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[alsa-devel] [PATCH] ASoC: Fix build error in sound/soc/kirkwood/kirkwood-i2s.c
by Axel Lin 28 Dec '11

28 Dec '11

Since commit db33f4de "ARM: Orion: Remove address map info from all platform data structures", the dram is removed from struct kirkwood_asoc_platform_data. This patch fixes below build error: CC sound/soc/kirkwood/kirkwood-i2s.o sound/soc/kirkwood/kirkwood-i2s.c: In function 'kirkwood_i2s_dev_probe': sound/soc/kirkwood/kirkwood-i2s.c:444: error: 'struct kirkwood_asoc_platform_data' has no member named 'dram' sound/soc/kirkwood/kirkwood-i2s.c:450: error: 'struct kirkwood_asoc_platform_data' has no member named 'dram' make[3]: *** [sound/soc/kirkwood/kirkwood-i2s.o] Error 1 make[2]: *** [sound/soc/kirkwood] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Andrew Lunn <andrew(a)lunn.ch> Cc: Nicolas Pitre <nico(a)fluxnic.net> --- This patch is against linux-next tree. sound/soc/kirkwood/kirkwood-i2s.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index f6bb211..3cb9aa4 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -441,13 +441,12 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) goto err_ioremap; } - if (!data || !data->dram) { + if (!data) { dev_err(&pdev->dev, "no platform data ?!\n"); err = -EINVAL; goto err_ioremap; } - priv->dram = data->dram; priv->burst = data->burst; return snd_soc_register_dai(&pdev->dev, &kirkwood_i2s_dai); -- 1.7.5.4

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Re: [alsa-devel] [PATCH 3/3 v2] sound/soc/lapis: add platform driver for ML7213 IOH I2S
by Mark Brown 27 Dec '11

27 Dec '11

On Tue, Dec 20, 2011 at 11:45:44AM +0900, Tomoya MORINAGA wrote: > +static int ignore_overrun = 1; > +module_param(ignore_overrun, int, 0444); > +MODULE_PARM_DESC(ignore_overrun, "ignore RX overruns (default=0)"); This shouldn't be a driver specific thing, and if we were to have such a feature a module parameter doesn't seem like a great way of doing it. > +/***************************************************************************** > + * I2S HAL (Hardware Abstruction Layer) > + *****************************************************************************/ Please follow the kernel coding style in terms of comments. > +static void ioh_i2s_enable_interrupts(int ch, enum dma_data_direction dir) > +{ > + unsigned int intr_lines; > + > + if (dir) > + intr_lines = 1 << (I2S_IMASK_RX_BIT_START + ch); > + > + else > + intr_lines = 1 << (I2S_IMASK_TX_BIT_START + ch); I'd expect a switch statement corresponding to the enum, not an if statement. > +static void ioh_i2s_disable_interrupts(int ch, enum dma_data_direction dir) > +{ > + unsigned int intr_lines; > + > + /*intr_lines&=I2S_ALL_INTERRUPT_BITS; */ > + intr_lines = ioread32(i2s_data->iobase + I2SIMASK_OFFSET); What's this commented out code for? Also as a coding style thing you should have a space between /* and the text in the comment (this applies throughout the driver). > + /*disable interrupts for specified channel */ > + if (dir) > + intr_lines |= 1 << (I2S_IMASK_RX_BIT_START + ch); > + else > + intr_lines |= 1 << (I2S_IMASK_TX_BIT_START + ch); > + > + /*Mask the specific interrupt bits */ > + iowrite32(intr_lines, i2s_data->iobase + I2SIMASK_OFFSET); What ensures that this read/modify/write cycle can't race with another caller? > +/* Clear interrupt status */ > +static void ioh_i2s_clear_tx_sts_ir(int ch) > +{ > + int offset = ch * 0x800; > + > + iowrite32(I2S_TX_FINT | I2S_TX_AFINT | I2S_TX_EINT | I2S_TX_AEINT, > + i2s_data->iobase + I2SISTTX_OFFSET + offset); > +} This appears to unconditionally acknowledge all interrupts, generally you should only acknowledge interrupts that have been handled. Otherwise it's possible that interrupts may be dropped if they're flagged between the status read and acknowledgement write. > +/***************************************************************************** > + * I2S Middle ware > + *****************************************************************************/ I'm really not convinced of the value of these layers, reading the code it really feels incredibly verbose in comparison with what I'd expect from such a driver and I can't help that think that a lot of this verbosity is down to muddling through these abstraction layers. > +static bool filter(struct dma_chan *chan, void *slave) This needs a better name. It's not namespaced and it's not clear what it's filtering. > +static struct dma_chan *ioh_request_dma_channel( > + int ch, struct ioh_i2s_dma *dma, enum dma_data_direction dir) > +{ > + dma_cap_mask_t mask; > + struct dma_chan *chan; > + struct pci_dev *dma_dev; > + dma_cap_zero(mask); > + dma_cap_set(DMA_SLAVE, mask); > + > + dma_dev = pci_get_bus_and_slot(2, PCI_DEVFN(0, 1)); /* Get DMA's dev > + information */ > + > + if (dir == DMA_FROM_DEVICE) { /* Rx */ switch statement to select between multiple options in the enum. This applies throughout the code. > +static void > +ioh_i2s_write(struct snd_pcm_substream *substream, const void *data, int len) > +{ > + int rem1; > + int rem2; What is this supposed to do? There's an awful lot of it, and it looks like it's supposed to be rewriting the data format for some reason which isn't something that a driver ought to be doing (apart from anything else it does rather defeat the point of DMA). > +static void ioh_i2s_configure_i2s_regs(int ch, enum ioh_direction dir) > +{ > + int offset = ch * 0x800; You should have a function to map the channel into a number. > + > + if (dir) { > + /* Rx register */ > + iowrite32(I2S_AFULL_THRESH / 2, > + i2s_data->iobase + I2SAFRX_OFFSET + offset); > + iowrite32(0x1F, i2s_data->iobase + I2SAERX_OFFSET + offset); > + iowrite32(0x1F, i2s_data->iobase + I2SMSKRX_OFFSET + offset); > + iowrite32(0xC, i2s_data->iobase + I2SISTRX_OFFSET + offset); Lots of magic numbers here and given that the function is called just "configure" it's not clear what the intended purpose is. This needs to be clarified. > + /* Rx configuration */ > + if (atomic_read(&dma->rx_busy)) { > + dev_err(i2s_data->dev, "rx i2s%dalready opened\n", ch); ... > + } else { > + /* Tx configuration */ > + if (atomic_read(&dma->tx_busy)) { There's a *lot* of duplicate code in the driver for TX and RX, it should be possible to abstract out much more common code. > +static void i2s_tx_almost_empty_ir(int ch) > +{ > + struct dma_async_tx_descriptor *desc; > + int num; > + int tx_comp_index; > + struct ioh_i2s_dma *dma = &dmadata[ch]; > + struct scatterlist *sg = dma->sg_tx_p; > + void *cb_ch; > + > + dev_dbg(i2s_data->dev, "%s: data_head=%p data_complete=%p\n", __func__, > + dma->tx_data_head, dma->tx_complete); > + > + num = ((int)dma->tx_avail) / (I2S_AEMPTY_THRESH * 4); That case looks *very* suspect. > + tx_comp_index = (((int)(dma->tx_complete - dma->tx_head))) /\ > + (I2S_AEMPTY_THRESH * 4); There is very rarely any need for line continuations outside of macros. > +static inline void ioh_i2s_interrupt_sub_tx(int ch) > +{ > + unsigned int status; > + int offset = ch * 0x800; > + > + status = ioread32(i2s_data->iobase + I2SISTTX_OFFSET + offset); > + if (status & I2S_TX_EINT) > + i2s_tx_empty_ir(ch); > + if (status & I2S_TX_AEINT) > + i2s_tx_almost_empty_ir(ch); Given that all you're doing is logging just open code the log message. > + default: > + return -1; Return error codes, you EPERM almost certainly isn't the error you meant to report. > +static int ioh_i2s_pci_suspend(struct pci_dev *pdev, pm_message_t state) > +{ > + int ret; > + > + ioh_i2s_save_reg_conf(pdev); You should be doing this in ASoC suspend/resume callbacks, not in PCI ones.

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[alsa-devel] [PATCH 0/4] Amstrad Delta: access MODEM_RESET GPIO pin over a regulator
by Janusz Krzysztofik 27 Dec '11

27 Dec '11

The Amstrad Delta on-board latch2 bit named MODEM_NRESET, now available as a GPIO pin AMS_DELTA_GPIO_PIN_NMODEM_RESET, is used to power up/down (bring into/out of a reset state) two distinct on-board devices simultaneously: the modem, and the voice codec. As a consequence, that bit is, or can be, manipulated concurrently by two drivers, or their platform provided hooks. Instead of updating those drivers to use the gpiolib API as a new method of controlling the MODEM_NRESET pin state, like it was done to other drivers accessing latch2 pins, and still being vulnerable to potential concurrency conflicts, or trying to solve that sharing issue with a custom piece of code, set up a fixed regulator device on top of that GPIO pin and update both drivers to manipulate that regulator, not the GPIO pin directly. Janusz Krzysztofik (4): ARM: OMAP1: ams-delta: set up regulator over modem reset GPIO pin ASoC: cx20442: add bias control over a platform provided regulator ARM: OMAP1: ams-delta: update the modem to use regulator API ASoC: OMAP: ams-delta: drop .set_bias_level callback arch/arm/mach-omap1/Kconfig | 2 + arch/arm/mach-omap1/board-ams-delta.c | 116 +++++++++++++++++++-- arch/arm/plat-omap/include/plat/board-ams-delta.h | 1 - sound/soc/codecs/cx20442.c | 58 ++++++++++- sound/soc/omap/ams-delta.c | 34 ------ 5 files changed, 165 insertions(+), 46 deletions(-) -- 1.7.3.4

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[alsa-devel] Driver Soudcard/Midi BCD2000 Behringer
by William Martins 27 Dec '11

27 Dec '11

Good evening Messrs. Developers, at times I'm trying to make this device work in linux (Behringer BCD2000-), without success, that happens a few days I discovered that the manufacturer built the same non-standard midi, so it is not recognized by the kernel as a midi device. I realized that there is a need to build a drive for him, I am not a developer to develop it, the more I have some idea, after some research, that this site will find it a Sr, can develop it for mac, and made available on its website http : / / evinyatar.ath.cx / sphpblog / static.php? page = bcd2ktech some codes to serve as instruction for those who want to develop. Come and ask that if possible, you develop the driver for this device because the community waits anxiously for the same support. I look back. Thank you! William R. Martins IT Consultant / Support Specialist Tels: (98) 8197-3909 / 8884-9517

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Re: [alsa-devel] [PATCH] ASoC: Allocate PCM operations dynamically to support multiple DAIs
by Sangsu Park 27 Dec '11

27 Dec '11

> This looks wrong. It will probably cause regression for all non ASoC sound > card drivers. > > The issue this patch addresses came up before and I think the conclusion was > that it is best to embed the snd_pcm_ops struct into the snd_soc_pcm_runtime > struct. Hi, I'm Sangsu Park. (sangsu4u.park(a)samsung.com) I'm using gmail because of company's firewall problems... I agree you, So I've changed some code like follows. If you like that, I'll make patch version 2nd using that. //------------------------------------------------------------------------------ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 42ad2db..d1468aa 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -918,6 +918,9 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) list_del(&cpu_dai->card_list); module_put(cpu_dai->dev->driver->owner); } + + /* free the ops memory */ + if (rtd->ops) kfree(rtd->ops); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec..4eb793b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -641,22 +645,23 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); rtd->pcm = pcm; + rtd->ops = soc_pcm_ops; pcm->private_data = rtd; if (platform->driver->ops) { ------------------------------------------------------------------------------// Thanks.

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Re: [alsa-devel] [PATCH] ASoC: Allocate PCM operations dynamically to support multiple DAIs
by Sangsu Park 27 Dec '11

27 Dec '11

> On Fri, Dec 23, 2011 at 10:37:51AM +0900, Sangbeom Kim wrote: > > > + soc_pcm_ops = kzalloc(sizeof(*soc_pcm_ops), GFP_KERNEL); > > + if (soc_pcm_ops == NULL) { > > + snd_printk(KERN_ERR "Cannot allocate PCM OPS\n"); > > + return -ENOMEM; > > + } > > This patch is good and fixes a real problem but can you please change it > slightly so that instead of dynamically allocating the soc_pcm_ops we > just embed it directly in the runtime structure. Since every runtime > needs an ops structure we may as well just have it embedded directly and > not write the allocation, freeing and error handling code. Hi, I'm Sangsu Park. (sangsu4u.park(a)samsung.com) I'm using gmail because of company's firewall problems... I've implement your guide like follows. //-------------------------------------------------------------------------------------------------------------------------------- struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops = platform->driver->ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); --------------------------------------------------------------------------------------------------------------------------------// But, it seems to make mutex dead lock. When starting playback, it tells some messages. //-------------------------------------------------------------------------------------------------------------------------------- # aplay -twav Test.wav INFO: task aplay:1071 blocked for more than 120 seconds. "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. aplay D c01d1ba4 0 1071 1051 0x00000000 [<c01d1ba4>] (__schedule+0x244/0x6cc) from [<c01d2db0>] (__mutex_lock_slowpath+0x180/0x360) [<c01d2db0>] (__mutex_lock_slowpath+0x180/0x360) from [<c01d2f9c>] (mutex_lock+0xc/0x24) [<c01d2f9c>] (mutex_lock+0xc/0x24) from [<c01c10c4>] (soc_pcm_open+0x38/0x48c) [<c01c10c4>] (soc_pcm_open+0x38/0x48c) from [<c01c1114>] (soc_pcm_open+0x88/0x48c) [<c01c1114>] (soc_pcm_open+0x88/0x48c) from [<c01b0fcc>] (snd_pcm_open_substream+0x60/0xa0) [<c01b0fcc>] (snd_pcm_open_substream+0x60/0xa0) from [<c01b1134>] (snd_pcm_open+0x128/0x270) [<c01b1134>] (snd_pcm_open+0x128/0x270) from [<c01a4034>] (snd_open+0x130/0x2ec) [<c01a4034>] (snd_open+0x130/0x2ec) from [<c00b00e4>] (chrdev_open+0x10c/0x1f0) [<c00b00e4>] (chrdev_open+0x10c/0x1f0) from [<c00aad24>] (__dentry_open+0x220/0x328) [<c00aad24>] (__dentry_open+0x220/0x328) from [<c00aaef4>] (nameidata_to_filp+0x60/0x68) [<c00aaef4>] (nameidata_to_filp+0x60/0x68) from [<c00bb5d4>] (do_last+0x214/0xcc0) [<c00bb5d4>] (do_last+0x214/0xcc0) from [<c00bc13c>] (path_openat+0xbc/0x3a8) [<c00bc13c>] (path_openat+0xbc/0x3a8) from [<c00bc510>] (do_filp_open+0x30/0x84) [<c00bc510>] (do_filp_open+0x30/0x84) from [<c00aa954>] (do_sys_open+0xe4/0x184) [<c00aa954>] (do_sys_open+0xe4/0x184) from [<c000f3c0>] (ret_fast_syscall+0x0/0x30) --------------------------------------------------------------------------------------------------------------------------------// "sound/core/" uses soc_pcm_ops using substream->ops, and "sound/soc/" uses soc_pcm_ops using platform->driver->ops. So, I think that they must have their own soc_pcm_ops object. How do you think about this?

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Re: [alsa-devel] [RFC] hda - is_jack_detectable()
by Raymond Yau 27 Dec '11

