August 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH RFC 0/9] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 19 Aug '13

19 Aug '13

Hi, This patch series attempts to fix a long starting problem with the concurrent usage of playback and capture on implicit feedback devices. As Clemens found out, when implicit feedback is used, playback and capture cannot be used in the same time - unless started at the same time (e.g., with jack). While a substream is active, attempting to start a substream in the opposite direction will fail and also (on some devices?) cause the currently active stream to stop. There are multiple issues that contribute to this: 1. URBs are unconditionally deactivated in hw_free. On implicit feedback devices, the sync_endpoint (= data_endpoint of the capture) will be deactivated here. Fixed in patch #4. 2. Changing the alternate settings of an interface might break currently running streams. This happens on my device, but I suspect other devices too. Fixed in patch #7. 3. The endpoint use_count has a subtle issue (see patch #9). This causes jack to work fine, while using two separate programs (most probably) fails. In short: set_format is called from hw_params, but use_count is incremented only later (prepare or trigger). 4. Concurrent usage should be allowed but only if the hw_params match up (patch #8) One problem with this series is that I query the device for the altsettting. This might not be reliable for all devices, so do I plan to remove this. Of course, I've only tested this on my device, but it seems to work good and I couldn't break it so far, so hopefully I got this right. Can someone give it a try on other devices (also on non-implicit-feedback)? Cheers, Eldad Eldad Zack (9): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: correct ep use_count semantics (add set_param flag) sound/usb/card.h | 1 + sound/usb/endpoint.c | 65 +++++++++++++++------ sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 155 +++++++++++++++++++++++++++++++++++++++------------ 4 files changed, 176 insertions(+), 56 deletions(-) -- 1.8.1.5

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[alsa-devel] [PATCH 0/3] HDSPM: Fix LTC ioctl
by Adrian Knoth 19 Aug '13

19 Aug '13

Hi! This series fixes the broken LTC ioctl to query the TCO. Apparently, nobody ever attempted to use the code, otherwise, this rather obvious problem would have been spotted way earlier. This is maybe because not too many people happen to have a TV studio around to make use of this feature. ;) I'll also send a patch to fix alsa-lib. Cheers Adrian Knoth (3): ALSA: hdspm - Fix default value in SNDRV_HDSPM_IOCTL_GET_LTC ALSA: hdspm - Use enums in hdspm_tco_ltc_frames() ALSA: hdspm - Fix SNDRV_HDSPM_IOCTL_GET_LTC include/uapi/sound/hdspm.h | 2 +- sound/pci/rme9652/hdspm.c | 12 ++++++------ 2 files changed, 7 insertions(+), 7 deletions(-) -- 1.8.3.2

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[alsa-devel] [PATCH 0/1] alsa-lib: update hdspm.h
by Adrian Knoth 19 Aug '13

19 Aug '13

Hi! Here's the patch for alsa-lib to fix the broken LTC ioctl. Cheers Adrian Knoth (1): hdspm.h: Update LTC ioctl to use struct hdspm_ltc include/sound/hdspm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) -- 1.8.3.2

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Re: [alsa-devel] [PATCH v11 0/7] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 19 Aug '13

19 Aug '13

On Mon, Aug 19, 2013 at 05:13:57PM +0200, Marc Kleine-Budde wrote: > On 08/19/2013 05:13 PM, Marc Kleine-Budde wrote: > > On 08/19/2013 05:05 PM, Markus Pargmann wrote: > >> Michael Grzeschik (2): > >> ASoC: fsl-ssi: imx-pcm-fiq bugfix > >> ASoC: fsl: disable ssi irq for imx > >> > >> Steffen Trumtrar (1): > >> ASoC: fsl-ssi: add SSIEN errata work around > > > > You should add you S-o-b to patches from other develops. > developers yes, I forgot that, thanks Markus -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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Re: [alsa-devel] [PATCH 3.11-rc5 1/2] ALSA: pcm: Add snd_printd_ratelimit()
by Takashi Iwai 19 Aug '13

19 Aug '13

At Fri, 16 Aug 2013 11:18:59 -0600, Tim Gardner wrote: > > Direct calls to printk_limit() will emit log noise even when CONFIG_SND_DEBUG is not > defined. Add a wrapper macro around printk_limit() that is conditionally defined by > CONFIG_SND_DEBUG. > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Cc: Lars-Peter Clausen <lars(a)metafoo.de> > Cc: Yacine Belkadi <yacine.belkadi.1(a)gmail.com> > Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> > --- > > checkpatch is moaning about the use of printk_ratelimit(), but > printk_ratelimited() seems like overkill for a debug feature. > > WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit > #38: FILE: include/sound/core.h:382: > +#define snd_printd_ratelimit() printk_ratelimit() Thanks, applied both patches now. But it's no real bug, so I postpone this for 3.12 inclusion. Takashi > > include/sound/core.h | 8 ++++++++ > 1 file changed, 8 insertions(+) > > diff --git a/include/sound/core.h b/include/sound/core.h > index c586617..2a14f1f 100644 > --- a/include/sound/core.h > +++ b/include/sound/core.h > @@ -27,6 +27,7 @@ > #include <linux/rwsem.h> /* struct rw_semaphore */ > #include <linux/pm.h> /* pm_message_t */ > #include <linux/stringify.h> > +#include <linux/printk.h> > > /* number of supported soundcards */ > #ifdef CONFIG_SND_DYNAMIC_MINORS > @@ -376,6 +377,11 @@ void __snd_printk(unsigned int level, const char *file, int line, > #define snd_BUG() WARN(1, "BUG?\n") > > /** > + * Suppress high rates of output when CONFIG_SND_DEBUG is enabled. > + */ > +#define snd_printd_ratelimit() printk_ratelimit() > + > +/** > * snd_BUG_ON - debugging check macro > * @cond: condition to evaluate > * > @@ -398,6 +404,8 @@ static inline void _snd_printd(int level, const char *format, ...) {} > unlikely(__ret_warn_on); \ > }) > > +static inline bool snd_printd_ratelimit(void) { return false; } > + > #endif /* CONFIG_SND_DEBUG */ > > #ifdef CONFIG_SND_DEBUG_VERBOSE > -- > 1.7.9.5 >

