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[alsa-devel] Request for Device ID addition : 6935G
by Karthik Ramgopal 14 Jun '09

14 Jun '09

Hi, Acer Aspire 6935G sound system is confirmed to be functional with model=acer-aspire-8930g. Kindly add device ID 0x1025 0x0146 to the quirks list in patch_realtek.c . Jack detetction and playback is working without any hda-verbing. Just the internal mic seems to be a problem. Can't hear my voice, can just hear static when I playback a recording. Anyways big thanks to Hector and Takashi for getting it all to work. Regards, Karthik

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Re: [alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Jean-Pierre André 14 Jun '09

14 Jun '09

Hi, > At Fri, 12 Jun 2009 15:01:09 -0300, > Gustavo Maciel Dias Vieira wrote: > >> > >> > Em Qui, 2009-06-11 às 21:27 +0200, Takashi Iwai escreveu: >> >>>> > > > Interrupt 22 stays always with 0. Even when correctly playing >>>> >>> > > sounds. I >>> >>>> > > > tested this on another machine with HD-Audio, and the equivalent >>>> > > > interrupt increased when playing. Also, in this other system >>>> >>> > > interrupt 0 >>> >>>> > > > (timer) is hardly issued, but in this laptop it is firing all the >>>> >>> > > time. >>> >>>> > > > >>>> > > > How can the driver be working without interrupts? >>>> >>> > > >>> > > That's weird. >>> > > >>> > > Did you try enable_msi=1 module option, BTW? >>> >> > >> > Just tried now. The interrupts started flowing and, tada, jack sense >> > works! >> > >> > Thanks, Takashi, that solves it for me. Is there a way to automate this >> > in the driver for the other less technically inclined users? >> > > Well, as I mentioned, it's basically a bug in the lower layer like > ACPI or BIOS. So, the best is to fix them. Using MSI is a kind of > workaround, and I'm wondering whether this is specific to your device > or in general with other HP dv4, since I didn't hear of the similar > bug reports. > > > Takashi > Weird indeed. Actually I had already reported this behaviour, see http://article.gmane.org/gmane.linux.alsa.devel/59809 So far I have not been able to get the interrupts at IRQ22. They still occur at IRQ20. However the problem is hidden when using boot options irqpoll or noapic. This is not a hardware or bios problem, as the sound is correct on Windows. Initially I was using kernel 2.6.27 and alsa-driver 1.0.19, currently using kernel 2.6.29 and alsa-driver 1.0.20 with the same behaviour. I have tried numerous modprobe options. Currently I am using : alias snd-card-0 snd-hda-intel options sound slots=snd-hda-intel options snd-hda-intel model=hp-m4 enable_msi=1 bdl_pos_adj=32 probe_mask=0x1ff As Gustavo seems to have succeeded, can he post kernel and alsa-driver versions, boot parameters and modprobe parameters ? Thanks in advance Jean-Pierre

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[alsa-devel] [RFC PATCH] ASoC: au1x: platform driver restructuring
by Manuel Lauss 13 Jun '09