27 Dec '11

2011/11/14 Takashi Iwai <tiwai(a)suse.de>: > At Sat, 12 Nov 2011 19:31:20 +0800, > Raymond Yau wrote: >> >> 2011/11/9 Takashi Iwai <tiwai(a)suse.de>: >> > At Wed, 09 Nov 2011 08:08:17 +0100, >> > Takashi Iwai wrote: >> >> >> >> At Wed, 9 Nov 2011 12:06:04 +0800, >> >> Raymond Yau wrote: >> >> > >> >> > 2011/11/6 Takashi Iwai <tiwai(a)suse.de>: >> >> > > At Sat, 5 Nov 2011 13:30:21 +0800, >> >> > > Raymond Yau wrote: >> >> > >> >> >> > >> 2011/11/2 Takashi Iwai <tiwai(a)suse.de>: >> >> > >> > At Tue, 01 Nov 2011 17:03:29 -0400, >> >> > >> > David Henningsson wrote: >> >> > >> >> >> >> > >> >> 2011-10-24 23:31, Raymond Yau skrev: >> >> > >> >> > sorry, it should be >> >> > >> >> > >> >> > >> >> > static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) >> >> > >> >> > { >> >> > >> >> > return (snd_hda_query_pin_caps(codec, nid)& AC_PINCAP_PRES_DETECT)&& >> >> > >> >> > + !(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid) >> >> > >> >> > & AC_DEFCFG_MISC_NO_PRESENCE))&& >> >> > >> >> > (get_wcaps(codec, nid)& AC_WCAP_UNSOL_CAP); >> >> > >> >> > } >> >> > >> >> > >> >> > >> >> I also think it is a good idea, in fact I thought it was this way >> >> > >> >> already. Can we merge this patch? >> >> > >> > >> >> > >> > OK, I checked through alsa-info series and confirmed that this causes >> >> > >> > no harm, so I merged it now. >> >> > >> > >> >> > >> > >> >> > >> > thanks, >> >> > >> > >> >> > >> > Takashi >> >> > >> > >> >> > >> > >> >> > >> >> >> > >> Sorry the previous patch is wrong >> >> > > >> >> > > Ugh, so actually changing it would may bring too many changes, and >> >> > > most likely regressions, too. I disabled the code for now for 3.2-rc1. >> >> > > Once when we cover all test cases and fix broken BIOS devices enough, >> >> > > we can re-enable it in near future. >> >> > > >> >> > >> attach the patch and test cases (pins 0x11 and 0x14) front panel >> >> > >> green and pink jacks when set Front Audio Panel to AC97 or HD in BIOS >> >> > >> setup >> >> > > >> >> > > OK, thanks, we can refer to this for more tests. >> >> > > >> >> > > >> >> > > Takashi >> >> > > >> >> > >> >> > If bit 0 is set, it indicates that the jack has no presence detect capability >> >> > >> >> > >> >> > The Configuration Default register is defined as shown in Figure 66. >> >> > 31:30 29:24 23:20 19:16 15:12 11:8 >> >> > 7:4 3:0 >> >> > Port Location Default Connection Color Misc >> >> > Default Sequence >> >> > Connectivity Device Type Association >> >> > Figure 66. Configuration Data Structure >> >> > >> >> > >> >> > This mean that bit 0 of Misc is bit 8 of the Configuration Default register >> >> > >> >> > >> >> > !(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) >> >> > & AC_DEFCFG_MISC_NO_PRESENCE)&& >> >> > >> >> >> >> Yeah, I already tried it and it still gives too many false-positives. >> >> Many ASUS laptops (also mobo) set this bit to all pins no matter >> >> whether it's an obviously detectable jack or not. Some Toshiba >> >> laptops do so, too. >> >> >> >> That being said, checking this bit alone isn't reliable because of too >> >> many BIOS bugs. A bit more clever test is necessary. >> > >> > It seems that a broken BIOS sets this bit on all pins in general. >> > So, the patch below (against the latest sound git tree or 3.2-rc1 >> > kernel) seems working well by filtering out such a brokenness. >> > >> > Let me know if this works. If it's OK, I'll try to merge it for >> > 3.2-rc2. >> > >> >> The patch seem ok >> >> Need a hack to build the hda-jack branch >> >> It is quite simple to enable unsolicited event for all proper jacks by >> >> snd_hda_jack_add_kctrls(codec,(codec->spec->autocfg)); >> >> and >> >> static void ad1988_unsol_event(struct hda_codec *codec, unsigned int res) >> { >> struct ad198x_spec *spec = codec->spec; >> struct hda_jack_tbl *jack; >> u32 sense, caps; >> int pin; >> jack = snd_hda_jack_tbl_get_from_tag(codec, res >> 26); >> pin = jack->nid; >> caps = snd_hda_codec_get_pincfg(codec, pin); >> sense = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_PIN_SENSE, 0); >> if (sense & AC_PINSENSE_PRESENCE) >> snd_printdd("Pin 0x%x pd=1 imp=%d %s at %s %s %s\n", >> pin, >> sense & AC_PINSENSE_IMPEDANCE_MASK, >> snd_hda_get_jack_type(caps), >> snd_hda_get_jack_connectivity(caps), >> snd_hda_get_jack_location(caps), >> get_jack_color(caps)); >> else >> snd_printdd("Pin 0x%x pd=0 %s at %s %s %s\n", >> pin, >> snd_hda_get_jack_type(caps), >> snd_hda_get_jack_connectivity(caps), >> snd_hda_get_jack_location(caps), >> get_jack_color(caps)); >> >> if (pin == spec->autocfg.hp_pins[0] && !spec->independent_hp) >> ad198x_hp_automute(codec); >> if (pin == spec->ext_mic_pin) >> ad198x_mic_automute(codec); >> } >> >> >> However it seem that result is different in kctl >> >> >> amixer -c1 contents >> numid=5,iface=CARD,name='Front Headphone Jack' >> ; type=BOOLEAN,access=r-------,values=1 >> : values=off >> numid=6,iface=CARD,name='Front Mic Jack' >> ; type=BOOLEAN,access=r-------,values=1 >> : values=off >> numid=8,iface=CARD,name='Line Jack' >> ; type=BOOLEAN,access=r-------,values=1 >> : values=off >> numid=1,iface=CARD,name='Line-Out Jack' >> ; type=BOOLEAN,access=r-------,values=1 >> : values=off >> numid=2,iface=CARD,name='Line-Out Jack',index=1 >> ; type=BOOLEAN,access=r-------,values=1 >> : values=off >> numid=3,iface=CARD,name='Line-Out Jack',index=2 >> ; type=BOOLEAN,access=r-------,values=1 >> : values=off >> numid=4,iface=CARD,name='Line-Out Jack',index=3 >> ; type=BOOLEAN,access=r-------,values=1 >> : values=off >> numid=7,iface=CARD,name='Rear Mic Jack' >> ; type=BOOLEAN,access=r-------,values=1 >> : values=off >> >> >> >> >> ./hda-jack-sense-test.py -c1 >> Pin 0x11 Green HP Out (Ext Front): Present = NO >> Pin 0x12 Green Line Out (Ext Rear): Present = YES Impedance = 19 >> Pin 0x14 Pink Mic (Ext Front): Present = NO >> Pin 0x15 Blue Line In (Ext Rear): Present = NO >> Pin 0x16 Black Line Out (Ext Rear): Present = YES Impedance = 3200 >> Pin 0x17 Pink Mic (Ext Rear): Present = YES Impedance = 896 >> Pin 0x18 Black CD (Int ATAPI): No jack detect capability >> Pin 0x1a Other (Int ATAPI): No jack detect capability >> Pin 0x1b Other SPDIF Out (Ext Rear): No jack detect capability >> Pin 0x24 Orange Line Out (Ext Rear): Present = NO >> Pin 0x25 Grey Line Out (Ext Rear): Present = YES Impedance = 3200 >> >> >> It need "SET_PIN_SENSE" and some delay to get the correct impedance >> of the jacks during driver load >> >> >> int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, >> unsigned char action) >> { >> + int ret; >> struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); >> if (!jack) >> return -ENOMEM; >> if (jack->jack_detect) >> return 0; /* already registered */ >> jack->jack_detect = 1; >> if (action) >> jack->action = action; >> + ret = snd_hda_codec_write_cache(codec, nid, 0, >> + AC_VERB_SET_UNSOLICITED_ENABLE, >> + AC_USRSP_EN | jack->tag); >> + if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_IMP_SENSE) { >> snd_hda_codec_read(codec, nid, 0, >> AC_VERB_SET_PIN_SENSE, 0); >> + msleep(200); >> + } >> + return ret; > > The lack of the initial pin-state of kclts was fixed in a different > patch now. But, the need of 200ms delay worries me whether it's > needed generically at each pin-sense reading or it's just only for the > initial reading. > > Does the pin-sense reading work in general after loading the driver > stably? > > > Presence detect bit is alway correct, without the delay. the only issue is the impedance measurement of the plugged jacks at computer boot need a delay 200ms For the user application to know the impedance , it seem the client application have to issue 1) set_pin_sense 2) delay 3) get_pin_sense to get the correct impedance of the attached device just like my modified hda-jack-sense-test in previous email Can it measure the impedance of the jacks of your computer ?

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[alsa-devel] Capture from 3+ cards connected to a USB hub - distorted samples
by Pavel Hofman 26 Dec '11

26 Dec '11

Hi, I have been informed about the following problem which I was able to reproduce afterwards: * Latest git.alsa-project.org drivers * USB 2.0 hub, 3 or more USB soundcards (inexpensive USB sticks, USB audio v.1 types, async capture, adaptive playback) hooked to the hub, identified A, B, C, D. * NONE of these soundcards is playing back Now: * Card A begins microphone capturing (mono 48/16), result OK * Card B begins capturing (mono 48/16), both results still OK * As soon as the third card C begins capturing, the signal captured from cards A and B gets distorted. Audacity shows some samples are incorrect, distorting the expected sine waveform of the testing signal. I can provide screenshots as well as recorded samples, if needed. * The signal recorded from card C is OK. BUT * If any card hooked to the HUB is playing back (even a fourth one D), the distortion does not occur. It happens only when there is no playback stream going through the hub. Traffic on another USB port has no effect, in fact the test signal for the three cards was provided by a multichannel USB card hooked to another USB port. It seems to me as if blocks of samples for each card in the USB frame sometimes overlap by a few bits, corrupting the neighbouring card samples, or are recovered incorrectly. I understand I have provided no real data, I am ready to do so plus any tests you would consider useful. Thanks a lot for suggestions of the next debugging steps I should do. Regards, Pavel.

2 4

[alsa-devel] [PATCH v2] ASoC: jz4740: Convert qi_lb60 to use snd_soc_register_card()
by Axel Lin 26 Dec '11

26 Dec '11

Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: fix a typo in v1. Only compile tested, I appreciate if someone can test it. arch/mips/jz4740/board-qi_lb60.c | 6 ++++ sound/soc/jz4740/qi_lb60.c | 56 +++++++++++++++++-------------------- 2 files changed, 32 insertions(+), 30 deletions(-) diff --git a/arch/mips/jz4740/board-qi_lb60.c b/arch/mips/jz4740/board-qi_lb60.c index 639e3ce..9a91fe9 100644 --- a/arch/mips/jz4740/board-qi_lb60.c +++ b/arch/mips/jz4740/board-qi_lb60.c @@ -418,6 +418,11 @@ static struct platform_device qi_lb60_charger_device = { }, }; +/* audio */ +static struct platform_device qi_lb60_audio_device = { + .name = "qi-lb60-audio", + .id = -1, +}; static struct platform_device *jz_platform_devices[] __initdata = { &jz4740_udc_device, @@ -434,6 +439,7 @@ static struct platform_device *jz_platform_devices[] __initdata = { &qi_lb60_gpio_keys, &qi_lb60_pwm_beeper, &qi_lb60_charger_device, + &qi_lb60_audio_device, }; static void __init board_gpio_setup(void) diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 0097c3b..b29aa55 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -91,56 +91,52 @@ static struct snd_soc_card qi_lb60 = { .num_dapm_routes = ARRAY_SIZE(qi_lb60_routes), }; -static struct platform_device *qi_lb60_snd_device; - static const struct gpio qi_lb60_gpios[] = { { QI_LB60_SND_GPIO, GPIOF_OUT_INIT_LOW, "SND" }, { QI_LB60_AMP_GPIO, GPIOF_OUT_INIT_LOW, "AMP" }, }; -static int __init qi_lb60_init(void) +static int __devinit qi_lb60_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &qi_lb60; int ret; - qi_lb60_snd_device = platform_device_alloc("soc-audio", -1); - - if (!qi_lb60_snd_device) - return -ENOMEM; - ret = gpio_request_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); - if (ret) { - pr_err("qi_lb60 snd: Failed to request gpios: %d\n", ret); - goto err_device_put; - } + if (ret) + return ret; - platform_set_drvdata(qi_lb60_snd_device, &qi_lb60); + card->dev = &pdev->dev; - ret = platform_device_add(qi_lb60_snd_device); + ret = snd_soc_register_card(card); if (ret) { - pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret); - goto err_unset_pdata; + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); } - - return 0; - -err_unset_pdata: - platform_set_drvdata(qi_lb60_snd_device, NULL); -/*err_gpio_free_array:*/ - gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); -err_device_put: - platform_device_put(qi_lb60_snd_device); - return ret; } -module_init(qi_lb60_init); -static void __exit qi_lb60_exit(void) +static int __devexit qi_lb60_remove(struct platform_device *pdev) { - platform_device_unregister(qi_lb60_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); + snd_soc_unregister_card(card); + return 0; } -module_exit(qi_lb60_exit); + +static struct platform_driver qi_lb60_driver = { + .driver = { + .name = "qi-lb60-audio", + .owner = THIS_MODULE, + }, + .probe = qi_lb60_probe, + .remove = __devexit_p(qi_lb60_remove), +}; + +module_platform_driver(qi_lb60_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:qi-lb60-audio"); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: jz4740: Convert qi_lb60 to use snd_soc_register_card()
by Axel Lin 24 Dec '11

24 Dec '11

Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- arch/mips/jz4740/board-qi_lb60.c | 6 ++++ sound/soc/jz4740/qi_lb60.c | 57 ++++++++++++++++++-------------------- 2 files changed, 33 insertions(+), 30 deletions(-) diff --git a/arch/mips/jz4740/board-qi_lb60.c b/arch/mips/jz4740/board-qi_lb60.c index 639e3ce..9a91fe9 100644 --- a/arch/mips/jz4740/board-qi_lb60.c +++ b/arch/mips/jz4740/board-qi_lb60.c @@ -418,6 +418,11 @@ static struct platform_device qi_lb60_charger_device = { }, }; +/* audio */ +static struct platform_device qi_lb60_audio_device = { + .name = "qi-lb60-audio", + .id = -1, +}; static struct platform_device *jz_platform_devices[] __initdata = { &jz4740_udc_device, @@ -434,6 +439,7 @@ static struct platform_device *jz_platform_devices[] __initdata = { &qi_lb60_gpio_keys, &qi_lb60_pwm_beeper, &qi_lb60_charger_device, + &qi_lb60_audio_device, }; static void __init board_gpio_setup(void) diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 0097c3b..f152b81a7 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -91,56 +91,53 @@ static struct snd_soc_card qi_lb60 = { .num_dapm_routes = ARRAY_SIZE(qi_lb60_routes), }; -static struct platform_device *qi_lb60_snd_device; - static const struct gpio qi_lb60_gpios[] = { { QI_LB60_SND_GPIO, GPIOF_OUT_INIT_LOW, "SND" }, { QI_LB60_AMP_GPIO, GPIOF_OUT_INIT_LOW, "AMP" }, }; -static int __init qi_lb60_init(void) +static int __devinit qi_lb60_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &qi_lb60; int ret; - qi_lb60_snd_device = platform_device_alloc("soc-audio", -1); - - if (!qi_lb60_snd_device) - return -ENOMEM; - ret = gpio_request_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); - if (ret) { - pr_err("qi_lb60 snd: Failed to request gpios: %d\n", ret); - goto err_device_put; - } + if (ret) + return ret; - platform_set_drvdata(qi_lb60_snd_device, &qi_lb60); + card->dev = &pdev->dev; - ret = platform_device_add(qi_lb60_snd_device); + ret = snd_soc_register_card(card); if (ret) { - pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret); - goto err_unset_pdata; + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + gpio_free_array(qi_lb60_audio_gpios, + ARRAY_SIZE(qi_lb60_audio_gpios)); } - - return 0; - -err_unset_pdata: - platform_set_drvdata(qi_lb60_snd_device, NULL); -/*err_gpio_free_array:*/ - gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); -err_device_put: - platform_device_put(qi_lb60_snd_device); - return ret; } -module_init(qi_lb60_init); -static void __exit qi_lb60_exit(void) +static int __devexit qi_lb60_remove(struct platform_device *pdev) { - platform_device_unregister(qi_lb60_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); + snd_soc_unregister_card(card); + return 0; } -module_exit(qi_lb60_exit); + +static struct platform_driver qi_lb60_driver = { + .driver = { + .name = "qi-lb60-audio", + .owner = THIS_MODULE, + }, + .probe = qi_lb60_probe, + .remove = __devexit_p(qi_lb60_remove), +}; + +module_platform_driver(qi_lb60_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:qi-lb60-audio"); -- 1.7.5.4

1 0

Re: [alsa-devel] [PATCH] ASoC: wm8776: add missing break in sample size switch
by Mark Brown 23 Dec '11

23 Dec '11

On Fri, Dec 23, 2011 at 05:32:25PM +0100, Joachim Eastwood wrote: > Broken in commit d1dc698a54259cb454284456483b45f67c865cf8 > > Signed-off-by: Joachim Eastwood <joachim.eastwood(a)jotron.com> Applied, but I had to manually fix the word wrapping your mail program introduced: > +++ b/sound/soc/codecs/wm8776.c > @@ -241,6 +241,7 @@ static int wm8776_hw_params(struct > snd_pcm_substream *substream, which meant patch couldn't understand it.

1 0

Re: [alsa-devel] [PATCH 5/5] ASoC: mxs: correct 'direction' of device_prep_dma_cyclic
by Vinod Koul 23 Dec '11

23 Dec '11

On Wed, 2011-12-14 at 15:47 +0800, Shawn Guo wrote: > The commit 49920bc (dmaengine: add new enum dma_transfer_direction) > changes the type of parameter 'direction' of device_prep_dma_cyclic > from dma_data_direction to dma_transfer_direction. > > Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> > --- > sound/soc/mxs/mxs-pcm.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c > index 0e12f4e..105f42a 100644 > --- a/sound/soc/mxs/mxs-pcm.c > +++ b/sound/soc/mxs/mxs-pcm.c > @@ -136,7 +136,7 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, > iprtd->period_bytes * iprtd->periods, > iprtd->period_bytes, > substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? > - DMA_TO_DEVICE : DMA_FROM_DEVICE); > + DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); > if (!iprtd->desc) { > dev_err(&chan->dev->device, "cannot prepare slave dma\n"); > return -EINVAL; Has this been applied in soc tree? I don't see it in linux-next or do you want this to go thru slave-dma? -- ~Vinod