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Re: [alsa-devel] [PATCH v1 45/49] sound: usb: caiaq: prepare for enabling irq in complete()
by Takashi Iwai 19 Aug '13

19 Aug '13

At Sun, 18 Aug 2013 00:25:10 +0800, Ming Lei wrote: > > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> Acked-by: Takashi Iwai <tiwai(a)suse.e> thanks, Takashi > Cc: alsa-devel(a)alsa-project.org > Acked-by: Daniel Mack <zonque(a)gmail.com> > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > sound/usb/caiaq/audio.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c > index 7103b09..e5675ab 100644 > --- a/sound/usb/caiaq/audio.c > +++ b/sound/usb/caiaq/audio.c > @@ -672,10 +672,11 @@ static void read_completed(struct urb *urb) > offset += len; > > if (len > 0) { > - spin_lock(&cdev->spinlock); > + unsigned long flags; > + spin_lock_irqsave(&cdev->spinlock, flags); > fill_out_urb(cdev, out, &out->iso_frame_desc[outframe]); > read_in_urb(cdev, urb, &urb->iso_frame_desc[frame]); > - spin_unlock(&cdev->spinlock); > + spin_unlock_irqrestore(&cdev->spinlock, flags); > check_for_elapsed_periods(cdev, cdev->sub_playback); > check_for_elapsed_periods(cdev, cdev->sub_capture); > send_it = 1; > -- > 1.7.9.5 >

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Re: [alsa-devel] [PATCH v1 44/49] sound: usb: midi: prepare for enabling irq in complete()
by Takashi Iwai 19 Aug '13

19 Aug '13

At Sun, 18 Aug 2013 00:25:09 +0800, Ming Lei wrote: > > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> Acked-by: Takashi Iwai <tiwai(a)suse.de> thanks, Takashi > Cc: Clemens Ladisch <clemens(a)ladisch.de> > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > sound/usb/midi.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/midi.c b/sound/usb/midi.c > index b901f46..86af276 100644 > --- a/sound/usb/midi.c > +++ b/sound/usb/midi.c > @@ -279,15 +279,16 @@ static void snd_usbmidi_out_urb_complete(struct urb* urb) > struct out_urb_context *context = urb->context; > struct snd_usb_midi_out_endpoint* ep = context->ep; > unsigned int urb_index; > + unsigned long flags; > > - spin_lock(&ep->buffer_lock); > + spin_lock_irqsave(&ep->buffer_lock, flags); > urb_index = context - ep->urbs; > ep->active_urbs &= ~(1 << urb_index); > if (unlikely(ep->drain_urbs)) { > ep->drain_urbs &= ~(1 << urb_index); > wake_up(&ep->drain_wait); > } > - spin_unlock(&ep->buffer_lock); > + spin_unlock_irqrestore(&ep->buffer_lock, flags); > if (urb->status < 0) { > int err = snd_usbmidi_urb_error(urb->status); > if (err < 0) { > -- > 1.7.9.5 >

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[alsa-devel] [PATCH] ALSA: hda - Try to allow haswell HDMI audio even without powerwell
by David Henningsson 19 Aug '13

19 Aug '13

If compiled without CONFIG_SND_HDA_I915, the audio driver cannot request power well. However, if the power well is on for other reasons, maybe audio can still work. Therefore, do not skip the card completely if compiled without CONFIG_SND_HDA_I915. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) I'm going to apply this on my local DKMS tree, it's important for our backports as we combine different versions of the kernel with backports for haswell etc. I'm not sure if this is beneficial to have on 3.11+ kernels as well, but I guess it could be. At least it shouldn't harm. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7f9e406..c6c9829 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3855,11 +3855,13 @@ static int azx_probe_continue(struct azx *chip) /* Request power well for Haswell HDA controller and codec */ if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { +#ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(); if (err < 0) { snd_printk(KERN_ERR SFX "Error request power-well from i915\n"); goto out_free; } +#endif hda_display_power(true); } -- 1.7.9.5

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[alsa-devel] [PATCH v2] ALSA: hda - Limit internal mic boost for a few more Thinkpad machines
by David Henningsson 19 Aug '13

19 Aug '13

The higher mic boosts (on internal mic) are so noisy they're unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213820 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d67a48c..00d00b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3704,9 +3704,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: ac97: Provide stub DAPM integration
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ac97.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index ec73518..8d9ba4ba 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -23,6 +23,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget ac97_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ac97_routes[] = { + { "AC97 Capture", NULL, "RX" }, + { "TX", NULL, "AC97 Playback" }, +}; + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -117,6 +127,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, + + .dapm_widgets = ac97_widgets, + .num_dapm_widgets = ARRAY_SIZE(ac97_widgets), + .dapm_routes = ac97_routes, + .num_dapm_routes = ARRAY_SIZE(ac97_routes), }; static int ac97_probe(struct platform_device *pdev) -- 1.8.4.rc3

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