13 Jun '09

Hi Mark! Please have a look at the patch below and verify it does what you suggested earlier, and if you have no objections, add it to your patchqueue. Both the AC97 and I2S modules are converted to full platform drivers; the PCM (DMA) interface is initialized during platform probe. Run-tested on DB1200 board. Thank you! Manuel Lauss --- From: Manuel Lauss <manuel.lauss(a)gmail.com> Subject: [PATCH] ASoC: au1x: platform driver restructuring Make the AC97/I2S modules full platform drivers and update the sample machine code. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- sound/soc/au1x/dbdma2.c | 24 +++++----- sound/soc/au1x/psc-ac97.c | 103 +++++++++++++++++++++++++----------------- sound/soc/au1x/psc-i2s.c | 103 ++++++++++++++++++++++++------------------ sound/soc/au1x/psc.h | 7 ++- sound/soc/au1x/sample-ac97.c | 48 +++++++++++-------- 5 files changed, 165 insertions(+), 120 deletions(-) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 594c6c5..10d21ec 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -331,7 +331,7 @@ static int au1xpsc_pcm_new(struct snd_card *card, return 0; } -static int au1xpsc_pcm_probe(struct platform_device *pdev) +int au1xpsc_asoc_dbdma_init(struct platform_device *pdev) { struct resource *r; int ret; @@ -375,8 +375,9 @@ out1: au1xpsc_audio_pcmdma[PCM_TX] = NULL; return ret; } +EXPORT_SYMBOL_GPL(au1xpsc_asoc_dbdma_init); -static int au1xpsc_pcm_remove(struct platform_device *pdev) +void au1xpsc_asoc_dbdma_exit(void) { int i; @@ -387,36 +388,33 @@ static int au1xpsc_pcm_remove(struct platform_device *pdev) au1xpsc_audio_pcmdma[i] = NULL; } } - - return 0; } +EXPORT_SYMBOL_GPL(au1xpsc_asoc_dbdma_exit); /* au1xpsc audio platform */ struct snd_soc_platform au1xpsc_soc_platform = { .name = "au1xpsc-pcm-dbdma", - .probe = au1xpsc_pcm_probe, - .remove = au1xpsc_pcm_remove, .pcm_ops = &au1xpsc_pcm_ops, .pcm_new = au1xpsc_pcm_new, .pcm_free = au1xpsc_pcm_free_dma_buffers, }; EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); -static int __init au1xpsc_audio_dbdma_init(void) +static int __init au1xpsc_audio_dbdma_load(void) { au1xpsc_audio_pcmdma[PCM_TX] = NULL; au1xpsc_audio_pcmdma[PCM_RX] = NULL; return snd_soc_register_platform(&au1xpsc_soc_platform); } -static void __exit au1xpsc_audio_dbdma_exit(void) +static void __exit au1xpsc_audio_dbdma_unload(void) { snd_soc_unregister_platform(&au1xpsc_soc_platform); } -module_init(au1xpsc_audio_dbdma_init); -module_exit(au1xpsc_audio_dbdma_exit); +module_init(au1xpsc_audio_dbdma_load); +module_exit(au1xpsc_audio_dbdma_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..ee8e642 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -236,8 +236,32 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, return ret; } -static int au1xpsc_ac97_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + +struct snd_soc_dai au1xpsc_ac97_dai = { + .name = "au1xpsc_ac97", + .ac97_control = 1, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &au1xpsc_ac97_dai_ops, +}; +EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); + + +static int __devinit au1xpsc_ac97_probe(struct platform_device *pdev) { int ret; struct resource *r; @@ -287,8 +311,18 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, * there may not be any codec clock yet. */ - return 0; + ret = au1xpsc_asoc_dbdma_init(pdev); + if (ret) + goto out2; + + ret = snd_soc_register_dai(&au1xpsc_ac97_dai); + if (!ret) + return 0; + + au1xpsc_asoc_dbdma_exit(); +out2: + iounmap(au1xpsc_ac97_workdata->mmio); out1: release_resource(au1xpsc_ac97_workdata->ioarea); kfree(au1xpsc_ac97_workdata->ioarea); @@ -298,9 +332,11 @@ out0: return ret; } -static void au1xpsc_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_ac97_remove(struct platform_device *pdev) { + snd_soc_unregister_dai(&au1xpsc_ac97_dai); + au1xpsc_asoc_dbdma_exit(); + /* disable PSC completely */ au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); au_sync(); @@ -312,9 +348,11 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev, kfree(au1xpsc_ac97_workdata->ioarea); kfree(au1xpsc_ac97_workdata); au1xpsc_ac97_workdata = NULL; + + return 0; } -static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) +static int au1xpsc_ac97_suspend(struct platform_device *pdev, pm_message_t m) { /* save interesting registers and disable PSC */ au1xpsc_ac97_workdata->pm[0] = @@ -328,7 +366,7 @@ static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) return 