2 2

[alsa-devel] ASoC updates for 3.3
by Mark Brown 23 Dec '11

23 Dec '11

The following changes since commit c45471eac2bdc271df40963ac8448d76ac434872: ASoC: Fix compile error in sound/soc/mid-x86/sst_platform.c (2011-12-16 07:50:15 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 Axel Lin (40): ASoC: Remove unused extern declarations for sh4_hac_dai and sh7760_soc_platform ASoC: Staticise sst_pcm_new and sst_soc_platform_drv ASoC: Staticise mfld_msic_dailink ASoC: pxa: Convert e740_wm9705 to use snd_soc_register_card() ASoC: pxa: Convert e750_wm9705 to use snd_soc_register_card() ASoC: pxa: Convert imote2 to use snd_soc_register_card() ASoC: pxa: Convert tosa to use snd_soc_register_card() ASoC: pxa: Convert e800_wm9712 to use snd_soc_register_card() ASoC: Convert max9850 to table based DAPM and control init ASoC: Convert uda1380 to table based DAPM and control init ASoC: Remove export of s3c_pcm_dai ASoC: Use dai_fmt in edb93xx machine driver ASoC: Use dai_fmt in snappercl15 machine driver ASoC: Rename rt562[1|2]_vol_snd_controls to alc562[1|2]_vol_snd_controls ASoC: Convert ak4104 to devm_kzalloc() ASoC: Convert ak4535 to devm_kzalloc() ASoC: Convert ak4641 to devm_kzalloc() ASoC: Convert ak4642 to devm_kzalloc() ASoC: Convert ak4671 to devm_kzalloc() ASoC: Use dai_fmt in hx4700 machine driver ASoC: Use dai_fmt in imote2 machine driver ASoC: Convert 88pm860x-codec to table based DAPM and control init ASoC: Convert cs42l51 to table based DAPM and control init ASoC: pxa: Add .owner to struct snd_soc_card ASoC: samsung: Add .owner to struct snd_soc_card ASoC: fsl: Add .owner to struct snd_soc_card ASoC: atmel: Add .owner to struct snd_soc_card ASoC: blackfin: Add .owner to struct snd_soc_card ASoC: davinci: Add .owner to struct snd_soc_card ASoC: ep93xx: Add .owner to struct snd_soc_card ASoC: tegra: Add .owner to struct snd_soc_card ASoC: au1x: Add .owner to struct snd_soc_card ASoC: imx: Add .owner to struct snd_soc_card ASoC: jz4740: Add .owner to struct snd_soc_card ASoC: nuc900: Add .owner to struct snd_soc_card ASoC: s6000: Add .owner to struct snd_soc_card ASoC: sh: Add .owner to struct snd_soc_card ASoC: txx9: Add .owner to struct snd_soc_card ASoC: kirkwood: Add .owner to struct snd_soc_card ASoC: mid-x86: Add .owner to struct snd_soc_card Leon Romanovsky (1): ASoC: Tegra machine ASoC driver for boards using ALC5332 codec Mark Brown (23): ASoC: Fix wm8995 regmap usage ASoC: Staticise asoc_idma_platform ASoC: Complete initialisation before registering Samsung PCM DAI ASoC: Remove cache default for volatile wm9081 reset register ASoC: Use standard snd_soc_cache_sync() for WM9090 ASoC: Convert wm8993 to devm_kzalloc() ASoC: Use standard register cache sync in wm8993 mfd: Test for jack detection when deciding if wm8994 should suspend ASoC: Tune the accessory detection rates for WM8996 ASoC: Use core pm_runtime callbacks for siu_dai ASoC: Use core pm_runtime callbacks for fsi ASoC: Fix partial cherry pick in wm8993 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound into for-3.3 ASoC: Convert WM8960 to devm_kzalloc() ASoC: Remove I2C ifdefs from wm8961 driver ASoC: Convert wm8961 to devm_kzalloc() ASoC: Use standard cache sync code in wm8961 Merge branch 'for-3.2' into for-3.3 ASoC: Remove ifdefs for GPIO_SYSFS Revert "ASoC: mxs: correct 'direction' of device_prep_dma_cyclic" Merge branch 'for-3.2' into for-3.3 ASoC: Declare soc_new_pcm() properly ASoC: Add trivial pm_runtime usage to Samsung DAI drivers Peter Ujfalusi (2): OMAP4: mcbsp: Enable FIFO usage ASoC: omap-mcbsp: Enable FIFO usage on OMAP4 Shawn Guo (1): ASoC: mxs: correct 'direction' of device_prep_dma_cyclic Stephen Warren (4): ASoC: Add utility to set a card's name from device tree ASoC: Add utility to parse DAPM routes from device tree ASoC: Tegra+WM8903 machine: Add device tree binding ASoC: Allow DAI links to be specified using device tree nodes .../bindings/sound/tegra-audio-wm8903.txt | 71 +++++++ arch/arm/mach-omap2/mcbsp.c | 3 + arch/arm/mach-pxa/eseries.c | 18 ++ arch/arm/mach-pxa/stargate2.c | 6 + arch/arm/mach-pxa/tosa.c | 6 + drivers/mfd/wm8994-core.c | 14 ++ include/sound/soc.h | 10 + sound/soc/atmel/sam9g20_wm8731.c | 1 + sound/soc/atmel/snd-soc-afeb9260.c | 1 + sound/soc/au1x/db1000.c | 1 + sound/soc/au1x/db1200.c | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 1 + sound/soc/blackfin/bf5xx-ad193x.c | 1 + sound/soc/blackfin/bf5xx-ad1980.c | 1 + sound/soc/blackfin/bf5xx-ad73311.c | 1 + sound/soc/blackfin/bf5xx-ssm2602.c | 1 + sound/soc/blackfin/bfin-eval-adau1373.c | 1 + sound/soc/blackfin/bfin-eval-adau1701.c | 1 + sound/soc/blackfin/bfin-eval-adav80x.c | 1 + sound/soc/codecs/88pm860x-codec.c | 15 +- sound/soc/codecs/ak4104.c | 6 +- sound/soc/codecs/ak4535.c | 6 +- sound/soc/codecs/ak4641.c | 7 +- sound/soc/codecs/ak4642.c | 6 +- sound/soc/codecs/ak4671.c | 6 +- sound/soc/codecs/alc5623.c | 12 +- sound/soc/codecs/cs42l51.c | 17 +- sound/soc/codecs/max9850.c | 17 +- sound/soc/codecs/uda1380.c | 24 +-- sound/soc/codecs/wm8960.c | 7 +- sound/soc/codecs/wm8961.c | 27 +-- sound/soc/codecs/wm8993.c | 29 +--- sound/soc/codecs/wm8995.c | 1 + sound/soc/codecs/wm8996.c | 18 +- sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9090.c | 13 +- sound/soc/davinci/davinci-evm.c | 6 + sound/soc/davinci/davinci-sffsdr.c | 1 + sound/soc/ep93xx/edb93xx.c | 15 +- sound/soc/ep93xx/simone.c | 1 + sound/soc/ep93xx/snappercl15.c | 13 +- sound/soc/fsl/efika-audio-fabric.c | 14 +- sound/soc/fsl/pcm030-audio-fabric.c | 14 +- sound/soc/imx/eukrea-tlv320.c | 1 + sound/soc/imx/mx27vis-aic32x4.c | 1 + sound/soc/imx/phycore-ac97.c | 1 + sound/soc/imx/wm1133-ev1.c | 1 + sound/soc/jz4740/qi_lb60.c | 1 + sound/soc/kirkwood/kirkwood-openrd.c | 1 + sound/soc/kirkwood/kirkwood-t5325.c | 1 + sound/soc/mid-x86/mfld_machine.c | 3 +- sound/soc/mid-x86/sst_platform.c | 4 +- sound/soc/nuc900/nuc900-audio.c | 1 + sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/e740_wm9705.c | 76 +++---- sound/soc/pxa/e750_wm9705.c | 67 +++---- sound/soc/pxa/e800_wm9712.c | 67 +++---- sound/soc/pxa/em-x270.c | 1 + sound/soc/pxa/hx4700.c | 17 +-- sound/soc/pxa/imote2.c | 58 +++--- sound/soc/pxa/magician.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/raumfeld.c | 2 + sound/soc/pxa/saarb.c | 1 + sound/soc/pxa/spitz.c | 1 + sound/soc/pxa/tavorevb3.c | 1 + sound/soc/pxa/tosa.c | 78 +++----- sound/soc/pxa/z2.c | 1 + sound/soc/pxa/zylonite.c | 1 + sound/soc/s6000/s6105-ipcam.c | 1 + sound/soc/samsung/goni_wm8994.c | 1 + sound/soc/samsung/h1940_uda1380.c | 1 + sound/soc/samsung/i2s.c | 6 +- sound/soc/samsung/idma.c | 2 +- sound/soc/samsung/jive_wm8750.c | 1 + sound/soc/samsung/littlemill.c | 1 + sound/soc/samsung/ln2440sbc_alc650.c | 1 + sound/soc/samsung/lowland.c | 1 + sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/pcm.c | 20 +- sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/samsung/s3c24xx_simtec_hermes.c | 1 + sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 1 + sound/soc/samsung/s3c24xx_uda134x.c | 1 + sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk2443_wm9710.c | 1 + sound/soc/samsung/smdk_spdif.c | 1 + sound/soc/samsung/smdk_wm8580.c | 1 + sound/soc/samsung/smdk_wm8580pcm.c | 1 + sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/smdk_wm8994pcm.c | 1 + sound/soc/samsung/smdk_wm9713.c | 1 + sound/soc/samsung/speyside.c | 1 + sound/soc/samsung/tobermory.c | 1 + sound/soc/sh/fsi-ak4642.c | 1 + sound/soc/sh/fsi-da7210.c | 1 + sound/soc/sh/fsi-hdmi.c | 1 + sound/soc/sh/fsi.c | 4 - sound/soc/sh/migor.c | 1 + sound/soc/sh/sh7760-ac97.c | 5 +- sound/soc/sh/siu_dai.c | 6 - sound/soc/soc-core.c | 148 +++++++++++++- sound/soc/soc-jack.c | 4 - sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_alc5632.c | 214 ++++++++++++++++++++ sound/soc/tegra/tegra_wm8903.c | 129 ++++++++++--- sound/soc/tegra/trimslice.c | 1 + sound/soc/txx9/txx9aclc-generic.c | 1 + 111 files changed, 940 insertions(+), 446 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt create mode 100644 sound/soc/tegra/tegra_alc5632.c

2 1

[alsa-devel] [RFC 0/5] compress: add compress data API to ALSA kernel
by Vinod Koul 23 Dec '11

23 Dec '11

As agreed during the ASoC workshop in May 2011, this set of patches are a proposal from Intel to add a new compressed data interface to ALSA. This proposal builds on the existing Intel solution and the experience gathered over two years with Meego and Android solutions. This proposal is no longer Intel-specific and was generalized with the addition of a basic core; codec-related definitions are based on the vendor-neutral OpenMAX AL standard. The code as is isn't functional and will need low-level hardware-specific drivers and a tighter integration with ALSA and ASoC. The main ideas will be presented during the Linux Plumbers Conference on September 9. Feedback through the mailing list would be greatly appreciated as well. Future versions will include support for capture, mmap, integration with alsa, asoc, dsp etc Support for effects is beyond the scope of this interface. Vinod Koul (5): compress API documentation compress: add compress parameter definations compress: add API header and driver header files compress: add the core file compress: add makefile and kconfig file .../sound/alsa/compress/snd_compress_data.txt | 186 ++++++ include/sound/compress_driver.h | 138 +++++ include/sound/compress_offload.h | 137 +++++ include/sound/snd_compress_params.h | 396 ++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/compress_offload/Kconfig | 12 + sound/compress_offload/Makefile | 3 + sound/compress_offload/core.c | 632 ++++++++++++++++++++ 9 files changed, 1508 insertions(+), 1 deletions(-) create mode 100644 Documentation/sound/alsa/compress/snd_compress_data.txt create mode 100644 include/sound/compress_driver.h create mode 100644 include/sound/compress_offload.h create mode 100644 include/sound/snd_compress_params.h create mode 100644 sound/compress_offload/Kconfig create mode 100644 sound/compress_offload/Makefile create mode 100644 sound/compress_offload/core.c

9 110

[alsa-devel] [PATCH] ASoC: Allocate PCM operations dynamically to support multiple DAIs
by Sangbeom Kim 23 Dec '11

23 Dec '11

From: Sangsu Park <sangsu4u.park(a)samsung.com> The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch allocates ASoC core PCM operations dynamically for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/core/pcm.c | 1 + sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++++-------------------- 2 files changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8928ca8..15cf27a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -769,6 +769,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream_next = substream->next; snd_pcm_timer_done(substream); snd_pcm_substream_proc_done(substream); + kfree(substream->ops); kfree(substream); substream = substream_next; } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec..3a52534 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -598,17 +598,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -616,10 +605,25 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + soc_pcm_ops = kzalloc(sizeof(*soc_pcm_ops), GFP_KERNEL); + if (soc_pcm_ops == NULL) { + snd_printk(KERN_ERR "Cannot allocate PCM OPS\n"); + return -ENOMEM; + } + + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_pcm_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->pointer = soc_pcm_pointer; + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -643,20 +647,20 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->pointer = platform->driver->ops->pointer; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); -- 1.7.1

2 1

[alsa-devel] [PATCH 1/8] ASoC: imx: Add .owner to struct snd_soc_card
by Axel Lin 23 Dec '11

23 Dec '11

Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/eukrea-tlv320.c | 1 + sound/soc/imx/mx27vis-aic32x4.c | 1 + sound/soc/imx/phycore-ac97.c | 1 + sound/soc/imx/wm1133-ev1.c | 1 + 4 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 75fb4b8..1c1fdd1 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -87,6 +87,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { static struct snd_soc_card eukrea_tlv320 = { .name = "cpuimx-audio", + .owner = THIS_MODULE, .dai_link = &eukrea_tlv320_dai, .num_links = 1, }; diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 054110b..3c2eed9 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -86,6 +86,7 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = { static struct snd_soc_card mx27vis_aic32x4 = { .name = "visstrim_m10-audio", + .owner = THIS_MODULE, .dai_link = &mx27vis_aic32x4_dai, .num_links = 1, }; diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c index a7deb5c..6ac1211 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/imx/phycore-ac97.c @@ -38,6 +38,7 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { static struct snd_soc_card imx_phycore = { .name = "PhyCORE-ac97-audio", + .owner = THIS_MODULE, .dai_link = imx_phycore_dai_ac97, .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), }; diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c index 75b4c72..96fa761 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/imx/wm1133-ev1.c @@ -254,6 +254,7 @@ static struct snd_soc_dai_link wm1133_ev1_dai = { static struct snd_soc_card wm1133_ev1 = { .name = "WM1133-EV1", + .owner = THIS_MODULE, .dai_link = &wm1133_ev1_dai, .num_links = 1, }; -- 1.7.5.4

4 10

[alsa-devel] [PATCH v2] ASoC: au1x: Add .owner to struct snd_soc_card
by Axel Lin 23 Dec '11

23 Dec '11

Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: regenerate the patch against asoc tree. sound/soc/au1x/db1000.c | 1 + sound/soc/au1x/db1200.c | 2 ++ 2 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c index 127477a..8387d42 100644 --- a/sound/soc/au1x/db1000.c +++ b/sound/soc/au1x/db1000.c @@ -29,6 +29,7 @@ static struct snd_soc_dai_link db1000_ac97_dai = { static struct snd_soc_card db1000_ac97 = { .name = "DB1000_AC97", + .owner = THIS_MODULE, .dai_link = &db1000_ac97_dai, .num_links = 1, }; diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 289312c..15e5138 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -45,6 +45,7 @@ static struct snd_soc_dai_link db1200_ac97_dai = { static struct snd_soc_card db1200_ac97_machine = { .name = "DB1200_AC97", + .owner = THIS_MODULE, .dai_link = &db1200_ac97_dai, .num_links = 1, }; @@ -94,6 +95,7 @@ static struct snd_soc_dai_link db1200_i2s_dai = { static struct snd_soc_card db1200_i2s_machine = { .name = "DB1200_I2S", + .owner = THIS_MODULE, .dai_link = &db1200_i2s_dai, .num_links = 1, }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH v3 0/9] OMAP4: ASoC: Support for PandaBoard family
by Peter Ujfalusi 23 Dec '11

23 Dec '11

Hello, the following series will add ASoC support for PandaBoards. PandaBoards have different audio routings compared to SDP4430/Blaze boards, but the differences not that big to justify a new ASoC machine driver. This series is based on the first three patch from the v1 sereis (those, which has been applied by Liam). Changes since v2: - Remove direct board dependency for the ASoC machine driver (replaced by ARCH_OMAP4) - mach-omap2/board-omap4panda: Generic board revision handling function instead of dedicated for audio. - Macro twl6040_disconnect_pin replaced with a function Changes since v1: As suggested by Mark the platform data, machine driver has been converted to process feature flags instead of Board IDs. - The ASoC machine driver registers all DAPM widgets. Based on the received configuration it will disable the unused paths - PCM for dmic is only created, if the board has digital mic conencted - Jack functionality will be only used on boards supporting it - Sound card name for SDP4430 has net been changed Regards, Peter --- Peter Ujfalusi (9): include: platform_data: Platform data header for OMAP4 ASoC audio OMAP4: 4430sdp: Register platform device for OMAP4 audio ASoC: omap-abe-twl6040: Convert to platform deriver ASoC: twl6040: Convert MICBIAS to SUPPLY widget ASoC: omap-abe-twl6040: Add complete DAPM routing ASoC: omap-abe-twl6040: DAI link selection based on platform data ASoC: omap-abe-twl6040: Configure card according to platform data OMAP4: omap4panda: Enable audio support ASoC: Kconfig: OMAP4: Enable support for PandaBoards arch/arm/mach-omap2/board-4430sdp.c | 27 ++++ arch/arm/mach-omap2/board-omap4panda.c | 57 ++++++++- include/linux/platform_data/omap-abe-twl6040.h | 48 +++++++ sound/soc/codecs/twl6040.c | 16 +- sound/soc/omap/Kconfig | 4 +- sound/soc/omap/omap-abe-twl6040.c | 181 ++++++++++++++++-------- 6 files changed, 263 insertions(+), 70 deletions(-) create mode 100644 include/linux/platform_data/omap-abe-twl6040.h -- 1.7.8.1

3 12

[alsa-devel] [PATCH 0/5] OMAP4: Fix CLKR/FSR mux for port with 6 pin mode
by Peter Ujfalusi 23 Dec '11

23 Dec '11

Hello, This series will fix the CLKR/FSR signal mux configuration for OMAP4 devices. McBSP4 has 6 pin configuration on McBSP4 port while OMAP2/3 has the same capability on McBSP1 port. The API for mux control will be simplified for the mcbsp users (ASoC). The series depends on earlier series sent for mcbsp: OMAP4: McBSP: Fix clock reparenting I think the ASoC patch can go via linux-omap, since we do not have other changes for McBSP coming via sound tree. Regards, Peter --- Peter Ujfalusi (5): OMAP2/3: mcbsp: Check McBSP port number in CLKR/FSR mux callback OMAP: mcbsp: Signle function for CLKR/FSR source mux configuration ASoC: omap-mcbsp: Use the new interface for configuring CLKR/FSR mux OMAP: mcbsp: Remove old CLKR/FSR mux functions OMAP4: mcbsp: Callback function for McBSP4 CLKR/FSR mux selection arch/arm/mach-omap2/mcbsp.c | 47 +++++++++++++++++++++++++++++ arch/arm/plat-omap/include/plat/mcbsp.h | 15 ++++------ arch/arm/plat-omap/mcbsp.c | 49 +++++++++++++++---------------- sound/soc/omap/omap-mcbsp.c | 12 +++++-- 4 files changed, 85 insertions(+), 38 deletions(-) -- 1.7.8.1

3 7

[alsa-devel] [PATCH] ASoC: Allocate PCM operations dynamically to support multiple DAIs
by Sangbeom Kim 23 Dec '11

23 Dec '11

The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch allocates ASoC core PCM operations dynamically for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/core/pcm.c | 1 + sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++++-------------------- 2 files changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8928ca8..15cf27a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -769,6 +769,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream_next = substream->next; snd_pcm_timer_done(substream); snd_pcm_substream_proc_done(substream); + kfree(substream->ops); kfree(substream); substream = substream_next; } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec..3a52534 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -598,17 +598,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -616,10 +605,25 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + soc_pcm_ops = kzalloc(sizeof(*soc_pcm_ops), GFP_KERNEL); + if (soc_pcm_ops == NULL) { + snd_printk(KERN_ERR "Cannot allocate PCM OPS\n"); + return -ENOMEM; + } + + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_pcm_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->pointer = soc_pcm_pointer; + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -643,20 +647,20 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->pointer = platform->driver->ops->pointer; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); -- 1.7.1

3 3

[alsa-devel] [PATCH 0/5] snd-usb endpoint rework, take 5
by Daniel Mack 22 Dec '11

22 Dec '11

I actually wanted to wait for more input, but as the flaws reported so far were rather minor, I addressed to following details in this new round as mentioned by Takashi and Clemens: - removed the faulty "(!u || u->urb)" check - guard the result of min() with the spinlock inside next_packet_size() - removed the tasklet and made the handler a function that is called in-place Thanks again for the reviews, and keep them coming :) Daniel Daniel Mack (5): ALSA: snd-usb: add snd_usb_audio-wide mutex ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: add support for implicit feedback sound/usb/card.c | 10 +- sound/usb/card.h | 62 +++ sound/usb/endpoint.c | 1454 +++++++++++++++++++++++++------------------------ sound/usb/endpoint.h | 32 +- sound/usb/pcm.c | 437 +++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 2 + 7 files changed, 1224 insertions(+), 804 deletions(-) -- 1.7.5.4

2 11

Re: [alsa-devel] [GIT PULL] ARM: amba: Enable module alias autogeneration for AMBA drivers
by Russell King - ARM Linux 22 Dec '11

22 Dec '11

On Thu, Dec 22, 2011 at 11:21:25AM +0000, Dave Martin wrote: > On Mon, Dec 19, 2011 at 09:39:42PM +0000, Russell King - ARM Linux wrote: > > I have pulled it, I just haven't merged it in anywhere yet. > > OK, great -- thanks. > > I wasn't too familiar with the procedure, so I just wanted > to be sure that I hadn't done things wrong. FYI, I'll be pushing out an update to my tree Friday evening.

