pulseaudio and see what's the _real_ speed that the audio announces. On Fedora, just removing pulsaudio-oss-plugin (or something like that) is enough. It seems doubtful that my em2820 WinTV USB2 is different than yours. I suspect that pulseaudio is passing the "extra" range, offering to interpolate the data. > Even with that patch though, I hit severe underrun/overrun conditions > at 30ms of latency (to the point where the audio is interrupted dozens > of times per second). Yes, it is the same here: 30ms on my notebook is not enough for WinTV USB2 (it is OK with HVR-950). > Turned it up to 50ms and it's much better. > That said, of course such a change would impact lipsync, so perhaps we > need to be adjusting the periods instead. We've added a parameter for that on xawtv3 (--alsa-latency). We've parametrized it at the alsa stream function call. So, all it needs is to add a new parameter at tvtime config file. > ALSA has never been my area of expertise, so I look to you and Hans to > offer some suggestions. > > Devin > -- To unsubscribe from this list: send the line "unsubscribe linux-media" in the body of a message to majordomo@xxxxxxxxxxxxxxx More majordomo info at http://vger.kernel.org/majordomo-info.html