0; } -static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) +static int au1xpsc_ac97_resume(struct platform_device *pdev) { /* restore PSC clock config */ au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, @@ -342,48 +380,31 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, -}; - -struct snd_soc_dai au1xpsc_ac97_dai = { - .name = "au1xpsc_ac97", - .ac97_control = 1, - .probe = au1xpsc_ac97_probe, - .remove = au1xpsc_ac97_remove, - .suspend = au1xpsc_ac97_suspend, - .resume = au1xpsc_ac97_resume, - .playback = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, +static struct platform_driver au1xpsc_ac97_driver = { + .driver = { + .name = "au1xpsc_ac97", + .owner = THIS_MODULE, }, - .capture = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, - }, - .ops = &au1xpsc_ac97_dai_ops, + .probe = au1xpsc_ac97_probe, + .remove = __devexit_p(au1xpsc_ac97_remove), + .suspend = au1xpsc_ac97_suspend, + .resume = au1xpsc_ac97_resume, }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); -static int __init au1xpsc_ac97_init(void) +static int __init au1xpsc_ac97_load(void) { au1xpsc_ac97_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_ac97_dai); + return platform_driver_register(&au1xpsc_ac97_driver); } -static void __exit au1xpsc_ac97_exit(void) +static void __exit au1xpsc_ac97_unload(void) { - snd_soc_unregister_dai(&au1xpsc_ac97_dai); + platform_driver_unregister(&au1xpsc_ac97_driver); } -module_init(au1xpsc_ac97_init); -module_exit(au1xpsc_ac97_exit); +module_init(au1xpsc_ac97_load); +module_exit(au1xpsc_ac97_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index bb58932..f830f5a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -262,8 +262,31 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int au1xpsc_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + +struct snd_soc_dai au1xpsc_i2s_dai = { + .name = "au1xpsc_i2s", + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = &au1xpsc_i2s_dai_ops, +}; +EXPORT_SYMBOL(au1xpsc_i2s_dai); + +static int au1xpsc_i2s_probe(struct platform_device *pdev) { struct resource *r; unsigned long sel; @@ -294,9 +317,7 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev, if (!au1xpsc_i2s_workdata->mmio) goto out1; - /* preserve PSC clock source set up by platform (dev.platform_data - * is already occupied by soc layer) - */ + /* preserve PSC clock source set up by platform */ sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); au_sync(); @@ -312,9 +333,16 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev, * be running yet; depending on clock input for PSC a wait might * time out. */ + ret = au1xpsc_asoc_dbdma_init(pdev); + if (ret) + goto out2; - return 0; + ret = snd_soc_register_dai(&au1xpsc_i2s_dai); + if (!ret) + return 0; +out2: + iounmap(au1xpsc_i2s_workdata->mmio); out1: release_resource(au1xpsc_i2s_workdata->ioarea); kfree(au1xpsc_i2s_workdata->ioarea); @@ -324,9 +352,11 @@ out0: return ret; } -static void au1xpsc_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_i2s_remove(struct platform_device *pdev) { + snd_soc_unregister_dai(&au1xpsc_i2s_dai); + au1xpsc_asoc_dbdma_exit(); + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); au_sync(); au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); @@ -337,9 +367,11 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev, kfree(au1xpsc_i2s_workdata->ioarea); kfree(au1xpsc_i2s_workdata); au1xpsc_i2s_workdata = NULL; + + return 0; } -static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_suspend(struct platform_device *pdev, pm_message_t m) { /* save interesting register and disable PSC */ au1xpsc_i2s_workdata->pm[0] = @@ -353,7 +385,7 @@ static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) return 0; } -static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_resume(struct platform_device *pdev) { /* select I2S mode and PSC clock */ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); @@ -367,48 +399,31 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) return 0; } -static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, -}; - -struct snd_soc_dai au1xpsc_i2s_dai = { - .name = "au1xpsc_i2s", - .probe = au1xpsc_i2s_probe, - .remove = au1xpsc_i2s_remove, - .suspend = au1xpsc_i2s_suspend, - .resume = au1xpsc_i2s_resume, - .playback = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ +static struct platform_driver au1xpsc_i2s_driver = { + .driver = { + .name = "au1xpsc_i2s", + .owner = THIS_MODULE, }, - .capture = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ - }, - .ops = &au1xpsc_i2s_dai_ops, + .probe = au1xpsc_i2s_probe, + .remove = __devexit_p(au1xpsc_i2s_remove), + .suspend = au1xpsc_i2s_suspend, + .resume = au1xpsc_i2s_resume, }; -EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_init(void) +static int __init au1xpsc_i2s_load(void) { au1xpsc_i2s_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_i2s_dai); + return platform_driver_register(&au1xpsc_i2s_driver); } -static void __exit au1xpsc_i2s_exit(void) +static void __exit au1xpsc_i2s_unload(void) { - snd_soc_unregister_dai(&au1xpsc_i2s_dai); + platform_driver_unregister(&au1xpsc_i2s_driver); } -module_init(au1xpsc_i2s_init); -module_exit(au1xpsc_i2s_exit); +module_init(au1xpsc_i2s_load); +module_exit(au1xpsc_i2s_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a0..9ee4f59 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -21,6 +21,9 @@ extern struct snd_soc_dai au1xpsc_i2s_dai; extern struct snd_soc_platform au1xpsc_soc_platform; extern struct snd_ac97_bus_ops soc_ac97_ops; +extern int au1xpsc_asoc_dbdma_init(struct platform_device *pdev); +extern void au1xpsc_asoc_dbdma_exit(void); + struct au1xpsc_audio_data { void __iomem *mmio; diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c index 27683eb..86380f6 100644 --- a/sound/soc/au1x/sample-ac97.c +++ b/sound/soc/au1x/sample-ac97.c @@ -1,7 +1,7 @@ /* * Sample Au12x0/Au1550 PSC AC97 sound machine. * - * Copyright (c) 2007-2008 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * Copyright (c) 2007-2009 Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms outlined in the file COPYING at the root of this @@ -44,13 +44,13 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { static struct snd_soc_card au1xpsc_sample_ac97_machine = { .name = "Au1xxx PSC AC97 Audio", + .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ .dai_link = &au1xpsc_sample_ac97_dai, .num_links = 1, }; static struct snd_soc_device au1xpsc_sample_ac97_devdata = { .card = &au1xpsc_sample_ac97_machine, - .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ .codec_dev = &soc_codec_dev_ac97, }; @@ -82,7 +82,14 @@ static struct resource au1xpsc_psc1_res[] = { }, }; -static struct platform_device *au1xpsc_sample_ac97_dev; +static struct platform_device psc_ac97_dev = { + .name = "au1xpsc_ac97", + .id = 0, + .resource = au1xpsc_psc1_res, + .num_resources = ARRAY_SIZE(au1xpsc_psc1_res), +}; + +static struct platform_device *au1xpsc_sample_asoc_dev; static int __init au1xpsc_sample_ac97_load(void) { @@ -99,41 +106,42 @@ static int __init au1xpsc_sample_ac97_load(void) au_sync(); #endif - ret = -ENOMEM; - /* setup PSC clock source for AC97 part: external clock provided * by codec. The psc-ac97.c driver depends on this setting! */ au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); au_sync(); - au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); - if (!au1xpsc_sample_ac97_dev) - goto out; + /* register the PSC-AC97 platform device */ + ret = platform_device_register(&psc_ac97_dev); + if (ret) + return -ENODEV; - au1xpsc_sample_ac97_dev->resource = - kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * - ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); - au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); - au1xpsc_sample_ac97_dev->id = 1; + /* register machine fabric with ASoC core */ + ret = -ENOMEM; + au1xpsc_sample_asoc_dev = platform_device_alloc("soc-audio", -1); + if (!au1xpsc_sample_asoc_dev) + goto out; - platform_set_drvdata(au1xpsc_sample_ac97_dev, + platform_set_drvdata(au1xpsc_sample_asoc_dev, &au1xpsc_sample_ac97_devdata); - au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; - ret = platform_device_add(au1xpsc_sample_ac97_dev); + au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_asoc_dev->dev; + ret = platform_device_add(au1xpsc_sample_asoc_dev); if (ret) { - platform_device_put(au1xpsc_sample_ac97_dev); - au1xpsc_sample_ac97_dev = NULL; + platform_device_put(au1xpsc_sample_asoc_dev); + au1xpsc_sample_asoc_dev = NULL; } out: + platform_device_unregister(&psc_ac97_dev); return ret; } static void __exit au1xpsc_sample_ac97_exit(void) { - platform_device_unregister(au1xpsc_sample_ac97_dev); + platform_device_unregister(au1xpsc_sample_asoc_dev); + platform_device_unregister(&psc_ac97_dev); } module_init(au1xpsc_sample_ac97_load); @@ -141,4 +149,4 @@ module_exit(au1xpsc_sample_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); -- 1.6.3.1