1 0

[alsa-devel] [PATCH v2 0/9] OMAP4: ASoC: Support for PandaBoard family
by Peter Ujfalusi 22 Dec '11

22 Dec '11

Hello, the following series will add ASoC support for PandaBoards. PandaBoards have different audio routings compared to SDP4430/Blaze boards, but the differences not that big to justify a new ASoC machine driver. This series is based on the first three patch from the v1 sereis (those, which has been applied by Liam). Changes since v1: As suggested by Mark the platform data, machine driver has been converted to process feature flags instead of Board IDs. - The ASoC machine driver registers all DAPM widgets. Based on the received configuration it will disable the unused paths - PCM for dmic is only created, if the board has digital mic conencted - Jack functionality will be only used on boards supporting it - Sound card name for SDP4430 has net been changed Regards, Peter --- Peter Ujfalusi (9): include: platform_data: Platform data header for OMAP4 ASoC audio OMAP4: 4430sdp: Register platform device for OMAP4 audio ASoC: omap-abe-twl6040: Convert to platform deriver ASoC: twl6040: Convert MICBIAS to SUPPLY widget ASoC: omap-abe-twl6040: Add complete DAPM routing ASoC: omap-abe-twl6040: DAI link selection based on platform data ASoC: omap-abe-twl6040: Configure card according to platform data OMAP4: omap4panda: Enable audio support ASoC: Kconfig: OMAP4: Enable support for PandaBoards arch/arm/mach-omap2/board-4430sdp.c | 27 ++++ arch/arm/mach-omap2/board-omap4panda.c | 56 ++++++++- include/linux/platform_data/omap-abe-twl6040.h | 48 +++++++ sound/soc/codecs/twl6040.c | 16 +- sound/soc/omap/Kconfig | 5 +- sound/soc/omap/omap-abe-twl6040.c | 178 ++++++++++++++++-------- 6 files changed, 260 insertions(+), 70 deletions(-) create mode 100644 include/linux/platform_data/omap-abe-twl6040.h -- 1.7.8.1

3 17

[alsa-devel] [PATCH v2 0/9] OMAP4: ASoC: Support for PandaBoard family
by Peter Ujfalusi 22 Dec '11

22 Dec '11

Hello, the following series will add ASoC support for PandaBoards. PandaBoards have different audio routings compared to SDP4430/Blaze boards, but the differences not that big to justify a new ASoC machine driver. This series is based on the first three patch from the v1 sereis (those, which has been applied by Liam). Changes since v1: As suggested by Mark the platform data, machine driver has been converted to process feature flags instead of Board IDs. - The ASoC machine driver registers all DAPM widgets. Based on the received configuration it will disable the unused paths - PCM for dmic is only created, if the board has digital mic conencted - Jack functionality will be only used on boards supporting it - Sound card name for SDP4430 has net been changed Regards, Peter --- Peter Ujfalusi (9): include: platform_data: Platform data header for OMAP4 ASoC audio OMAP4: 4430sdp: Register platform device for OMAP4 audio ASoC: omap-abe-twl6040: Convert to platform deriver ASoC: twl6040: Convert MICBIAS to SUPPLY widget ASoC: omap-abe-twl6040: Add complete DAPM routing ASoC: omap-abe-twl6040: DAI link selection based on platform data ASoC: omap-abe-twl6040: Configure card according to platform data OMAP4: omap4panda: Enable audio support ASoC: Kconfig: OMAP4: Enable support for PandaBoards arch/arm/mach-omap2/board-4430sdp.c | 27 ++++ arch/arm/mach-omap2/board-omap4panda.c | 56 ++++++++- include/linux/platform_data/omap-abe-twl6040.h | 48 +++++++ sound/soc/codecs/twl6040.c | 16 +- sound/soc/omap/Kconfig | 5 +- sound/soc/omap/omap-abe-twl6040.c | 178 ++++++++++++++++-------- 6 files changed, 260 insertions(+), 70 deletions(-) create mode 100644 include/linux/platform_data/omap-abe-twl6040.h -- 1.7.8.1

4 20

[alsa-devel] Audio BoF @ ELC 2012 US
by Patrick Lai 22 Dec '11

22 Dec '11

Hi, Is there going to be Audio BOF session at ELC 2012 US. Liam/Mark: Are you guys going to attend ELC? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

3 2

[alsa-devel] [PATCH] ASoC: Declare soc_new_pcm() properly
by Mark Brown 22 Dec '11

22 Dec '11

Ensure that everything is seeing the same declaration by moving it to a header file rather than putting the declaration in soc-core.c Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 2 -- 2 files changed, 1 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e5414ed..312dfd3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -323,6 +323,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, unsigned int reg); int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d3fcdc1..d19d0d1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -59,8 +59,6 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); - /* * This is a timeout to do a DAPM powerdown after a stream is closed(). * It can be used to eliminate pops between different playback streams, e.g. -- 1.7.7.3

2 1

[alsa-devel] [PATCH] ALSA: hda/jack - Also add jack kctls for Conexant codecs
by David Henningsson 22 Dec '11

22 Dec '11

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Hi Takashi, A patch for your jack branch - it seems you forgot to add kctls for conexant? Also it would be nice to see this merged, soon it might be too late for 3.3? sound/pci/hda/patch_conexant.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ae9c028..99942e6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4325,6 +4325,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) static int cx_auto_build_controls(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; int err; err = cx_auto_build_output_controls(codec); @@ -4333,6 +4334,9 @@ static int cx_auto_build_controls(struct hda_codec *codec) err = cx_auto_build_input_controls(codec); if (err < 0) return err; + err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + if (err < 0) + return err; return conexant_build_controls(codec); } -- 1.7.5.4

3 10

[alsa-devel] [PATCH] ASoC: fsl: Add .owner to struct snd_soc_card
by Axel Lin 22 Dec '11

22 Dec '11

Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/fsl/efika-audio-fabric.c | 14 +++++++------- sound/soc/fsl/pcm030-audio-fabric.c | 14 +++++++------- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index 108b5d8..b2acd329 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -31,8 +31,6 @@ #define DRV_NAME "efika-audio-fabric" -static struct snd_soc_card card; - static struct snd_soc_dai_link efika_fabric_dai[] = { { .name = "AC97", @@ -52,6 +50,13 @@ static struct snd_soc_dai_link efika_fabric_dai[] = { }, }; +static struct snd_soc_card card = { + .name = "Efika", + .owner = THIS_MODULE, + .dai_link = efika_fabric_dai, + .num_links = ARRAY_SIZE(efika_fabric_dai), +}; + static __init int efika_fabric_init(void) { struct platform_device *pdev; @@ -60,11 +65,6 @@ static __init int efika_fabric_init(void) if (!of_machine_is_compatible("bplan,efika")) return -ENODEV; - card.name = "Efika"; - card.dai_link = efika_fabric_dai; - card.num_links = ARRAY_SIZE(efika_fabric_dai); - - pdev = platform_device_alloc("soc-audio", 1); if (!pdev) { pr_err("efika_fabric_init: platform_device_alloc() failed\n"); diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index ba4d85e..b3af55d 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -31,8 +31,6 @@ #define DRV_NAME "pcm030-audio-fabric" -static struct snd_soc_card card; - static struct snd_soc_dai_link pcm030_fabric_dai[] = { { .name = "AC97", @@ -52,6 +50,13 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { }, }; +static struct snd_soc_card card = { + .name = "pcm030", + .owner = THIS_MODULE, + .dai_link = pcm030_fabric_dai, + .num_links = ARRAY_SIZE(pcm030_fabric_dai), +}; + static __init int pcm030_fabric_init(void) { struct platform_device *pdev; @@ -60,11 +65,6 @@ static __init int pcm030_fabric_init(void) if (!of_machine_is_compatible("phytec,pcm030")) return -ENODEV; - - card.name = "pcm030"; - card.dai_link = pcm030_fabric_dai; - card.num_links = ARRAY_SIZE(pcm030_fabric_dai); - pdev = platform_device_alloc("soc-audio", 1); if (!pdev) { pr_err("pcm030_fabric_init: platform_device_alloc() failed\n"); -- 1.7.5.4

2 1

Re: [alsa-devel] ASoC updates for 3.2
by Mark Brown 22 Dec '11

22 Dec '11

On Thu, Dec 22, 2011 at 08:38:39PM +0800, Shawn Guo wrote: > On 22 December 2011 20:02, Mark Brown > > If the API change is only in -next this shouldn't have been applied to > > the ASoC tree at all. This needs go along with the change that adds the > > new API otherwise you break bisection. Things like this *really* need > > to be made clear when posting patches. > I did mention that the patch series addresses the issue seen on -next > tree in the cover letter. And I replied your query as below. You shouldn't rely on cover letters getting noticed, they're too easy to disassociate from the patch. > > With patches like this you really need to supply more context - knowing > > which release the commit you're talking about appears in is useful for > > exammple, > This patch only applies to -next, as the offending patch only sits on > -next. Yeah, that was after I'd applied the patch. TBH given that this is an incompatible API change the original patch relly ought to be updating all users, there shouldn't be separate patches for the drivers.

1 0

Re: [alsa-devel] ASoC updates for 3.2
by Mark Brown 22 Dec '11

22 Dec '11

On Thu, Dec 22, 2011 at 07:21:44PM +0800, Shawn Guo wrote: > On 22 December 2011 19:04, Mark Brown > > Shawn Guo (1): > > ASoC: mxs: correct 'direction' of device_prep_dma_cyclic > If I remember correctly, the API change only sits on -next. You may > not want to push this patch for 3.2, I guess? Oh, ffs. git describe shows the change as being based off 3.1 so it would be expected to appear in the 3.2 merge window but looking at Linus' tree it doesn't seem to be there. I'll revert the change. If the API change is only in -next this shouldn't have been applied to the ASoC tree at all. This needs go along with the change that adds the new API otherwise you break bisection. Things like this *really* need to be made clear when posting patches. I'll revert the change, you should get anyone else who's applied these changes to do so as well.

2 1

[alsa-devel] ASoC updates for 3.2
by Mark Brown 22 Dec '11

22 Dec '11

The following changes since commit 497d496598f411cb7f0690725a0e8b9029cebdb9: ASoC: Fix hx4700 error handling to free gpios if snd_soc_register_card fails (2011-12-12 12:58:42 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.2 An update for an API change that didn't get propagated properly. Shawn Guo (1): ASoC: mxs: correct 'direction' of device_prep_dma_cyclic sound/soc/mxs/mxs-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

2 1

[alsa-devel] [PATCH v2 1/7 v2] ARM: OMAP1: ams-delta: register latch dependent devices later
by Janusz Krzysztofik 22 Dec '11

22 Dec '11

In preparation to converting Amstrad Delta on-board latches to basic_mmio_gpio devices, registration of platform devices which depend on latches and will require initialization of their GPIO pins first, should be moved out of .machine_init down to late_initcall level, as the gpio-generic driver is not available until device_initcall time. The latch reset operation, which will be replaced with GPIO initialization, must also be moved to late_initcall for the same reason. Since there was already another, separate arch_initcall function for setting up one of those latch dependent devices, the on-board modem device, reuse that function, i.e., rename it to a name that matches the new purpose, extend with other device setup relocated from .machine_init, and move down to the late_initcall level. While being at it, add missing gpio_free() in case the modem platform device registration fails. Thanks to Tony Lindgren <tony(a)atomide.com> who suggested this approach instead of shifting up the gpio-generic driver initialization. In addition, defer registration of the Amstrad Delta ASoC and serio devices, done from their device driver files, until late_initcall time, as those drivers will depend on their GPIO pins already requested from the board late_init() function until updated to register their GPIO pins themselves. Created and tested against linux-3.2-rc6. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Changes since the initial version of this patch: * use late_initcall() unconditionally in modules, requested by Russell King (thanks!); an updated patch 7/7 will follow, * in the changelog message, move credits to Tony up a bit, since the person responsible for other changes (he might not necessarily like) is me, not him ;). Comments copied from this patch initial submission: This patch was not present in the initial submission, it replaces the old patch 1/10, providing an alternative solution not touching the gpio-generic driver. arch/arm/mach-omap1/board-ams-delta.c | 28 ++++++++++++++++++++-------- drivers/input/serio/ams_delta_serio.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- 3 files changed, 22 insertions(+), 10 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index b0f15d2..50987c9 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -275,11 +275,14 @@ static struct omap1_cam_platform_data ams_delta_camera_platform_data = { }; static struct platform_device *ams_delta_devices[] __initdata = { - &ams_delta_nand_device, &ams_delta_kp_device, + &ams_delta_camera_device, +}; + +static struct platform_device *late_devices[] __initdata = { + &ams_delta_nand_device, &ams_delta_lcd_device, &ams_delta_led_device, - &ams_delta_camera_device, }; static void __init ams_delta_init(void) @@ -307,9 +310,6 @@ static void __init ams_delta_init(void) omap_serial_init(); omap_register_i2c_bus(1, 100, NULL, 0); - /* Clear latch2 (NAND, LCD, modem enable) */ - ams_delta_latch2_write(~0, 0); - omap1_usb_init(&ams_delta_usb_config); omap1_set_camera_info(&ams_delta_camera_platform_data); #ifdef CONFIG_LEDS_TRIGGERS @@ -345,13 +345,18 @@ static struct platform_device ams_delta_modem_device = { }, }; -static int __init ams_delta_modem_init(void) +static int __init late_init(void) { int err; if (!machine_is_ams_delta()) return -ENODEV; + /* Clear latch2 (NAND, LCD, modem enable) */ + ams_delta_latch2_write(~0, 0); + + platform_add_devices(late_devices, ARRAY_SIZE(late_devices)); + omap_cfg_reg(M14_1510_GPIO2); ams_delta_modem_ports[0].irq = gpio_to_irq(AMS_DELTA_GPIO_PIN_MODEM_IRQ); @@ -367,9 +372,16 @@ static int __init ams_delta_modem_init(void) AMS_DELTA_LATCH2_MODEM_NRESET | AMS_DELTA_LATCH2_MODEM_CODEC, AMS_DELTA_LATCH2_MODEM_NRESET | AMS_DELTA_LATCH2_MODEM_CODEC); - return platform_device_register(&ams_delta_modem_device); + err = platform_device_register(&ams_delta_modem_device); + if (err) + goto gpio_free; + return 0; + +gpio_free: + gpio_free(AMS_DELTA_GPIO_PIN_MODEM_IRQ); + return err; } -arch_initcall(ams_delta_modem_init); +late_initcall(late_init); static void __init ams_delta_map_io(void) { diff --git a/drivers/input/serio/ams_delta_serio.c b/drivers/input/serio/ams_delta_serio.c index d4d08bd..835d37a 100644 --- a/drivers/input/serio/ams_delta_serio.c +++ b/drivers/input/serio/ams_delta_serio.c @@ -165,7 +165,7 @@ serio: kfree(ams_delta_serio); return err; } -module_init(ams_delta_serio_init); +late_initcall(ams_delta_serio_init); static void __exit ams_delta_serio_exit(void) { diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index ccb8a6a..be81bc7 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -636,7 +636,7 @@ err: platform_device_put(ams_delta_audio_platform_device); return ret; } -module_init(ams_delta_module_init); +late_initcall(ams_delta_module_init); static void __exit ams_delta_module_exit(void) { -- 1.7.3.4

4 6

[alsa-devel] [PATCH] ASoC: samsung: Add .owner to struct snd_soc_card
by Axel Lin 22 Dec '11

22 Dec '11

Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/goni_wm8994.c | 1 + sound/soc/samsung/h1940_uda1380.c | 1 + sound/soc/samsung/jive_wm8750.c | 1 + sound/soc/samsung/littlemill.c | 1 + sound/soc/samsung/ln2440sbc_alc650.c | 1 + sound/soc/samsung/lowland.c | 1 + sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/samsung/s3c24xx_simtec_hermes.c | 1 + sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 1 + sound/soc/samsung/s3c24xx_uda134x.c | 1 + sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk2443_wm9710.c | 1 + sound/soc/samsung/smdk_spdif.c | 1 + sound/soc/samsung/smdk_wm8580.c | 1 + sound/soc/samsung/smdk_wm8580pcm.c | 1 + sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/smdk_wm8994pcm.c | 1 + sound/soc/samsung/smdk_wm9713.c | 1 + sound/soc/samsung/speyside.c | 1 + sound/soc/samsung/tobermory.c | 1 + 21 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index 84f9c3c..c23c2ae 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -244,6 +244,7 @@ static struct snd_soc_dai_link goni_dai[] = { static struct snd_soc_card goni = { .name = "goni", + .owner = THIS_MODULE, .dai_link = goni_dai, .num_links = ARRAY_SIZE(goni_dai), diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 03cfa5f..6e32577 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -215,6 +215,7 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = { static struct snd_soc_card h1940_asoc = { .name = "h1940", + .owner = THIS_MODULE, .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 8e523fd..1578663 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -127,6 +127,7 @@ static struct snd_soc_dai_link jive_dai = { /* jive audio machine driver */ static struct snd_soc_card snd_soc_machine_jive = { .name = "Jive", + .owner = THIS_MODULE, .dai_link = &jive_dai, .num_links = 1, diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 5cea59b..9dd818b 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -194,6 +194,7 @@ static int littlemill_late_probe(struct snd_soc_card *card) static struct snd_soc_card littlemill = { .name = "Littlemill", + .owner = THIS_MODULE, .dai_link = littlemill_dai, .num_links = ARRAY_SIZE(littlemill_dai), diff --git a/sound/soc/samsung/ln2440sbc_alc650.c b/sound/soc/samsung/ln2440sbc_alc650.c index cde38b8..69c4a59 100644 --- a/sound/soc/samsung/ln2440sbc_alc650.c +++ b/sound/soc/samsung/ln2440sbc_alc650.c @@ -34,6 +34,7 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = { static struct snd_soc_card ln2440sbc = { .name = "LN2440SBC", + .owner = THIS_MODULE, .dai_link = ln2440sbc_dai, .num_links = ARRAY_SIZE(ln2440sbc_dai), }; diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 4216a06..4adff93 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -177,6 +177,7 @@ static struct snd_soc_dapm_route audio_paths[] = { static struct snd_soc_card lowland = { .name = "Lowland", + .owner = THIS_MODULE, .dai_link = lowland_dai, .num_links = ARRAY_SIZE(lowland_dai), .aux_dev = lowland_aux_dev, diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7207189..7ac0ba2 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -465,6 +465,7 @@ static const struct gpio neo1973_gta02_gpios[] = {}; static struct snd_soc_card neo1973 = { .name = "neo1973", + .owner = THIS_MODULE, .dai_link = neo1973_dai, .num_links = ARRAY_SIZE(neo1973_dai), .aux_dev = neo1973_aux_devs, diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 71b4c02..21e1236 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -114,6 +114,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_card rx1950_asoc = { .name = "rx1950", + .owner = THIS_MODULE, .dai_link = rx1950_uda1380_dai, .num_links = ARRAY_SIZE(rx1950_uda1380_dai), diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 5027981..7ace6a8 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -89,6 +89,7 @@ static struct snd_soc_dai_link simtec_dai_aic33 = { /* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic33 = { .name = "Simtec-Hermes", + .owner = THIS_MODULE, .dai_link = &simtec_dai_aic33, .num_links = 1, diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 89b57b5..c42d5f0 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -78,6 +78,7 @@ static struct snd_soc_dai_link simtec_dai_aic23 = { /* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", + .owner = THIS_MODULE, .dai_link = &simtec_dai_aic23, .num_links = 1, diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 62b69fb..d731042 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -229,6 +229,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { static struct snd_soc_card snd_soc_s3c24xx_uda134x = { .name = "S3C24XX_UDA134X", + .owner = THIS_MODULE, .dai_link = &s3c24xx_uda134x_dai_link, .num_links = 1, }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index a22fc44..f2dcb42 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -198,6 +198,7 @@ static struct snd_soc_dai_link smartq_dai[] = { static struct snd_soc_card snd_soc_smartq = { .name = "SmartQ", + .owner = THIS_MODULE, .dai_link = smartq_dai, .num_links = ARRAY_SIZE(smartq_dai), diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c index 8bd1dc5..720ba29 100644 --- a/sound/soc/samsung/smdk2443_wm9710.c +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -30,6 +30,7 @@ static struct snd_soc_dai_link smdk2443_dai[] = { static struct snd_soc_card smdk2443 = { .name = "SMDK2443", + .owner = THIS_MODULE, .dai_link = smdk2443_dai, .num_links = ARRAY_SIZE(smdk2443_dai), }; diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index e0fd8ad..beaa9c1 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -160,6 +160,7 @@ static struct snd_soc_dai_link smdk_dai = { static struct snd_soc_card smdk = { .name = "SMDK-S/PDIF", + .owner = THIS_MODULE, .dai_link = &smdk_dai, .num_links = 1, }; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 81b4478..bff8758 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -203,6 +203,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk = { .name = "SMDK-I2S", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = 2, diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 49dfafb..fab5322 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -143,6 +143,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk_pcm = { .name = "SMDK-PCM", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = 2, }; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index ad9ac42..8eb309f 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -144,6 +144,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk = { .name = "SMDK-I2S", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = ARRAY_SIZE(smdk_dai), }; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 23c7fb7..77ecba9 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -124,6 +124,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk_pcm = { .name = "SMDK-PCM", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = 1, }; diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c index 31c6daf..8e26a73 100644 --- a/sound/soc/samsung/smdk_wm9713.c +++ b/sound/soc/samsung/smdk_wm9713.c @@ -50,6 +50,7 @@ static struct snd_soc_dai_link smdk_dai = { static struct snd_soc_card smdk = { .name = "SMDK WM9713", + .owner = THIS_MODULE, .dai_link = &smdk_dai, .num_links = 1, }; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 0222d86..f9ab770 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -291,6 +291,7 @@ static struct snd_soc_dapm_route audio_paths[] = { static struct snd_soc_card speyside = { .name = "Speyside", + .owner = THIS_MODULE, .dai_link = speyside_dai, .num_links = ARRAY_SIZE(speyside_dai), .aux_dev = speyside_aux_dev, diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 6f91c65..9199649 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -196,6 +196,7 @@ static int tobermory_late_probe(struct snd_soc_card *card) static struct snd_soc_card tobermory = { .name = "Tobermory", + .owner = THIS_MODULE, .dai_link = tobermory_dai, .num_links = ARRAY_SIZE(tobermory_dai), -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: pxa: Add .owner to struct snd_soc_card
by Axel Lin 22 Dec '11

