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[alsa-devel] How to report error in _pointer function?
by Timur Tabi 11 Aug '08

11 Aug '08

In my driver, I can detect in the _pointer callback function whether or not the DMA engine has actually started. Sometimes, when I have a programming error, the DMA will not start, so my _pointer function calculates a crazy value for the current position. The number it returns causes ALSA to go haywire (see my post titled, "underrun!!! (at least 1786051083.613 ms long)". What is the best way to handle this? Is there a way I can tell ALSA, "hey, this is really screwed up, just abort playback and return an error to the app"? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH 1/2] spitz: MONO -> MONO1
by Dmitry Baryshkov 11 Aug '08

11 Aug '08

Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/spitz.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 8f5c2ac..dff0f0e 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -303,7 +303,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "LINPUT3"); snd_soc_dapm_disable_pin(codec, "RINPUT3"); snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO"); + snd_soc_dapm_disable_pin(codec, "MONO1"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { -- 1.5.6.3

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[alsa-devel] clicking/noise when upsampling with ESI juli@
by Amaury De Ganseman 11 Aug '08

11 Aug '08

Hi, There's a problem, when I start to play a file the sound is bad during 1sec or less (clicking,cracking,...) or when I switch between tracks I hear clicks. Any idea ? Note: It works very well with OSS driver (last mercurial) Thanks for help. Here my asoundrc: defaults.pcm.rate_converter "speexrate_best" pcm.!default { type plug slave.pcm { type dmix ipc_key 1478 slave { pcm "hw:0,1" format S32_LE #period_time 0 period_size 2048 buffer_size 8192 rate 48000 } } #rate_converter "speexrate_best" #rate_converter "samplerate_best" }

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[alsa-devel] Kill snd_assert()
by Takashi Iwai 11 Aug '08

11 Aug '08

Hi, as some people already mentioned, we have a pretty ugly macro, snd_assert(). Actually, this macro is sometimes useful, but the implementation including the code flow in arguments is really bad. I tried to kill this, either the following way: - replace snd_assert(cond, flow...); with if (snd_BUG_ON(!cond)) flow...; - use WARN_ON() if the check is needed essentially regardless of CONFIG_SND_DBEBUG value - remove snd_assert() for obviously unneeded code-paths The new snd_BUG_ON() macro takes the condition, and works like WARN_ON() if CONFIG_SND_DEBUG=y: prints the warning with stack trace, and returns the evaluated value. When CONFIG_SND_DEBUG=n, the macro returns always zero, so that the compiler will optimize out the unreached flows. Since snd_BUG_ON() takes the condition of a bug, it's a negative condition against snd_assert(). It follows BUG_ON() and WARN_ON(), and easier to understand, IMO. The patches are found on topic/remove-assert branch of my git tree: git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-2.6.git For gitweb, http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog;h=… I haven't merged this topic yet. If you have a big objection or a better implementation idea, please let me know. Takashi

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[alsa-devel] [PATCH] wm8750: it's MONO1, not MONO
by Dmitry Baryshkov 11 Aug '08

11 Aug '08

Since first commit wm8750 contained output named MONO, but all routes mentioned MONO1. Correct MONO to be MONO1. Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8750.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e23cb09..2e71394 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -348,7 +348,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("ROUT1"), SND_SOC_DAPM_OUTPUT("LOUT2"), SND_SOC_DAPM_OUTPUT("ROUT2"), - SND_SOC_DAPM_OUTPUT("MONO"), + SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_OUTPUT("OUT3"), SND_SOC_DAPM_INPUT("LINPUT1"), -- 1.5.6.3

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[alsa-devel] [PATCH] sound/pci/Kconfig: update for ice1712/24
by Alexander Beregalov 11 Aug '08

11 Aug '08

From: Alexander Beregalov <a.beregalov(a)gmail.com> sound/pci/Kconfig: update for ice1712/24 Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/Kconfig | 14 +++++++++----- 1 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index f7d95b2..e14dc54 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -649,8 +649,9 @@ config SND_ICE1712 Currently supported hardware is: M-Audio Delta 1010(LT), DiO 2496, 66, 44, 410, Audiophile 24/96; Digigram VX442; - TerraTec EWX 24/96, EWS 88MT, 88D, DMX 6Fire, Phase 88; - Hoontech SoundTrack DSP 24/Value/Media7.1; Event EZ8. + TerraTec EWX 24/96, EWS 88MT/D, DMX 6Fire, Phase 88; + Hoontech SoundTrack DSP 24/Value/Media7.1; Event EZ8; + Lionstracs Mediastation, Terrasoniq TS 88. To compile this driver as a module, choose M here: the module will be called snd-ice1712. @@ -665,9 +666,12 @@ config SND_ICE1724 ICE/VT1724/1720 (Envy24HT/PT) chips. Currently supported hardware is: AMP AUDIO2000; M-Audio - Revolution 7.1; TerraTec Aureon 5.1 Sky, 7.1 Space/Universe; - AudioTrak Prodigy 7.1; Pontis MS300; Albatron K8X800 Pro II; - Chaintech ZNF3-150/250. + Revolution 5.1, 7.1, Audiophile 192; TerraTec Aureon 5.1 Sky, + 7.1 Space/Universe, Phase 22/28; Onkyo SE-90PCI, SE-200PCI; + AudioTrak Prodigy 192, 7.1 (HIFI/LT/XT), HD2; Hercules + Fortissimo IV; ESI Juli@; Pontis MS300; EGO-SYS WaveTerminal + 192M; Albatron K8X800 Pro II; Chaintech ZNF3-150/250, 9CJS, + AV-710; Shuttle SN25P. To compile this driver as a module, choose M here: the module will be called snd-ice1724.