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[PATCH] ASoC: Add stub suspend and resume calls for ASoC subdevices
by Mark Brown 13 Jun '09

13 Jun '09

Now that ASoC subdevices can be regular devices they can have normal suspend and resume calls from their buses. However, suspending them individually is not desirable since this can lead to problems such as pops and clicks from devices being suspended with their signals being amplified or clocks being stopped suddenly. This will be resolved by having the normal device model suspend and resume calls call into ASoC which will suspend the entire card while any of its components are suspended. At present this is not yet implemented but in order to aid the transition of drivers to the standard device model this patch adds API calls for the notifications. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 39 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5297ba7..e6704c0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -192,6 +192,11 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); +#ifdef CONFIG_PM +int snd_soc_suspend_device(struct device *dev); +int snd_soc_resume_device(struct device *dev); +#endif + /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..4414117 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -788,6 +788,45 @@ static int soc_resume(struct platform_device *pdev) return 0; } +/** + * snd_soc_suspend_device: Notify core of device suspend + * + * @dev: Device being suspended. + * + * In order to ensure that the entire audio subsystem is suspended in a + * coordinated fashion ASoC devices should suspend themselves when + * called by ASoC. When the standard kernel suspend process asks the + * device to suspend it should call this function to initiate a suspend + * of the entire ASoC card. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_suspend_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_suspend_device); + +/** + * snd_soc_resume_device: Notify core of device resume + * + * @dev: Device being resumed. + * + * In order to ensure that the entire audio subsystem is resumed in a + * coordinated fashion ASoC devices should resume themselves when called + * by ASoC. When the standard kernel resume process asks the device + * to resume it should call this function. Once all the components of + * the card have notified that they are ready to be resumed the card + * will be resumed. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_resume_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_resume_device); + #else #define soc_suspend NULL #define soc_resume NULL -- 1.6.3.1

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Re: [alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Takashi Iwai 13 Jun '09

13 Jun '09

At Sat, 18 Apr 2009 21:36:35 -0300, Gustavo Vieira wrote: > > Em Sáb, 2009-04-18 às 10:57 +0200, Takashi Iwai escreveu: > > > The jack sense is still broken, though. I'm attaching the alsa-info > > with > > > the snapshot loaded. > > > > Could you attach alsa-info.sh output at both HP plugged / unplugged > > states? > > > > Now things got really interesting. :) > > When trying to get the output you requested I discovered that running > alsa-info with the headphones plugged made them work and silenced the > speakers. Then, removing the HP do not make the sound come back to the > speakers, but running alsa-info again does. That is, I can force the > jack sense by running alsa-info. > > I'm attaching the following files: > * alsa-info-snapshot.txt: The output when the system is turned on, > without HP. When I plug the HP and run alsa-info, the HP work > and the out is the same as this file. > * alsa-info-snapshot-hp-1.txt: The output when HP work. > * alsa-info-snapshot-hp-out.txt: The output of the execution of > alsa-info that makes the speakers work when HP are removed. Hm, what happens if you pass model=hp-dv5 option? Takashi

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[alsa-devel] [PATCH] ASoC: Zoom2: Add machine driver for Zoom2 board
by Lopez Cruz, Misael 12 Jun '09