22 Dec '11

Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/e740_wm9705.c | 1 + sound/soc/pxa/e750_wm9705.c | 1 + sound/soc/pxa/e800_wm9712.c | 1 + sound/soc/pxa/em-x270.c | 1 + sound/soc/pxa/hx4700.c | 1 + sound/soc/pxa/imote2.c | 1 + sound/soc/pxa/magician.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/raumfeld.c | 2 ++ sound/soc/pxa/saarb.c | 1 + sound/soc/pxa/spitz.c | 1 + sound/soc/pxa/tavorevb3.c | 1 + sound/soc/pxa/tosa.c | 1 + sound/soc/pxa/z2.c | 1 + sound/soc/pxa/zylonite.c | 1 + 17 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index b0e2fb7..5e5004a 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -317,6 +317,7 @@ static struct snd_soc_dai_link corgi_dai = { /* corgi audio machine driver */ static struct snd_soc_card snd_soc_corgi = { .name = "Corgi", + .owner = THIS_MODULE, .dai_link = &corgi_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 203ab78a..7b1bc2390 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -133,6 +133,7 @@ static struct snd_soc_dai_link e740_dai[] = { static struct snd_soc_card e740 = { .name = "Toshiba e740", + .owner = THIS_MODULE, .dai_link = e740_dai, .num_links = ARRAY_SIZE(e740_dai), }; diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 27f90cc..47b89d7 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -116,6 +116,7 @@ static struct snd_soc_dai_link e750_dai[] = { static struct snd_soc_card e750 = { .name = "Toshiba e750", + .owner = THIS_MODULE, .dai_link = e750_dai, .num_links = ARRAY_SIZE(e750_dai), }; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 858bf94..ea9707e 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -106,6 +106,7 @@ static struct snd_soc_dai_link e800_dai[] = { static struct snd_soc_card e800 = { .name = "Toshiba e800", + .owner = THIS_MODULE, .dai_link = e800_dai, .num_links = ARRAY_SIZE(e800_dai), }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index b13a425..64743a0 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -54,6 +54,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { static struct snd_soc_card em_x270 = { .name = "EM-X270", + .owner = THIS_MODULE, .dai_link = em_x270_dai, .num_links = ARRAY_SIZE(em_x270_dai), }; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 8260207..2a342c9 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -169,6 +169,7 @@ static struct snd_soc_dai_link hx4700_dai = { /* hx4700 audio machine driver */ static struct snd_soc_card snd_soc_card_hx4700 = { .name = "iPAQ hx4700", + .owner = THIS_MODULE, .dai_link = &hx4700_dai, .num_links = 1, .dapm_widgets = hx4700_dapm_widgets, diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index dc905ae..b93dafd 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -60,6 +60,7 @@ static struct snd_soc_dai_link imote2_dai = { static struct snd_soc_card imote2 = { .name = "Imote2", + .owner = THIS_MODULE, .dai_link = &imote2_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index e79f516..3f7a8ec 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -452,6 +452,7 @@ static struct snd_soc_dai_link magician_dai[] = { /* magician audio machine driver */ static struct snd_soc_card snd_soc_card_magician = { .name = "Magician", + .owner = THIS_MODULE, .dai_link = magician_dai, .num_links = ARRAY_SIZE(magician_dai), diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 0e73a7f..9c585af 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -181,6 +181,7 @@ static struct snd_soc_dai_link mioa701_dai[] = { static struct snd_soc_card mioa701 = { .name = "MioA701", + .owner = THIS_MODULE, .dai_link = mioa701_dai, .num_links = ARRAY_SIZE(mioa701_dai), }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index f313eca..db24bc6 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -146,6 +146,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { static struct snd_soc_card palm27x_asoc = { .name = "Palm/PXA27x", + .owner = THIS_MODULE, .dai_link = palm27x_dai, .num_links = ARRAY_SIZE(palm27x_dai), }; diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index b899a3b..ba15451 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -260,6 +260,7 @@ static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = static struct snd_soc_card snd_soc_raumfeld_connector = { .name = "Raumfeld Connector", + .owner = THIS_MODULE, .dai_link = snd_soc_raumfeld_connector_dai, .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), .suspend_post = raumfeld_analog_suspend, @@ -268,6 +269,7 @@ static struct snd_soc_card snd_soc_raumfeld_connector = { static struct snd_soc_card snd_soc_raumfeld_speaker = { .name = "Raumfeld Speaker", + .owner = THIS_MODULE, .dai_link = snd_soc_raumfeld_speaker_dai, .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), .suspend_post = raumfeld_analog_suspend, diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index d9467a2..2e21712 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -125,6 +125,7 @@ static struct snd_soc_dai_link saarb_dai[] = { static struct snd_soc_card snd_soc_card_saarb = { .name = "Saarb", + .owner = THIS_MODULE, .dai_link = saarb_dai, .num_links = ARRAY_SIZE(saarb_dai), }; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index c2d6ff9..bb06048 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -319,6 +319,7 @@ static struct snd_soc_dai_link spitz_dai = { /* spitz audio machine driver */ static struct snd_soc_card snd_soc_spitz = { .name = "Spitz", + .owner = THIS_MODULE, .dai_link = &spitz_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c index eeec892..4bef12c 100644 --- a/sound/soc/pxa/tavorevb3.c +++ b/sound/soc/pxa/tavorevb3.c @@ -125,6 +125,7 @@ static struct snd_soc_dai_link evb3_dai[] = { static struct snd_soc_card snd_soc_card_evb3 = { .name = "Tavor EVB3", + .owner = THIS_MODULE, .dai_link = evb3_dai, .num_links = ARRAY_SIZE(evb3_dai), }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 3f394de..564ef08 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -236,6 +236,7 @@ static struct snd_soc_dai_link tosa_dai[] = { static struct snd_soc_card tosa = { .name = "Tosa", + .owner = THIS_MODULE, .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), }; diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index b311ffe..d6807e0 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -202,6 +202,7 @@ static struct snd_soc_dai_link z2_dai = { /* z2 audio machine driver */ static struct snd_soc_card snd_soc_z2 = { .name = "Z2", + .owner = THIS_MODULE, .dai_link = &z2_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 580aae3..ceb6566 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -249,6 +249,7 @@ static int zylonite_resume_pre(struct snd_soc_card *card) static struct snd_soc_card zylonite = { .name = "Zylonite", + .owner = THIS_MODULE, .probe = &zylonite_probe, .remove = &zylonite_remove, .suspend_post = &zylonite_suspend_post, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: omap: Add .owner to struct snd_soc_card
by Axel Lin 22 Dec '11

22 Dec '11

Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/omap/am3517evm.c | 1 + sound/soc/omap/ams-delta.c | 1 + sound/soc/omap/igep0020.c | 1 + sound/soc/omap/n810.c | 1 + sound/soc/omap/omap3evm.c | 1 + sound/soc/omap/omap3pandora.c | 1 + sound/soc/omap/omap4-hdmi-card.c | 1 + sound/soc/omap/osk5912.c | 1 + sound/soc/omap/overo.c | 1 + sound/soc/omap/rx51.c | 1 + sound/soc/omap/sdp3430.c | 1 + sound/soc/omap/sdp4430.c | 1 + sound/soc/omap/zoom2.c | 1 + 13 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index c1cd4a0..add4866 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -107,6 +107,7 @@ static struct snd_soc_dai_link am3517evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_am3517evm = { .name = "am3517evm", + .owner = THIS_MODULE, .dai_link = &am3517evm_dai, .num_links = 1, diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index a04a433..3e523a7 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -597,6 +597,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = { /* Audio card driver */ static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", + .owner = THIS_MODULE, .dai_link = &ams_delta_dai_link, .num_links = 1, .set_bias_level = ams_delta_set_bias_level, diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index 591fbf8..ccae58a 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link igep2_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_igep2 = { .name = "igep2", + .owner = THIS_MODULE, .dai_link = &igep2_dai, .num_links = 1, }; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fc6209b..597be41 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -289,6 +289,7 @@ static struct snd_soc_dai_link n810_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_n810 = { .name = "N810", + .owner = THIS_MODULE, .dai_link = &n810_dai, .num_links = 1, diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 6857895..071fcb0 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -70,6 +70,7 @@ static struct snd_soc_dai_link omap3evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_omap3evm = { .name = "omap3evm", + .owner = THIS_MODULE, .dai_link = &omap3evm_dai, .num_links = 1, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 7605c37..07794bd 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -233,6 +233,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_omap3pandora = { .name = "omap3pandora", + .owner = THIS_MODULE, .dai_link = omap3pandora_dai, .num_links = ARRAY_SIZE(omap3pandora_dai), }; diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c index 52d471c..28d689b 100644 --- a/sound/soc/omap/omap4-hdmi-card.c +++ b/sound/soc/omap/omap4-hdmi-card.c @@ -74,6 +74,7 @@ static struct snd_soc_dai_link omap4_hdmi_dai = { static struct snd_soc_card snd_soc_omap4_hdmi = { .name = "OMAP4HDMI", + .owner = THIS_MODULE, .dai_link = &omap4_hdmi_dai, .num_links = 1, }; diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 351ec9d..d859b59 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -108,6 +108,7 @@ static struct snd_soc_dai_link osk_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", + .owner = THIS_MODULE, .dai_link = &osk_dai, .num_links = 1, diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index c3550ae..2ee889c 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link overo_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_overo = { .name = "overo", + .owner = THIS_MODULE, .dai_link = &overo_dai, .num_links = 1, }; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index ad16db5..fada6ef 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -383,6 +383,7 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = { /* Audio card */ static struct snd_soc_card rx51_sound_card = { .name = "RX-51", + .owner = THIS_MODULE, .dai_link = rx51_dai, .num_links = ARRAY_SIZE(rx51_dai), .aux_dev = rx51_aux_dev, diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index e8fbf8e..2c85066 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -213,6 +213,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", + .owner = THIS_MODULE, .dai_link = sdp3430_dai, .num_links = ARRAY_SIZE(sdp3430_dai), diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 2735fa0..175ba9a 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -226,6 +226,7 @@ static struct snd_soc_dai_link sdp4430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp4430 = { .name = "SDP4430", + .owner = THIS_MODULE, .dai_link = sdp4430_dai, .num_links = ARRAY_SIZE(sdp4430_dai), diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 7641a7f..981616d 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -157,6 +157,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_zoom2 = { .name = "Zoom2", + .owner = THIS_MODULE, .dai_link = zoom2_dai, .num_links = ARRAY_SIZE(zoom2_dai), -- 1.7.5.4

2 1

[alsa-devel] [PATCH V5] ASoC: Allow DAI links to be specified using device tree nodes
by Stephen Warren 22 Dec '11

22 Dec '11

DAI link endpoints and platform (DMA) devices are currently specified by name. When instantiating sound cards from device tree, it may be more convenient to refer to these devices by phandle in the device tree, and for code to describe DAI links using the "struct device_node *" ("of_node") those phandles map to. This change adds new fields to snd_soc_dai_link which can "name" devices using of_node, enhances soc_bind_dai_link() to allow binding based on of_node, and enhances snd_soc_register_card() to ensure that illegal combinations of name and of_node are not used. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- v2: New patch implementing a new feature v3: Add Liam's ack v4: Rebased to re-order this patch directly onto ASoC for-next v5: Rebased again now that my other patches are ASoC for-next include/sound/soc.h | 4 +++ sound/soc/soc-core.c | 64 ++++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 61 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index db8acd2..8391b0e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -231,6 +231,7 @@ enum snd_soc_bias_level { SND_SOC_BIAS_ON = 3, }; +struct device_node; struct snd_jack; struct snd_soc_card; struct snd_soc_pcm_stream; @@ -703,8 +704,11 @@ struct snd_soc_dai_link { const char *name; /* Codec name */ const char *stream_name; /* Stream name */ const char *codec_name; /* for multi-codec */ + const struct device_node *codec_of_node; const char *platform_name; /* for multi-platform */ + const struct device_node *platform_of_node; const char *cpu_dai_name; + const struct device_node *cpu_dai_of_node; const char *codec_dai_name; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 42ad2db..a4592cb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -764,8 +764,13 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { - if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; + if (dai_link->cpu_dai_of_node) { + if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) + continue; + } else { + if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; + } rtd->cpu_dai = cpu_dai; goto find_codec; @@ -781,8 +786,13 @@ find_codec: /* no, then find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { - if (strcmp(codec->name, dai_link->codec_name)) - continue; + if (dai_link->codec_of_node) { + if (codec->dev->of_node != dai_link->codec_of_node) + continue; + } else { + if (strcmp(codec->name, dai_link->codec_name)) + continue; + } rtd->codec = codec; @@ -814,13 +824,19 @@ find_platform: /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; - if (!platform_name) + if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; /* no, then find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { - if (strcmp(platform->name, platform_name)) - continue; + if (dai_link->platform_of_node) { + if (platform->dev->of_node != + dai_link->platform_of_node) + continue; + } else { + if (strcmp(platform->name, platform_name)) + continue; + } rtd->platform = platform; goto out; @@ -2831,6 +2847,40 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai_link *link = &card->dai_link[i]; + + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codec_name == !!link->codec_of_node) { + dev_err(card->dev, + "Neither/both codec name/of_node are set\n"); + return -EINVAL; + } + + /* + * Platform may be specified by either name or OF node, but + * can be left unspecified, and a dummy platform will be used. + */ + if (link->platform_name && link->platform_of_node) { + dev_err(card->dev, + "Both platform name/of_node are set\n"); + return -EINVAL; + } + + /* + * CPU DAI must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { + dev_err(card->dev, + "Neither/both cpu_dai name/of_node are set\n"); + return -EINVAL; + } + } + dev_set_drvdata(card->dev, card); snd_soc_initialize_card_lists(card); -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/3 v5] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Tomoya MORINAGA 22 Dec '11