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[alsa-devel] [PATCH] ice1724/revo: simple clean up
by Alexander Beregalov 11 Aug '08

11 Aug '08

From: Alexander Beregalov <a.beregalov(a)gmail.com> ice1724/revo: simple clean up Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/ice1712/revo.c | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 4d26314..5e1b156 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -524,16 +524,20 @@ static int __devinit revo_init(struct snd_ice1712 *ice) ak = ice->akm = kcalloc(2, sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ak) return -ENOMEM; - ice->akm_codecs = 2; switch (ice->eeprom.subvendor) { case VT1724_SUBDEVICE_REVOLUTION71: ice->akm_codecs = 2; - if ((err = snd_ice1712_akm4xxx_init(ak, &akm_revo_front, &akm_revo_front_priv, ice)) < 0) + err = snd_ice1712_akm4xxx_init(ak, &akm_revo_front, + &akm_revo_front_priv, ice); + if (err < 0) return err; - if ((err = snd_ice1712_akm4xxx_init(ak + 1, &akm_revo_surround, &akm_revo_surround_priv, ice)) < 0) + err = snd_ice1712_akm4xxx_init(ak+1, &akm_revo_surround, + &akm_revo_surround_priv, ice); + if (err < 0) return err; /* unmute all codecs */ - snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, VT1724_REVO_MUTE); + snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, + VT1724_REVO_MUTE); break; case VT1724_SUBDEVICE_REVOLUTION51: ice->akm_codecs = 2;

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Re: [alsa-devel] [PATCH 2.6.26.2] hda_intel: ALSA HD Audio patch for Intel Ibex Peak DeviceIDs
by Takashi Iwai 11 Aug '08

11 Aug '08

At Fri, 8 Aug 2008 15:56:39 -0700, Seth Heasley wrote: > > This patch adds the Intel Ibex Peak (PCH) HD Audio Controller DeviceIDs. > > Signed-off by: Seth Heasley <seth.heasley(a)intel.com> Your embedded patch couldn't be applied properly, maybe because of broken spaces. Check your MUA setting. Anyway, I applied it manually. Thanks! Takashi > > --- linux-2.6.26.2/sound/pci/hda/hda_intel.c.orig 2008-08-08 11:25:05.000000000 -0700 > +++ linux-2.6.26.2/sound/pci/hda/hda_intel.c 2008-08-08 11:27:24.000000000 -0700 > @@ -98,6 +98,7 @@ > "{Intel, ICH8}," > "{Intel, ICH9}," > "{Intel, ICH10}," > + "{Intel, PCH}," > "{Intel, SCH}," > "{ATI, SB450}," > "{ATI, SB600}," > @@ -2181,6 +2182,8 @@ > { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, > { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, > { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, > + /* PCH */ > + { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, > /* ATI SB 450/600 */ > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

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[alsa-devel] [PATCH] powerpc/drivers: use linux/of_device.h instead of asm/of_device.h
by Stephen Rothwell 11 Aug '08

11 Aug '08

Signed-off-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- drivers/hwmon/ams/ams.h | 2 +- sound/aoa/soundbus/soundbus.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) These are the last two users of asm/of_device.h. If everyone is happy, we can put this through the powerpc tree. I have build tested for ppc64_defconfig and ppc6xx_defconfig (which uses the relevant drivers). After this we can remove the include of linux/of_device.h from asm/of_device.h diff --git a/drivers/hwmon/ams/ams.h b/drivers/hwmon/ams/ams.h index a6221e5..221ef69 100644 --- a/drivers/hwmon/ams/ams.h +++ b/drivers/hwmon/ams/ams.h @@ -4,7 +4,7 @@ #include <linux/mutex.h> #include <linux/spinlock.h> #include <linux/types.h> -#include <asm/of_device.h> +#include <linux/of_device.h> enum ams_irq { AMS_IRQ_FREEFALL = 0x01, diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h index 622cd37..a0f223c 100644 --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -8,7 +8,7 @@ #ifndef __SOUNDBUS_H #define __SOUNDBUS_H -#include <asm/of_device.h> +#include <linux/of_device.h> #include <sound/pcm.h> #include <linux/list.h> -- 1.5.6.3 -- Cheers, Stephen Rothwell sfr(a)canb.auug.org.au http://www.canb.auug.org.au/~sfr/

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[alsa-devel] clicking/noise when upsampling with ESI juli@
by Amaury De Ganseman 10 Aug '08

10 Aug '08

Hi, There's a problem when I start to play a file the sound is bad during 1sec or less (clicking,cracking,...) or when I switch between tracks I hear clicks. Any idea ? Note: It works very well with OSS driver (last mercurial) Thanks for help. Here my asoundrc: defaults.pcm.rate_converter "speexrate_best" pcm.!default { type plug slave.pcm { type dmix ipc_key 1478 slave { pcm "hw:0,1" format S32_LE #period_time 0 period_size 2048 buffer_size 8192 rate 48000 } } #rate_converter "speexrate_best" #rate_converter "samplerate_best" }

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