12 Jun '09

Add support for Zoom2 board. Zoom2 machine driver connects both codec DAIs (audio and voice) to omap3 McBSP ports in the following way: HiFi <-> McBSP2 Voice <-> McBSP3 The zoom2 driver has the following DAPM widgets: * Ext Mic: MAINMIC, SUBMIC (with bias) * Ext Spk: HFL, HFR * Headset Stereophone: HSOL, HSOR * Headset Mic: HSMIC (with bias) * Aux In: AUXL, AUXR Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/Kconfig | 7 + sound/soc/omap/Makefile | 2 + sound/soc/omap/zoom2.c | 301 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 310 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/zoom2.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 675732e..397f3d8 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -64,4 +64,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE help Say Y if you want to add support for SoC audio on the Beagleboard. +config SND_OMAP_SOC_ZOOM2 + tristate "SoC Audio support for Zoom2" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for Soc audio on Zoom2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0c9e4ac..5016840 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -13,6 +13,7 @@ snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o @@ -21,3 +22,4 @@ obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o +obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c new file mode 100644 index 0000000..be2e307 --- /dev/null +++ b/sound/soc/omap/zoom2.c @@ -0,0 +1,301 @@ +/* + * zoom2.c -- SoC audio for Zoom2 + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15) + +static int zoom2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops zoom2_ops = { + .hw_params = zoom2_hw_params, +}; + +static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFM); + if (ret) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops zoom2_voice_ops = { + .hw_params = zoom2_hw_voice_params, +}; + +/* Zoom2 machine DAPM */ +static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Aux In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Mic Bias 1"}, + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 1", NULL, "Ext Mic"}, + {"Mic Bias 2", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Stereophone: HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Aux In: AUXL, AUXR */ + {"Aux In", NULL, "AUXL"}, + {"Aux In", NULL, "AUXR"}, +}; + +static int zoom2_twl4030_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add Zoom2 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets, + ARRAY_SIZE(zoom2_twl4030_dapm_widgets)); + if (ret) + return ret; + + /* Set up Zoom2 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* Zoom2 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); + snd_soc_dapm_enable_pin(codec, "Aux In"); + + /* TWL4030 not connected pins */ + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); + + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + +static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Enable voice interface */ + reg = codec->read(codec, TWL4030_REG_VOICE_IF); + reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN; + codec->write(codec, TWL4030_REG_VOICE_IF, reg); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link zoom2_dai[] = { + { + .name = "TWL4030 I2S", + .stream_name = "TWL4030 Audio", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .init = zoom2_twl4030_init, + .ops = &zoom2_ops, + }, + { + .name = "TWL4030 PCM", + .stream_name = "TWL4030 Voice", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE], + .init = zoom2_twl4030_voice_init, + .ops = &zoom2_voice_ops, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_zoom2 = { + .name = "Zoom2", + .platform = &omap_soc_platform, + .dai_link = zoom2_dai, + .num_links = ARRAY_SIZE(zoom2_dai), +}; + +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 2, /* 81 ms */ + .sysclk = 26000, +}; + +/* Audio subsystem */ +static struct snd_soc_device zoom2_snd_devdata = { + .card = &snd_soc_zoom2, + .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, +}; + +static struct platform_device *zoom2_snd_device; + +static int __init zoom2_soc_init(void) +{ + int ret; + + if (!machine_is_omap_zoom2()) { + pr_debug("Not Zoom2!\n"); + return -ENODEV; + } + printk(KERN_INFO "Zoom2 SoC init\n"); + + zoom2_snd_device = platform_device_alloc("soc-audio", -1); + if (!zoom2_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata); + zoom2_snd_devdata.dev = &zoom2_snd_device->dev; + *(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + *(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + + ret = platform_device_add(zoom2_snd_device); + if (ret) + goto err1; + + BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0); + gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(zoom2_snd_device); + + return ret; +} +module_init(zoom2_soc_init); + +static void __exit zoom2_soc_exit(void) +{ + gpio_free(ZOOM2_HEADSET_MUX_GPIO); + + platform_device_unregister(zoom2_snd_device); +} +module_exit(zoom2_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC Zoom2"); +MODULE_LICENSE("GPL"); + -- 1.5.4.3

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[alsa-devel] snd-ctxfi surround volume
by The Source 12 Jun '09

12 Jun '09

Hello again. I have another minor problem with this driver. When I use hw:0,1 (surround) device for output, the volume even with slider on maximum is much lower than through front (hw:0,0) device. Can you do something about it? Thank you.