22 Dec '11

ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high resistance to voltage noise. On chip regulator realizes power supply rejection ratio be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ ML26124-02GD can deliver stable audio performance without being affected by noise from the power supply circuit and peripheral components. The chip also includes a composite video signal output, which can be applied to various portable device requirements. The ML26124 is realized these functions into very small package the size is only 2.56mm x 2.46mm therefore can be construct high quality sound system easily. ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package. Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- V5: Add clock setting processing --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/ml26124.c | 594 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 129 ++++++++++ 4 files changed, 728 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ml26124.c create mode 100644 sound/soc/codecs/ml26124.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7c205e7..2b4435f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_ML26124 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -224,6 +225,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_ML26124 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9aa6e66..22adcbd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-ml26124-objs := ml26124.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 0000000..fe2dc26 --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,594 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +struct ml26124_priv { + u32 mclk; + u32 channels; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limitter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limitter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("Zero Cross Comparator Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limitter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limitter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in Switch", ML26124_SPK_AMP_OUT, 3, 1, 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", + "Analog MIC Differential in", + "Digital MIC in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 2, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_MICBIAS("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0), + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLK"}, + {"ADC", NULL, "MCLK"}, + {"DAC", NULL, "PLL"}, + {"ADC", NULL, "PLL"}, + + /* output mixer */ + {"Output Mixer", "PGA Switch", "PGA"}, + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", "Line in Switch", "LIN"}, + + /* outputs */ + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + + /* input */ + {"Input Mux", "Analog MIC SingleEnded in", "MIN"}, + {"Input Mux", "Analog MIC Differential in", "MIN"}, + {"ADC", NULL, "Input Mux"}, +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288800, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->channels = params_channels(hw_params); + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + return 0; +} + +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned char mask = ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + /* SAI configuration */ + if (priv->channels == 1) { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + } else { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY); + } + + return 0; +} + +static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ML26124_USE_PLL: + break; + default: + return -EINVAL; + } + + switch (freq) { + case 12288000: + priv->mclk = freq; + break; + default: + return -EINVAL; + } + return 0; +} + +#define ML26134_CACHESIZE 79 +static const u16 ml26124_reg[ML26134_CACHESIZE] = { + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, + 0x0, 0x0, 0x0, /* System Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* Power Management */ + 0x4, /* Analog Reference Control */ + 0x10, 0x0, 0x33, 0x0, 0x0, /* Input/Output Amplifier Control */ + 0x0, 0x0, 0x1, /* Analog Path Contorl */ + 0x0, 0x0, 0x0, /* Audio Interface Control */ + 0x1, 0x0, 0x0, 0xff, 0xff, 0x10, /* DSP Control */ + 0x7, 0x7, 0x7, 0x7, 0x7, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x2, 0x3, 0x0, 0xb, 0x70, 0x0, 0x0, /* ALC Control */ + 0x4, 0x5, 0xd, 0x70, 0x10, 0x0, /* Playback Limiter Control */ + 0x1, 0x1, 0x1 /* Video Amplifer Control */ +}; + +#define ML26124_VMID BIT(0) + +static void ml26124_sync_cache(struct snd_soc_codec *codec) +{ + u8 *reg_cache = codec->reg_cache; + int i; + + if (!codec->cache_sync) + return; + + codec->cache_only = 0; + + for (i = 0; i < codec->driver->reg_cache_size; i++) { + if (reg_cache[i] == ml26124_reg[i]) + continue; + + snd_soc_write(codec, i, reg_cache[i]); + } + + codec->cache_sync = 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* VMID ON */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ml26124_sync_cache(codec); + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int ml26124_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) { + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0); + return 0; + } else if (cmd == SNDRV_PCM_TRIGGER_START) { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 1); + else + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 2); + } + + return 0; +} + +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, + .trigger = ml26124_pcm_trigger, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .reg_cache_size = ML26134_CACHESIZE, + .reg_word_size = sizeof(u8), + .reg_cache_default = ml26124_reg, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +static int __init ml26124_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&ml26124_i2c_driver); + if (ret != 0) + pr_err("Failed to register ML26124 I2C driver: %d\n", ret); + + return ret; +} +module_init(ml26124_modinit); + +static void __exit ml26124_exit(void) +{ + i2c_del_driver(&ml26124_i2c_driver); +} +module_exit(ml26124_exit); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 0000000..4fe7337 --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,129 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +/* Clock select for machine driver */ +#define ML26124_USE_PLL 0 +#define ML26124_USE_MCLKI_256FS 1 +#define ML26124_USE_MCLKI_512FS 2 +#define ML26124_USE_MCLKI_1024FS 3 + + +#endif -- 1.7.4.4

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Re: [alsa-devel] [PATCH 07/10] ASoC: OMAP: ams-delta: Drive modem/codec pins over GPIO API
by Liam Girdwood 22 Dec '11

22 Dec '11

On Sun, 2011-12-11 at 21:12 +0100, Janusz Krzysztofik wrote: > Don't use Amstrad Delta custom I/O functions any longer, replace them > with GPIO. Old pin definitions, no longer used by the modem bits either, > can be dropped. > > Depends on patch 2/10 "ARM: OMAP1: Convert Amstrad E3 latches to > basic_mmio_gpio". > > Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> Applied. Thanks Liam

3 2

[alsa-devel] [PATCH 00/21] ALSA: asihpi - update to 4.10
by linux＠audioscience.com 22 Dec '11

22 Dec '11

2 22

[alsa-devel] [PATCH] ALSA: asihpi - fix pcm dma pointer tracking
by linux＠audioscience.com 22 Dec '11

22 Dec '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Elapsed counter should only count data committed to snd_pcm_period_elapsed, rather than all data available Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/asihpi.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e9de799..322ddfe 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -888,8 +888,8 @@ static void snd_card_asihpi_timer_function(unsigned long data) pd, xfer2)); } } - ds->pcm_buf_host_rw_ofs = ds->pcm_buf_host_rw_ofs + xfercount; - ds->pcm_buf_elapsed_dma_ofs = pcm_buf_dma_ofs; + ds->pcm_buf_host_rw_ofs = += xfercount; + ds->pcm_buf_elapsed_dma_ofs += xfercount; snd_pcm_period_elapsed(s); } } -- 1.7.0.4

2 1

[alsa-devel] how to enumerate useful ALSA devices?
by Joerg-Cyril.Hoehle＠t-systems.com 22 Dec '11

22 Dec '11

Hi, The 2008 audio guide recommends http://0pointer.de/blog/projects/guide-to-sound-apis.html "Do not use the snd_card_xxx() APIs. For enumerating use snd_device_name_hint() (and related functions). snd_card_xxx() is obsolete... Bluetooth audio are not included" Is that correct advice? Wine still uses snd_card, so it knows only about "default" and "plughw:x,y". I believe that's wrong but don't know what should be used instead in order not to come up with a long list of artificial names mentioned in /usr/share/alsa/alsa.conf such as surround51 when the poor on-board card knows nothing but 16 bit stereo. I'm looking for a list of working devices on this machine, which likely should contain "default" (usually leading to PA on a recent Ubuntu machine), "plug:dmix" (PA getting out of the way when trying to use other devices), "hw:0,0" and presumably not much more. Thank you for your help, Jörg Höhle

2 1

[alsa-devel] ASoC: tegra_wm8903 build error
by Axel Lin 22 Dec '11

22 Dec '11

Seems we need below patch to fix the build error: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-December/047403.h… CC sound/soc/tegra/tegra_wm8903.o sound/soc/tegra/tegra_wm8903.c: In function 'tegra_wm8903_driver_probe': sound/soc/tegra/tegra_wm8903.c:411: error: 'struct snd_soc_dai_link' has no member named 'codec_of_node' sound/soc/tegra/tegra_wm8903.c:413: error: 'struct snd_soc_dai_link' has no member named 'codec_of_node' sound/soc/tegra/tegra_wm8903.c:421: error: 'struct snd_soc_dai_link' has no member named 'cpu_dai_of_node' sound/soc/tegra/tegra_wm8903.c:423: error: 'struct snd_soc_dai_link' has no member named 'cpu_dai_of_node' make[3]: *** [sound/soc/tegra/tegra_wm8903.o] Error 1 make[2]: *** [sound/soc/tegra] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Regards, Axel

1 0

[alsa-devel] [PATCH 00/10] OMAP4: ASoC: Support for PandaBoard family
by Peter Ujfalusi 21 Dec '11

21 Dec '11

Hello, the following series will add ASoC support for PandaBoards. PandaBoards have different audio routings compared to SDP4430/Blaze boards, but the differences not that big to justify a new ASoC machine driver. Main changes: - Rename the sdp4430 ASoC machine driver to use generic name: omap-abe-twl6040 - Convert the ASoC machine driver to platform driver - The type of the board, and the desired sound card name is passed via platform data to the ASoC machine driver - Based on the board type the driver selects different audio routings - Registration of the needed platform devices in board files (sdp4403, panda) After this series the sound card names will be different for easier UCM integration: "OMAP4-SDP4430" for SDP4430/Blaze boards "OMAP4-Panda" for PandaBoard 4430 "OMAP4-PandaES" for PandaBoard ES (4460) The series has been tested on Blaze, and PandaBoard ES. Regards, Peter --- Peter Ujfalusi (10): ASoC: sdp4430: Correct author e-mail address ASoC: OMAP4: Rename the sdp4430 machine driver ASoC: omap-abe-twl6040: Correct internal prefix, Kconfig entry include: platform_data: Platform data header for OMAP4 ASoC audio OMAP4: 4430sdp: Register platform device for OMAP4 audio ASoC: omap-abe-twl6040: Convert to platform deriver ASoC: omap-abe-twl6040: Add support for PandaBoard OMAP4: omap4panda: Enable audio support ASoC: omap-abe-twl6040: Add missing audio route information ASoC: omap-abe-twl6040: Fix DAPM widget type for FM input arch/arm/mach-omap2/board-4430sdp.c | 15 ++ arch/arm/mach-omap2/board-omap4panda.c | 48 +++++- include/linux/platform_data/omap-abe-twl6040.h | 33 ++++ sound/soc/omap/Kconfig | 14 +- sound/soc/omap/Makefile | 4 +- sound/soc/omap/{sdp4430.c => omap-abe-twl6040.c} | 198 ++++++++++++++++------ 6 files changed, 251 insertions(+), 61 deletions(-) create mode 100644 include/linux/platform_data/omap-abe-twl6040.h rename sound/soc/omap/{sdp4430.c => omap-abe-twl6040.c} (52%) -- 1.7.8

4 28

[alsa-devel] [PATCH] [salsa] Fix 4bit enable/disable flag
by Arun Raghavan 21 Dec '11

21 Dec '11

The documentation and implementation disagree on the name of the switch. --- configure.ac | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/configure.ac b/configure.ac index 6a55813..b72adf3 100644 --- a/configure.ac +++ b/configure.ac @@ -116,7 +116,7 @@ AC_ARG_ENABLE(float, [support floatin-point unit]), support_float="$enableval", support_float="no") -AC_ARG_ENABLE(support_4bit, +AC_ARG_ENABLE(4bit, AS_HELP_STRING([--disable-4bit], [drop the support for 4bit PCM (IMA ADPCM)]), support_4bit="$enableval", support_4bit="yes") -- 1.7.8

2 1

[alsa-devel] [PATCH] Implement snd_mixer_selem_ask_*
by Arun Raghavan 21 Dec '11

21 Dec '11

This implements the snd_mixer_selem_ask_* family of functions. These are needed by PulseAudio. --- src/mixer.c | 19 +++++++++++++++ src/mixer_macros.h | 64 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 83 insertions(+), 0 deletions(-) diff --git a/src/mixer.c b/src/mixer.c index 3f87eca..c502d33 100644 --- a/src/mixer.c +++ b/src/mixer.c @@ -1067,6 +1067,25 @@ int _snd_selem_vol_get_dB(snd_selem_vol_item_t *item, int channel, item->vol[RAW_IDX(channel)], value); } +int _snd_selem_ask_vol_dB(snd_selem_vol_item_t *item, long value, long *dBvalue) +{ + if (init_db_info(item) < 0) + return -EINVAL; + return snd_tlv_convert_to_dB(item->db_info, + item->raw_min, item->raw_max, + value, dBvalue); +} + +int _snd_selem_ask_dB_vol(snd_selem_vol_item_t *item, long dBvalue, long *value, + int xdir) +{ + if (init_db_info(item) < 0) + return -EINVAL; + return snd_tlv_convert_from_dB(item->db_info, + item->raw_min, item->raw_max, + dBvalue, value, xdir); +} + int _snd_selem_vol_get_dB_range(snd_selem_vol_item_t *item, long *min, long *max) { diff --git a/src/mixer_macros.h b/src/mixer_macros.h index 8c8ae86..4c8d9c5 100644 --- a/src/mixer_macros.h +++ b/src/mixer_macros.h @@ -564,6 +564,11 @@ extern int _snd_selem_vol_get_dB(snd_selem_vol_item_t *item, int channel, extern int _snd_selem_vol_set_dB(snd_selem_vol_item_t *item, snd_mixer_selem_channel_id_t channel, long db_gain, int xdir); +extern int _snd_selem_ask_vol_dB(snd_selem_vol_item_t *item, long value, + long *dBvalue); +extern int _snd_selem_ask_dB_vol(snd_selem_vol_item_t *item, long dBvalue, + long *value, int xdir); + __SALSA_EXPORT_FUNC int snd_mixer_selem_get_playback_dB_range(snd_mixer_elem_t *elem, long *min, long *max) @@ -582,6 +587,22 @@ int snd_mixer_selem_get_playback_dB(snd_mixer_elem_t *elem, } __SALSA_EXPORT_FUNC +int snd_mixer_selem_ask_playback_vol_dB(snd_mixer_elem_t *elem, long value, + long *dBvalue) +{ + return _snd_selem_ask_vol_dB((snd_selem_vol_item_t *)elem->items[SND_SELEM_ITEM_PVOLUME], + value, dBvalue); +} + +__SALSA_EXPORT_FUNC +int snd_mixer_selem_ask_playback_dB_vol(snd_mixer_elem_t *elem, long dBvalue, + int dir, long *value) +{ + return _snd_selem_ask_dB_vol((snd_selem_vol_item_t *)elem->items[SND_SELEM_ITEM_PVOLUME], + dBvalue, value, dir); +} + +__SALSA_EXPORT_FUNC int snd_mixer_selem_set_playback_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) @@ -628,6 +649,22 @@ int snd_mixer_selem_get_capture_dB(snd_mixer_elem_t *elem, } __SALSA_EXPORT_FUNC +int snd_mixer_selem_ask_capture_vol_dB(snd_mixer_elem_t *elem, long value, + long *dBvalue) +{ + return _snd_selem_ask_vol_dB((snd_selem_vol_item_t *)elem->items[SND_SELEM_ITEM_CVOLUME], + value, dBvalue); +} + +__SALSA_EXPORT_FUNC +int snd_mixer_selem_ask_capture_dB_vol(snd_mixer_elem_t *elem, long dBvalue, + int dir, long *value) +{ + return _snd_selem_ask_dB_vol((snd_selem_vol_item_t *)elem->items[SND_SELEM_ITEM_CVOLUME], + dBvalue, value, dir); +} + +__SALSA_EXPORT_FUNC int snd_mixer_selem_set_capture_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) @@ -673,6 +710,19 @@ int snd_mixer_selem_get_playback_dB(snd_mixer_elem_t *elem, } __SALSA_EXPORT_FUNC __SALSA_NOT_IMPLEMENTED +int snd_mixer_selem_ask_playback_vol_dB(snd_mixer_elem_t *elem, long value, + long *dBvalue) +{ + return -ENXIO; +} + +__SALSA_EXPORT_FUNC __SALSA_NOT_IMPLEMENTED +int snd_mixer_selem_ask_playback_dB_vol(snd_mixer_elem_t *elem, long dBvalue, + int dir, long *value) +{ + return -ENXIO; +} +__SALSA_EXPORT_FUNC __SALSA_NOT_IMPLEMENTED int snd_mixer_selem_set_playback_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) @@ -703,6 +753,20 @@ int snd_mixer_selem_get_capture_dB(snd_mixer_elem_t *elem, } __SALSA_EXPORT_FUNC __SALSA_NOT_IMPLEMENTED +int snd_mixer_selem_ask_capture_vol_dB(snd_mixer_elem_t *elem, long value, + long *dBvalue) +{ + return -ENXIO; +} + +__SALSA_EXPORT_FUNC __SALSA_NOT_IMPLEMENTED +int snd_mixer_selem_ask_capture_dB_vol(snd_mixer_elem_t *elem, long value, + int dir, long *dBvalue) +{ + return -ENXIO; +} + +__SALSA_EXPORT_FUNC __SALSA_NOT_IMPLEMENTED int snd_mixer_selem_set_capture_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) -- 1.7.8

2 2

[alsa-devel] [PATCH] hda - add ALC889 quick id for Intel Los Lunas 2 desktop reference board
by Wu Fengguang 21 Dec '11

21 Dec '11

Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/alc882_quirks.c | 1 + 1 file changed, 1 insertion(+) --- linux.orig/sound/pci/hda/alc882_quirks.c 2011-12-20 14:16:51.880000058 +0800 +++ linux/sound/pci/hda/alc882_quirks.c 2011-12-20 14:16:58.740000115 +0800 @@ -2955,6 +2955,7 @@ static const struct snd_pci_quirk alc882 SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL), + SND_PCI_QUIRK(0x8086, 0x7270, "Intel Los Lunas 2", ALC889_INTEL), SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL), SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC882_6ST_DIG),

3 7

[alsa-devel] ASoC: missed hx4700-audio platform device for hx4700
by Axel Lin 21 Dec '11

21 Dec '11

The iPAQ hx4700 machine driver needs a hx4700-audio platform device which is not (yet) merged. The patchs was post on LKML: https://lkml.org/lkml/2011/5/18/222 but [PATCH v3 3/3] pxa/hx4700: add platform device and I2C info for AK4641 codec is missed. Regards, Axel

3 2

[alsa-devel] [PATCH] ASoC: Remove ifdefs for GPIO_SYSFS
by Mark Brown 21 Dec '11

21 Dec '11

It is part of the GPIO API so should be stubbed appropriately. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-jack.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 6c5ebd3..ee4353f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -341,10 +341,8 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpios[i].gpio, ret); } -#ifdef CONFIG_GPIO_SYSFS /* Expose GPIO value over sysfs for diagnostic purposes */ gpio_export(gpios[i].gpio, false); -#endif /* Update initial jack status */ snd_soc_jack_gpio_detect(&gpios[i]); @@ -376,9 +374,7 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, int i; for (i = 0; i < count; i++) { -#ifdef CONFIG_GPIO_SYSFS gpio_unexport(gpios[i].gpio); -#endif free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); cancel_delayed_work_sync(&gpios[i].work); gpio_free(gpios[i].gpio); -- 1.7.7.3

1 0

Re: [alsa-devel] [PATCH v2 1/7][RESEND] ARM: OMAP1: ams-delta: register latch dependent devices later
by Tony Lindgren 20 Dec '11

20 Dec '11

* Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> [111220 12:03]: > On Tuesday 20 of December 2011 at 19:06:11, Tony Lindgren wrote: > > > To deal with the init order issues, you can pass a set_power > > function pointer in platform_data that the driver can use. Or > > set up a fixed regulator for it. > > I'm already working on the follow up patch series mentioned in the cover > letter, and yes, I'm using the fixed regulator. Be patient, please, as > my resources are somehow limited. Yes, I understand. For this patch I'm fine with the late_initcall changes Russell suggested, then the rest can be done in separate patches later on. Regards, Tony

1 0

Re: [alsa-devel] [PATCH v2 1/7][RESEND] ARM: OMAP1: ams-delta: register latch dependent devices later
by Russell King - ARM Linux 20 Dec '11

20 Dec '11

On Tue, Dec 20, 2011 at 12:28:32AM +0100, Janusz Krzysztofik wrote: > diff --git a/drivers/input/serio/ams_delta_serio.c b/drivers/input/serio/ams_delta_serio.c > index d4d08bd..56ffd7c 100644 > --- a/drivers/input/serio/ams_delta_serio.c > +++ b/drivers/input/serio/ams_delta_serio.c > @@ -165,6 +165,9 @@ serio: > kfree(ams_delta_serio); > return err; > } > +#ifndef MODULE > +late_initcall(ams_delta_serio_init); > +#else > module_init(ams_delta_serio_init); > > static void __exit ams_delta_serio_exit(void) > @@ -175,3 +178,4 @@ static void __exit ams_delta_serio_exit(void) > gpio_free(AMS_DELTA_GPIO_PIN_KEYBRD_DATA); > } > module_exit(ams_delta_serio_exit); > +#endif It's worth noting: #ifndef MODULE #define late_initcall(fn) __define_initcall("7",fn,7) #else /* MODULE */ #define late_initcall(fn) module_init(fn) So really, all these ifndefs aren't required. Just change the module_init() to late_initcall().

