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[alsa-devel] ASoC suspend/resume
by Dmitry Baryshkov 12 Aug '08

12 Aug '08

Hi, I'm trying to clean up spitz ASoC driver: I need to put some GPIO pins to low during suspend and restore them during resume. What is the most logical place to do that? I can't find any suitable suspend hook. -- With best wishes Dmitry

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[alsa-devel] [PATCH] spitz: MONO -> MONO1
by Dmitry Baryshkov 12 Aug '08

12 Aug '08

Correct route name to be MONO1 instead of MONO to follow recent fix in wm8750. 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 eefc25b..37cb768 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -297,7 +297,7 @@ 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] HDA-Intel - Patch to support new AMD HDMI Audio
by Yang, Libin 12 Aug '08

12 Aug '08

Hi Takashi, This patch is to support new HDMI audio device. Would you please review it? Thanks a lot! --------------------------------------------------------------------- Signed-off-by: Libin Yang <libin.yang(a)amd.com> --- alsa-driver-1.0.17/alsa-kernel/pci/hda/hda_intel.c.orig 2008-08-12 15:33:37.000000000 +0800 +++ alsa-driver-1.0.17/alsa-kernel/pci/hda/hda_intel.c 2008-08-12 15:34:30.000000000 +0800 @@ -2268,6 +2268,7 @@ { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI }, Thanks & Regards, Libin

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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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