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Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted:
by bill＠tellytopia.com 12 Jun '09

12 Jun '09

Looking at the board, it is model SB1040, with CA0110-IBG LF chip. Is that an hda type chip? > -------- Original Message -------- > Subject: Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted: > From: Takashi Iwai <tiwai(a)suse.de> > Date: Thu, June 11, 2009 6:54 pm > To: bill(a)tellytopia.com > Cc: "Gernot Kohlhaas" <gernot.kohlhaas(a)googlemail.com>, > alsa-devel(a)alsa-project.org > At Thu, 11 Jun 2009 14:28:02 -0700, > bill(a)tellytopia.com wrote: > > > > Yes, I have loaded the module snd-ctfxi many times, > > and unloaded snd-hda-intl. > > Also blacklisted snd-hda-intel, and put snd-ctxfi into modprobe.conf. > Are you sure that your board has really emu20k1 or emu20k2 chip? > X-Fi isn't always with these chips. Some are HD-audio only (thus > snd-hda-intel is the right driver) and some are with CA0106 chip. > Takashi

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[alsa-devel] Backported sbxfi driver (UNTESTED!)
by Takashi Iwai 12 Jun '09

12 Jun '09

Hi, $SUBJECT is now on my sound-unstable git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git together with other experimental patches. If you're using 2.6.27-rc* git tree, pull the master branch of the tree above into yours, and run make oldconfig. That is, % cd /your/git-tree % git pull git://...../sound-unstable-2.6.git master If you are not using 2.6.27-rc*, or not familiar with git, you can try alsa-driver-unstable snapshot tarball available at ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ Run configure, make and make install as usual ALSA-driver tarball. (I didn't check whether the tarball correctly includes the sbxfi stuff. It should have been generated automatically. If not, wait for a while. If it still doesn't include sbxfi code, please report. I'll fix it tomorrow morning.) The driver is built only for 2.6.26 or later. If you have an older kernel, edit alsa-driver*/kconfig-vers and change the version of CONFIG_SND_SBXFI to 2.6.24 or whatever you want. Then run ./gitcompile, instead of configure in this case to update the configure script. **NOTE** The driver is totally untested. It's just compiled without errors, but not reviewed after a quick writing. So, don't expect it ever runs at the first try. A crash is highly possible. There are some build conditions found in sound/pci/sbxfi/sbxfi.c, starting with XXX_*. You can change it if you want. As default, it's for non-fullduplex but accept different rates. Not sure whether this works at all. Any test- (and better debugging-) reports are appreciated. thanks, Takashi

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Re: [alsa-devel] plug:dmix slows playback
by Jaroslav Kysela 12 Jun '09

12 Jun '09

On Fri, 12 Jun 2009, Steve Langstaff wrote: >> From: alsa-devel-bounces(a)alsa-project.org [mailto:alsa-devel- >> bounces(a)alsa-project.org] On Behalf Of Jaroslav Kysela > >> On Fri, 12 Jun 2009, Steve Langstaff wrote: > >>> Slave: Hardware PCM card 0 'LM4549' device 0 subdevice 0 >> >>> msbits : 16 >> >> If I remember correctly, this value was 32 in your dmix setup. >> Could you try command: >> >> aplay -D"plug:'dmix:FORMAT=S16_LE'" <filename> > > Yes, that works fine - is there something wrong with my configuration do you > think? It seems that 32-bit samples are not handled correctly in driver. What's your hardware? Output from alsa-info.sh script might be helpful for identification. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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