2 1

[alsa-devel] [PATCH 0/5] snd-usb endpoint rework, take 4
by Daniel Mack 20 Dec '11

20 Dec '11

As stated in reply to my previous series, the last version caused deadlocks when quickly starting and stopping the PCM streams. These should finally be fixed now. Tested with both the FTU and a sync-endpoint driven device. Please check and report :) Daniel Daniel Mack (5): ALSA: snd-usb: add snd_usb_audio-wide mutex ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: add support for implicit feedback sound/usb/card.c | 10 +- sound/usb/card.h | 63 +++ sound/usb/endpoint.c | 1458 ++++++++++++++++++++++++++------------------------ sound/usb/endpoint.h | 32 +- sound/usb/pcm.c | 435 +++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 2 + 7 files changed, 1229 insertions(+), 802 deletions(-) -- 1.7.5.4

3 10

Re: [alsa-devel] [PATCH v2 1/7][RESEND] ARM: OMAP1: ams-delta: register latch dependent devices later
by Tony Lindgren 20 Dec '11

20 Dec '11

* Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> [111219 14:57]: > Resending because of a typo in the Cc: list, sorry. > > 8<-------------------------- > > In preparation to converting Amstrad Delta on-board latches to > basic_mmio_gpio devices, registration of platform devices which depend > on latches and will require initialization of their GPIO pins first, > should be moved out of .machine_init down to late_initcall level, as the > gpio-generic driver is not available until device_initcall time. The > latch reset operation, which will be replaced with GPIO initialization, > must also be moved to late_initcall for the same reason. > > Since there was already another, separate arch_initcall function for > setting up one of those latch dependent devices, the on-board modem > device, reuse that function, i.e., rename it to a name that matches the > new purpose, extend with other device setup relocated from > .machine_init, and move down to the late_initcall level. > > While being at it, add missing gpio_free() in case the modem platform > device registration fails. > > In addition, defer registration of the Amstrad Delta ASoC and serio > devices, done from their device driver files, until late_initcall time, > as those drivers will depend on their GPIO pins already requested from > the board late_init() function until updated to register their GPIO pins > themselves. > > Thanks to Tony Lindgren <tony(a)atomide.com> who suggested this approach > instead of shifting up the gpio-generic driver initialization. ... > --- a/drivers/input/serio/ams_delta_serio.c > +++ b/drivers/input/serio/ams_delta_serio.c > @@ -165,6 +165,9 @@ serio: > kfree(ams_delta_serio); > return err; > } > +#ifndef MODULE > +late_initcall(ams_delta_serio_init); > +#else > module_init(ams_delta_serio_init); > > static void __exit ams_delta_serio_exit(void) > @@ -175,3 +178,4 @@ static void __exit ams_delta_serio_exit(void) > gpio_free(AMS_DELTA_GPIO_PIN_KEYBRD_DATA); > } > module_exit(ams_delta_serio_exit); > +#endif > diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c > index ccb8a6a..1764a3b 100644 > --- a/sound/soc/omap/ams-delta.c > +++ b/sound/soc/omap/ams-delta.c > @@ -636,6 +636,9 @@ err: > platform_device_put(ams_delta_audio_platform_device); > return ret; > } > +#ifndef MODULE > +late_initcall(ams_delta_module_init); > +#else > module_init(ams_delta_module_init); > > static void __exit ams_delta_module_exit(void) > @@ -657,6 +660,7 @@ static void __exit ams_delta_module_exit(void) > platform_device_unregister(ams_delta_audio_platform_device); > } > module_exit(ams_delta_module_exit); > +#endif > > MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>"); > MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone"); This looks a bit odd.. I think these drivers should be converted to a proper platform_driver so you can get rid of the machine_is_ams_delta check in the init. Then you can rely on the probe to match the device. To deal with the init order issues, you can pass a set_power function pointer in platform_data that the driver can use. Or set up a fixed regulator for it. Regards, Tony

1 0

[alsa-devel] [PATCH v3] ASoC: Tegra machine ASoC driver for boards using ALC5332 codec
by Leon Romanovsky 20 Dec '11

20 Dec '11

At this stage only Toshiba AC100/Dynabook supported. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> -- Changes since v2: * Remove unsupported frequencies (more then 48Khz) - Suggested by Stephen Warren * Remove GPIO related code, it will be added later by platform_data patchset. Changes since v1: * Use module_platform_driver. * Move controls, dapm_widgets and dapm_routes to snd_soc_card structure. * Set dai_fmt in snd_soc_dai_link structure instead probe function. * Remove check board in probe function * Fix warning about unused codec_dai. * Use devm_kzalloc to manage memory allocation. --- sound/soc/tegra/Kconfig | 9 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_alc5632.c | 233 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 244 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/tegra_alc5632.c diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index c6af1fd..ce1b773 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -47,3 +47,12 @@ config SND_SOC_TEGRA_TRIMSLICE help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. + +config SND_SOC_TEGRA_ALC5632 + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 4d943b3..8e584b8 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -14,6 +14,8 @@ obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o +snd-soc-tegra-alc5632-objs := tegra_alc5632.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o +obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c new file mode 100644 index 0000000..89697c6 --- /dev/null +++ b/sound/soc/tegra/tegra_alc5632.c @@ -0,0 +1,213 @@ +/* +* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Marc Dietrich <marvin24(a)gmx.de> +* +* 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. +*/ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/alc5632.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-alc5632" + +struct tegra_alc5632 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_alc5632_asoc_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->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + mclk = 512 * srate; + + err = tegra_asoc_utils_set_rate(&alc5632->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_alc5632_asoc_ops = { + .hw_params = tegra_alc5632_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_alc5632_hs_jack; + +static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Int Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = { + /* Internal Speaker */ + {"Int Spk", NULL, "SPKOUT"}, + {"Int Spk", NULL, "SPKOUTN"}, + + /* Headset Mic */ + {"MIC1", NULL, "MICBIAS1"}, + {"MICBIAS1", NULL, "Headset Mic"}, + + /* Headset Stereophone */ + {"Headset Stereophone", NULL, "HPR"}, + {"Headset Stereophone", NULL, "HPL"}, +}; + +static const struct snd_kcontrol_new tegra_alc5632_controls[] = { + SOC_DAPM_PIN_SWITCH("Int Spk"), +}; + +static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack); + snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins), + tegra_alc5632_hs_jack_pins); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static struct snd_soc_dai_link tegra_alc5632_dai = { + .name = "ALC5632", + .stream_name = "ALC5632 PCM", + .codec_name = "alc5632.0-001e", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "alc5632-hifi", + .init = tegra_alc5632_asoc_init, + .ops = &tegra_alc5632_asoc_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_alc5632 = { + .name = "tegra-alc5632", + .dai_link = &tegra_alc5632_dai, + .num_links = 1, + .controls = tegra_alc5632_controls, + .num_controls = ARRAY_SIZE(tegra_alc5632_controls), + .dapm_widgets = tegra_alc5632_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_alc5632_dapm_widgets), + .dapm_routes = tegra_alc5632_audio_map, + .num_dapm_routes = ARRAY_SIZE(tegra_alc5632_audio_map), + .fully_routed = true, +}; + +static __devinit int tegra_alc5632_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_alc5632; + struct tegra_alc5632 *alc5632; + int ret; + + alc5632 = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_alc5632), GFP_KERNEL); + if (!alc5632) { + dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + return ret; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, alc5632); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + tegra_asoc_utils_fini(&alc5632->util_data); + return ret; + } + + return 0; +} + +static int __devexit tegra_alc5632_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&alc5632->util_data); + + return 0; +} + +static struct platform_driver tegra_alc5632_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tegra_alc5632_probe, + .remove = __devexit_p(tegra_alc5632_remove), +}; +module_platform_driver(tegra_alc5632_driver); + +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.3.4

5 17

[alsa-devel] [PATCH] ASoC: Convert cs42l51 to table based DAPM and control init
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs42l51.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 528510b..ffce9f2 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -511,7 +511,6 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_probe(struct snd_soc_codec *codec) { struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, reg; ret = cs42l51_fill_cache(codec); @@ -539,20 +538,20 @@ static int cs42l51_probe(struct snd_soc_codec *codec) if (ret < 0) return ret; - snd_soc_add_controls(codec, cs42l51_snd_controls, - ARRAY_SIZE(cs42l51_snd_controls)); - snd_soc_dapm_new_controls(dapm, cs42l51_dapm_widgets, - ARRAY_SIZE(cs42l51_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, cs42l51_routes, - ARRAY_SIZE(cs42l51_routes)); - return 0; } static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { - .probe = cs42l51_probe, + .probe = cs42l51_probe, .reg_cache_size = CS42L51_NUMREGS + 1, .reg_word_size = sizeof(u8), + + .controls = cs42l51_snd_controls, + .num_controls = ARRAY_SIZE(cs42l51_snd_controls), + .dapm_widgets = cs42l51_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l51_dapm_widgets), + .dapm_routes = cs42l51_routes, + .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), }; static int cs42l51_i2c_probe(struct i2c_client *i2c_client, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Convert 88pm860x-codec to table based DAPM and control init
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/88pm860x-codec.c | 15 ++++++++------- 1 files changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 2d39123..99ca53c 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -861,7 +861,7 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route pm860x_dapm_routes[] = { /* supply */ {"Left DAC", NULL, "VCODEC"}, {"Right DAC", NULL, "VCODEC"}, @@ -1361,7 +1361,6 @@ EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect); static int pm860x_probe(struct snd_soc_codec *codec) { struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int i, ret; pm860x->codec = codec; @@ -1388,11 +1387,6 @@ static int pm860x_probe(struct snd_soc_codec *codec) goto out; } - snd_soc_add_controls(codec, pm860x_snd_controls, - ARRAY_SIZE(pm860x_snd_controls)); - snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets, - ARRAY_SIZE(pm860x_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; out: @@ -1420,6 +1414,13 @@ static struct snd_soc_codec_driver soc_codec_dev_pm860x = { .reg_cache_size = REG_CACHE_SIZE, .reg_word_size = sizeof(u8), .set_bias_level = pm860x_set_bias_level, + + .controls = pm860x_snd_controls, + .num_controls = ARRAY_SIZE(pm860x_snd_controls), + .dapm_widgets = pm860x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pm860x_dapm_widgets), + .dapm_routes = pm860x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pm860x_dapm_routes), }; static int __devinit pm860x_codec_probe(struct platform_device *pdev) -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Use dai_fmt in imote2 machine driver
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/imote2.c | 16 ++-------------- 1 files changed, 2 insertions(+), 14 deletions(-) diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 97d3aec..dc905ae 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -30,20 +30,6 @@ static int imote2_asoc_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* CPU should be clock master */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, SND_SOC_CLOCK_IN); if (ret < 0) @@ -67,6 +53,8 @@ static struct snd_soc_dai_link imote2_dai = { .codec_dai_name = "wm8940-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm8940-codec.0-0034", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &imote2_asoc_ops, }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Use dai_fmt in hx4700 machine driver
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/hx4700.c | 16 ++-------------- 1 files changed, 2 insertions(+), 14 deletions(-) diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 03ef9f3..8260207 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -65,20 +65,6 @@ static int hx4700_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the I2S system clock as output */ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_OUT); @@ -175,6 +161,8 @@ static struct snd_soc_dai_link hx4700_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "ak4641.0-0012", .init = hx4700_ak4641_init, + .dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &hx4700_ops, }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/5] ASoC: Convert ak4104 to devm_kzalloc()
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4104.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 152420c..d27b5e4 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -261,7 +261,8 @@ static int ak4104_spi_probe(struct spi_device *spi) if (ret < 0) return ret; - ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL); + ak4104 = devm_kzalloc(&spi->dev, sizeof(struct ak4104_private), + GFP_KERNEL); if (ak4104 == NULL) return -ENOMEM; @@ -271,15 +272,12 @@ static int ak4104_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_device_ak4104, &ak4104_dai, 1); - if (ret < 0) - kfree(ak4104); return ret; } static int __devexit ak4104_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } -- 1.7.5.4

1 4

Re: [alsa-devel] [PATCH 5/5] ASoC: mxs: correct 'direction' of device_prep_dma_cyclic
by Mark Brown 20 Dec '11

20 Dec '11

On Wed, Dec 14, 2011 at 03:47:46PM +0800, Shawn Guo wrote: > The commit 49920bc (dmaengine: add new enum dma_transfer_direction) > changes the type of parameter 'direction' of device_prep_dma_cyclic > from dma_data_direction to dma_transfer_direction. Applied, thanks. With patches like this you really need to supply more context - knowing which release the commit you're talking about appears in is useful for exammple, as is having some hint as to what the rest of the series is (does this patch depend on any of the earlier patches, and if not why is it part of a series).

2 1

[alsa-devel] [PATCH v4 1/3] ASoC: Add utility to set a card's name from device tree
by Stephen Warren 20 Dec '11

20 Dec '11

Implement snd_soc_of_parse_card_name(), a utility function that sets a card's name from device tree. The machine driver specifies the DT property to use, since this is binding-specific. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This series is based on my previously posted patch "ASoC: Allow DAI links to be specified using device tree nodes" v4. Patches 1 & 2 at most rely on that for context. Patch 3 relies on it for functionality. v2: New patch implementing new functionality v3: Name this embedded-audio-complex instead of soc-audio-complex. Change property name from user-visible-name to model. Remove compatible property from binding. v4: Convert this to a utility function that cards can call if desired. Modify snd_soc_register_card() not to call this automatically. Allow cards to specify the name of the device tree property that stores that card's name. Don't specify the property as a generic binding; remove embedded-audio-complex.txt include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 25 +++++++++++++++++++++++++ 2 files changed, 28 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 40c256e..6e51a3f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -966,6 +966,9 @@ static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) int snd_soc_util_init(void); void snd_soc_util_exit(void); +int snd_soc_of_parse_card_name(struct snd_soc_card *card, + const char *propname); + #include <sound/soc-dai.h> #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83aa82e..4e8d762 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -32,6 +32,7 @@ #include <linux/platform_device.h> #include <linux/ctype.h> #include <linux/slab.h> +#include <linux/of.h> #include <sound/ac97_codec.h> #include <sound/core.h> #include <sound/jack.h> @@ -3367,6 +3368,30 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); +/* Retrieve a card's name from device tree */ +int snd_soc_of_parse_card_name(struct snd_soc_card *card, + const char *propname) +{ + struct device_node *np = card->dev->of_node; + int ret; + + ret = of_property_read_string_index(np, propname, 0, &card->name); + /* + * EINVAL means the property does not exist. This is fine providing + * card->name was previously set, which is checked later in + * snd_soc_register_card. + */ + if (ret < 0 && ret != -EINVAL) { + dev_err(card->dev, + "Property '%s' could not be read: %d\n", + propname, ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name); + static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS -- 1.7.0.4

2 3

[alsa-devel] [PATCH] ASoC: Rename rt562[1|2]_vol_snd_controls to alc562[1|2]_vol_snd_controls
by Axel Lin 20 Dec '11

20 Dec '11

The module desciption says this is ASoC alc5621/2/3 driver. Make the naming consistent with the reset of the code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index da97f02..6a9b621 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -99,7 +99,7 @@ static const unsigned int boost_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); -static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { +static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", @@ -110,7 +110,7 @@ static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { ALC5623_HP_OUT_VOL, 15, 7, 1, 1), }; -static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = { +static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", @@ -925,12 +925,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) switch (alc5623->id) { case 0x21: - snd_soc_add_controls(codec, rt5621_vol_snd_controls, - ARRAY_SIZE(rt5621_vol_snd_controls)); + snd_soc_add_controls(codec, alc5621_vol_snd_controls, + ARRAY_SIZE(alc5621_vol_snd_controls)); break; case 0x22: - snd_soc_add_controls(codec, rt5622_vol_snd_controls, - ARRAY_SIZE(rt5622_vol_snd_controls)); + snd_soc_add_controls(codec, alc5622_vol_snd_controls, + ARRAY_SIZE(alc5622_vol_snd_controls)); break; case 0x23: snd_soc_add_controls(codec, alc5623_vol_snd_controls, -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Use dai_fmt in edb93xx machine driver
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/ep93xx/edb93xx.c | 14 ++------------ 1 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index 6b90c75..9f6fecd 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -48,18 +48,6 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, else mclk_rate = rate * 64 * 2; - err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS); - if (err) - return err; - - err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS); - if (err) - return err; - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, SND_SOC_CLOCK_IN); if (err) @@ -80,6 +68,8 @@ static struct snd_soc_dai_link edb93xx_dai = { .cpu_dai_name = "ep93xx-i2s", .codec_name = "spi0.0", .codec_dai_name = "cs4271-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &edb93xx_ops, }; -- 1.7.5.4

4 4

[alsa-devel] [PATCH] ASoC: Remove export of s3c_pcm_dai
by Axel Lin 20 Dec '11

20 Dec '11

We don't need to export s3c_pcm_dai after multi-component patch. Thus remove export of s3c_pcm_dai and make it static. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/pcm.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index beef63f..786811d 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -478,7 +478,7 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .formats = SNDRV_PCM_FMTBIT_S16_LE, \ } -struct snd_soc_dai_driver s3c_pcm_dai[] = { +static struct snd_soc_dai_driver s3c_pcm_dai[] = { [0] = { .name = "samsung-pcm.0", S3C_PCM_DAI_DECLARE, @@ -488,7 +488,6 @@ struct snd_soc_dai_driver s3c_pcm_dai[] = { S3C_PCM_DAI_DECLARE, }, }; -EXPORT_SYMBOL_GPL(s3c_pcm_dai); static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) { -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: dsd1791: Introduce driver for TI DSD1791 stereo codec
by Michael Williamson 20 Dec '11

20 Dec '11

This patch introduces a (spi) codec driver for the Texas Instruments DSD1791 24 bit audio stereo DAC. http://www.ti.com/product/dsd1791 Testing for basic operation using 16 and 24 bit I2S mode has been performed using a MityDSP-L138 SOM and an Industrial I/O board from Critical Link. Signed-off-by: Michael Williamson <michael.williamson(a)criticallink.com> --- This patch incorporates changes from the original RFC as a result of comments received from Mark Brown, Leon Romanovsky, and Lars-Peter Clausen. Thanks. Summary of Changes: - Use devm_kzalloc() - use regmap cached I/O feature of framework - simplify code applying codec fmt configuration - clean up section attributes on local functions - make local functions/variables static where appropriate - remove set_sysclk method - remove unused tlv header file - remove chatter - strip "-codec" from driver name - fixup driver variable name for consistency - Add Left and Right volume control - Add digital mute sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dsd1791.c | 256 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 262 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/dsd1791.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..95b7969 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 + select SND_SOC_DSD1791 if SPI_MASTER select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C @@ -205,6 +206,9 @@ config SND_SOC_DA7210 config SND_SOC_DFBMCS320 tristate +config SND_SOC_DSD1791 + tristate + config SND_SOC_DMIC tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..d6b5f6a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -21,6 +21,7 @@ snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o +snd-soc-dsd1791-objs := dsd1791.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o @@ -120,6 +121,7 @@ obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_DSD1791) += snd-soc-dsd1791.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dsd1791.c b/sound/soc/codecs/dsd1791.c new file mode 100644 index 0000000..41bbc61 --- /dev/null +++ b/sound/soc/codecs/dsd1791.c @@ -0,0 +1,256 @@ +/* + * ALSA SoC codec driver for Texas Instruments DSD1791. + * + * Author: (C) Michael Williamson <michael.williamson(a)criticallink.com> + * Copyright: (C) 2011 Critical Link, LLC + * + * 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. + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/spi/spi.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#define DSD1791_REG_DIGATT_L 16 +#define DSD1791_REG_DIGATT_R 17 +#define DSD1791_REG_AUDFMT 18 +#define DSD1791_REG_SRST 20 + +#define DSD1791_FMT_16RJ (0<<4) +#define DSD1791_FMT_20RJ (1<<4) +#define DSD1791_FMT_24RJ (2<<4) +#define DSD1791_FMT_24LJ (3<<4) +#define DSD1791_FMT_16I2S (4<<4) +#define DSD1791_FMT_24I2S (5<<4) +#define DSD1791_FMT_MASK 0x70 + +/* DSD1791 register cache (16 through 23 are used) */ +static const u8 dsd1791_reg[] = { + [16] = 0xFF, + [17] = 0xFF, + [18] = 0x50, + [19] = 0x00, + [20] = 0x00, + [21] = 0x01, + [22] = 0x00, + [23] = 0x00, +}; + +struct dsd1791 { + struct spi_device *spi; + struct snd_soc_codec codec; + int dai_fmt; + int pcm_fmt; +}; + +static int dsd1791_set_format_word(struct dsd1791 *dsd1791, + struct snd_soc_codec *codec) +{ + u8 fmt = 0; + u8 reg; + + switch (dsd1791->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + + case SND_SOC_DAIFMT_I2S: + switch (dsd1791->pcm_fmt) { + case SNDRV_PCM_FORMAT_S16_LE: + fmt = DSD1791_FMT_16I2S; + break; + case SNDRV_PCM_FORMAT_S24_LE: + fmt = DSD1791_FMT_24I2S; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_RIGHT_J: + switch (dsd1791->pcm_fmt) { + case SNDRV_PCM_FORMAT_S16_LE: + fmt = DSD1791_FMT_16RJ; + break; + case SNDRV_PCM_FORMAT_S24_LE: + fmt = DSD1791_FMT_24RJ; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_LEFT_J: + switch (dsd1791->pcm_fmt) { + case SNDRV_PCM_FORMAT_S24_LE: + fmt = DSD1791_FMT_24LJ; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + reg = snd_soc_read(codec, DSD1791_REG_AUDFMT); + reg &= ~(DSD1791_FMT_MASK); + reg |= fmt; + return snd_soc_write(codec, DSD1791_REG_AUDFMT, reg); +} + +static int dsd1791_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg; + + reg = snd_soc_read(codec, DSD1791_REG_AUDFMT); + if (mute) + reg |= 1; + else + reg &= ~1; + return snd_soc_write(codec, DSD1791_REG_AUDFMT, reg); +} + +static int dsd1791_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct dsd1791 *dsd1791 = snd_soc_codec_get_drvdata(codec); + + dsd1791->pcm_fmt = params_format(params); + + return dsd1791_set_format_word(dsd1791, codec); +} + +static int dsd1791_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct dsd1791 *dsd1791 = snd_soc_codec_get_drvdata(codec); + + dsd1791->dai_fmt = fmt; + + return dsd1791_set_format_word(dsd1791, codec); +} + +#define DSD1791_RATES SNDRV_PCM_RATE_8000_192000 +#define DSD1791_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops dsd1791_dai_ops = { + .hw_params = dsd1791_hw_params, + .set_fmt = dsd1791_set_fmt, + .digital_mute = dsd1791_mute, +}; + +static struct snd_soc_dai_driver dsd1791_dai = { + .name = "dsd1791", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = DSD1791_RATES, + .formats = DSD1791_FORMATS, + }, + .ops = &dsd1791_dai_ops, +}; + +static const struct snd_kcontrol_new dsd1791_snd_controls[] = { + SOC_SINGLE("Left Playback Volume", DSD1791_REG_DIGATT_L, 0, 255, 0), + SOC_SINGLE("Right Playback Volume", DSD1791_REG_DIGATT_R, 0, 255, 0), +}; + +static int dsd1791_probe(struct snd_soc_codec *codec) +{ + u8 reg; + int ret; + struct dsd1791 *dsd1791 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_SPI); + if (ret) { + dev_err(codec->dev, "Failed to set Cache I/O: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, DSD1791_REG_SRST, 0x40); + if (ret) { + dev_err(codec->dev, "Unable to reset device: %d\n", ret); + goto err; + } + + /* default format after reset */ + dsd1791->dai_fmt = SND_SOC_DAIFMT_I2S; + dsd1791->pcm_fmt = SNDRV_PCM_FORMAT_S24_LE; + + /* enable attenuation control */ + reg = snd_soc_read(codec, DSD1791_REG_AUDFMT); + reg |= 0x80; + snd_soc_write(codec, DSD1791_REG_AUDFMT, reg); + + snd_soc_add_controls(codec, dsd1791_snd_controls, + ARRAY_SIZE(dsd1791_snd_controls)); + return 0; +err: + return ret; +} + +struct snd_soc_codec_driver dsd1791_codec_driver = { + .probe = dsd1791_probe, + .reg_cache_size = ARRAY_SIZE(dsd1791_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = dsd1791_reg, +}; + +static int __devinit dsd1791_spi_probe(struct spi_device *spi) +{ + struct dsd1791 *dsd1791; + + dsd1791 = devm_kzalloc(&spi->dev, sizeof *dsd1791, GFP_KERNEL); + if (!dsd1791) + return -ENOMEM; + + spi_set_drvdata(spi, dsd1791); + + return snd_soc_register_codec(&spi->dev, + &dsd1791_codec_driver, &dsd1791_dai, 1); +}; + +static int __devexit dsd1791_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver dsd1791_spi_driver = { + .driver = { + .name = "dsd1791", + .owner = THIS_MODULE, + }, + .probe = dsd1791_spi_probe, + .remove = __devexit_p(dsd1791_spi_remove), +}; + +static int __init dsd1791_init(void) +{ + return spi_register_driver(&dsd1791_spi_driver); +} +module_init(dsd1791_init); + +static void __exit dsd1791_exit(void) +{ + spi_unregister_driver(&dsd1791_spi_driver); +} +module_exit(dsd1791_exit); + +MODULE_DESCRIPTION("ASoC DSD1791 codec driver"); +MODULE_AUTHOR("Michael Williamson"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

3 4

[alsa-devel] [PATCH] ASoC: Convert uda1380 to table based DAPM and control init
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/uda1380.c | 24 ++++++++---------------- 1 files changed, 8 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 83e45d2..8f734d6 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -373,7 +373,7 @@ static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route uda1380_dapm_routes[] = { /* output mux */ {"HeadPhone Driver", NULL, "Output Mux"}, @@ -410,17 +410,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Right PGA", NULL, "VINR"}, }; -static int uda1380_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -764,10 +753,6 @@ static int uda1380_probe(struct snd_soc_codec *codec) break; } - snd_soc_add_controls(codec, uda1380_snd_controls, - ARRAY_SIZE(uda1380_snd_controls)); - uda1380_add_widgets(codec); - return 0; err_free_gpio: @@ -802,6 +787,13 @@ static struct snd_soc_codec_driver soc_codec_dev_uda1380 = { .reg_word_size = sizeof(u16), .reg_cache_default = uda1380_reg, .reg_cache_step = 1, + + .controls = uda1380_snd_controls, + .num_controls = ARRAY_SIZE(uda1380_snd_controls), + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = uda1380_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda1380_dapm_routes), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Convert max9850 to table based DAPM and control init
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/max9850.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 7dfd6e8..47060d2 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -86,7 +86,7 @@ SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route max9850_dapm_routes[] = { /* output mixer */ {"Output Mixer", NULL, "DAC"}, {"Output Mixer", "Line In Switch", "Line Input"}, @@ -293,7 +293,6 @@ static int max9850_resume(struct snd_soc_codec *codec) static int max9850_probe(struct snd_soc_codec *codec) { - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); @@ -309,13 +308,6 @@ static int max9850_probe(struct snd_soc_codec *codec) /* set slew-rate 125ms */ snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0); - snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets, - ARRAY_SIZE(max9850_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - snd_soc_add_controls(codec, max9850_controls, - ARRAY_SIZE(max9850_controls)); - return 0; } @@ -328,6 +320,13 @@ static struct snd_soc_codec_driver soc_codec_dev_max9850 = { .reg_word_size = sizeof(u8), .reg_cache_default = max9850_reg, .volatile_register = max9850_volatile_register, + + .controls = max9850_controls, + .num_controls = ARRAY_SIZE(max9850_controls), + .dapm_widgets = max9850_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9850_dapm_widgets), + .dapm_routes = max9850_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9850_dapm_routes), }; static int __devinit max9850_i2c_probe(struct i2c_client *i2c, -- 1.7.5.4

2 1

Re: [alsa-devel] [GIT PULL] ARM: amba: Enable module alias autogeneration for AMBA drivers
by Russell King - ARM Linux 19 Dec '11

19 Dec '11

On Mon, Dec 19, 2011 at 01:24:09PM +0000, Dave Martin wrote: > Hi Russell, > > This one isn't urgent, but I'm not seeing the amba modalias patches > anywhere yet. Did you have any outstanding concerns which need to be > resolved? > > If you can suggest when/if these are likely to merge that would be great. > > Since Alessandro is now expecting to have to rebase on top of the amba > additions anyway, we shouldn't need to worry about conflicting with his > patches. > > Of course, if you've already merged these somewhere, then there's no > problem. I have pulled it, I just haven't merged it in anywhere yet.

1 0

Re: [alsa-devel] [LAU] kernel crashes with jackd (with Dell Precision M6600's integrated Intel HDA sound)
by Clemens Ladisch 19 Dec '11

19 Dec '11

Csillag Kristof wrote: > Lately, when experimenting with JACK, I began getting kernel crashes :( > > I am using stock Debian kernel v3.1.5. > > [ 7555.763914] BUG: unable to handle kernel NULL pointer dereference at 0000000000000028 > [ 7555.763952] IP: [<ffffffffa08ca7b8>] setup_bdle.isra.21+0x44/0xd9 [snd_hda_intel] > [ 7555.764735] RAX: 0000000000000000 RBX: ffff880221825e00 RCX: 0000000000000000 > [ 7555.765126] Call Trace: > [ 7555.765170] [<ffffffffa08caa5c>] ? azx_pcm_prepare+0x20f/0x3c7 [snd_hda_intel] > [ 7555.765589] Code: 8b 32 e9 94 00 00 00 81 7d 00 ff 00 00 00 0f 87 96 00 00 00 48 8b 83 c0 00 00 00 44 89 e2 44 89 e6 48 89 df 48 8b 80 d8 02 00 00 > [ 7555.765683] 8b 48 28 48 89 d0 81 e2 ff 0f 00 00 48 c1 e8 0c 48 c1 e0 04 7: 81 7d 00 ff 00 00 00 cmpl $0xff,0x0(%rbp) e: 0f 87 96 00 00 00 ja aa 14: 48 8b 83 c0 00 00 00 mov 0xc0(%rbx),%rax 1b: 44 89 e2 mov %r12d,%edx 1e: 44 89 e6 mov %r12d,%esi 21: 48 89 df mov %rbx,%rdi 24: 48 8b 80 d8 02 00 00 mov 0x2d8(%rax),%rax 2b: ?? 8b 48 28 mov 0x28(%rax),%r?? <-- crash here 2f: 48 89 d0 mov %rdx,%rax 32: 81 e2 ff 0f 00 00 and $0xfff,%edx 38: 48 c1 e8 0c shr $0xc,%rax 3c: 48 c1 e0 04 shl $0x4,%rax substream->runtime->dma_buffer_p is NULL. > https://bugzilla.redhat.com/show_bug.cgi?id=702056 > https://bugzilla.redhat.com/show_bug.cgi?id=718056 > https://bugzilla.redhat.com/show_bug.cgi?id=749186 Same bug. Regards, Clemens

2 1

[alsa-devel] [PATCH] atmel ac97c: using software reset instead hardware reset if not available
by voice 19 Dec '11

19 Dec '11

Using software reset instead hardware reset if not available Signed-off-by: Bo Shen <voice.shen(a)atmel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/atmel/ac97c.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 6e5adde..73516f6 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -899,6 +899,10 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) /* AC97 v2.2 specifications says minimum 1 us. */ udelay(2); gpio_set_value(chip->reset_pin, 1); + } else { + ac97c_writel(chip, MR, AC97C_MR_WRST | AC97C_MR_ENA); + udelay(2); + ac97c_writel(chip, MR, AC97C_MR_ENA); } } -- 1.6.3.3

2 1

[alsa-devel] FW: [PATCH] atmel ac97c: using software reset instead hardware reset if not available
by Shen, Voice 19 Dec '11

19 Dec '11

Hi Sir, Add patch to Atmel AC97C. Object: using software reset instead hardware reset if not available Best Regards, Voice Shen -----Original Message----- From: Nicolas Ferre [mailto:nicolas.ferre@atmel.com] Sent: Thursday, November 24, 2011 18:44 voice To: Shen, Voice Cc: linux-kernel(a)vger.kernel.org; linux-sound(a)vger.kernel.org; Lin, JM; Xu, Hong Subject: Re: [PATCH] atmel ac97c: using software reset instead hardware reset if not available On 11/23/2011 11:05 AM, voice : > Using software reset instead hardware reset if not available > > Signed-off-by: Bo Shen <voice.shen(a)atmel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> You may need to sent it to the sound subsystem mailing-list and maintainers. Anyway, I let you follow this patch... As it is a fixup and will benefit to existing platforms, I think that it can make its way to mainline during 3.2 phase. Best regards, > --- > sound/atmel/ac97c.c | 4 ++++ > 1 files changed, 4 insertions(+), 0 deletions(-) > > diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c > index 6e5adde..73516f6 100644 > --- a/sound/atmel/ac97c.c > +++ b/sound/atmel/ac97c.c > @@ -899,6 +899,10 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) > /* AC97 v2.2 specifications says minimum 1 us. */ > udelay(2); > gpio_set_value(chip->reset_pin, 1); > + } else { > + ac97c_writel(chip, MR, AC97C_MR_WRST | AC97C_MR_ENA); > + udelay(2); > + ac97c_writel(chip, MR, AC97C_MR_ENA); > } > } > -- Nicolas Ferre

2 1

Re: [alsa-devel] [PATCH 13/15] module_param: make bool parameters really bool (sound)
by Takashi Iwai 19 Dec '11

19 Dec '11

At Thu, 15 Dec 2011 13:49:36 +1030, Rusty Russell wrote: > > module_param(bool) used to counter-intuitively take an int. In > fddd5201 (mid-2009) we allowed bool or int/unsigned int using a messy > trick. > > It's time to remove the int/unsigned int option. For this version > it'll simply give a warning, but it'll break next kernel version. > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Rusty Russell <rusty(a)rustcorp.com.au> Acked-by: Takashi Iwai <tiwai(a)suse.de> (Maybe it'd be good to change with true and false instead of 1 and 0, too. But it's subtle things and we don't have to mix up in a single patch.) Rusty, would you like to merge from your tree, or shall I merge through sound tree? thanks, Takashi > --- > Documentation/DocBook/writing-an-alsa-driver.tmpl | 2 +- > sound/core/oss/pcm_oss.c | 2 +- > sound/core/seq/seq_dummy.c | 2 +- > sound/drivers/aloop.c | 2 +- > sound/drivers/dummy.c | 6 +++--- > sound/drivers/ml403-ac97cr.c | 2 +- > sound/drivers/mpu401/mpu401.c | 6 +++--- > sound/drivers/mts64.c | 2 +- > sound/drivers/opl3/opl3_midi.c | 2 +- > sound/drivers/opl3/opl3_seq.c | 2 +- > sound/drivers/pcsp/pcsp.c | 4 ++-- > sound/drivers/pcsp/pcsp_lib.c | 2 +- > sound/drivers/portman2x4.c | 2 +- > sound/drivers/serial-u16550.c | 4 ++-- > sound/drivers/virmidi.c | 2 +- > sound/isa/ad1816a/ad1816a.c | 2 +- > sound/isa/ad1848/ad1848.c | 4 ++-- > sound/isa/adlib.c | 2 +- > sound/isa/als100.c | 2 +- > sound/isa/azt2320.c | 2 +- > sound/isa/cmi8330.c | 4 ++-- > sound/isa/cs423x/cs4231.c | 2 +- > sound/isa/cs423x/cs4236.c | 4 ++-- > sound/isa/es1688/es1688.c | 4 ++-- > sound/isa/es18xx.c | 4 ++-- > sound/isa/galaxy/galaxy.c | 2 +- > sound/isa/gus/gusclassic.c | 2 +- > sound/isa/gus/gusextreme.c | 2 +- > sound/isa/gus/gusmax.c | 2 +- > sound/isa/gus/interwave.c | 4 ++-- > sound/isa/msnd/msnd_pinnacle.c | 2 +- > sound/isa/opl3sa2.c | 4 ++-- > sound/isa/opti9xx/miro.c | 2 +- > sound/isa/opti9xx/opti92x-ad1848.c | 2 +- > sound/isa/sb/jazz16.c | 2 +- > sound/isa/sb/sb16.c | 4 ++-- > sound/isa/sb/sb8.c | 2 +- > sound/isa/sc6000.c | 2 +- > sound/isa/wavefront/wavefront.c | 6 +++--- > sound/oss/ad1848.c | 8 ++++---- > sound/oss/msnd_pinnacle.c | 2 +- > sound/oss/pas2_card.c | 12 ++++++------ > sound/oss/pss.c | 10 +++++----- > sound/oss/trix.c | 2 +- > sound/pci/ac97/ac97_codec.c | 2 +- > sound/pci/ad1889.c | 2 +- > sound/pci/ali5451/ali5451.c | 4 ++-- > sound/pci/als4000.c | 2 +- > sound/pci/asihpi/asihpi.c | 4 ++-- > sound/pci/atiixp.c | 4 ++-- > sound/pci/atiixp_modem.c | 2 +- > sound/pci/au88x0/au88x0.c | 2 +- > sound/pci/aw2/aw2-alsa.c | 2 +- > sound/pci/azt3328.c | 2 +- > sound/pci/bt87x.c | 4 ++-- > sound/pci/ca0106/ca0106_main.c | 2 +- > sound/pci/cmipci.c | 4 ++-- > sound/pci/cs4281.c | 4 ++-- > sound/pci/cs46xx/cs46xx.c | 8 ++++---- > sound/pci/cs5535audio/cs5535audio.c | 2 +- > sound/pci/ctxfi/cttimer.c | 4 ++-- > sound/pci/ctxfi/xfi.c | 2 +- > sound/pci/echoaudio/echoaudio.c | 2 +- > sound/pci/emu10k1/emu10k1.c | 4 ++-- > sound/pci/emu10k1/emu10k1x.c | 2 +- > sound/pci/ens1370.c | 4 ++-- > sound/pci/es1938.c | 2 +- > sound/pci/es1968.c | 4 ++-- > sound/pci/fm801.c | 2 +- > sound/pci/hda/hda_intel.c | 8 ++++---- > sound/pci/ice1712/ice1712.c | 4 ++-- > sound/pci/ice1712/ice1724.c | 2 +- > sound/pci/intel8x0.c | 6 +++--- > sound/pci/intel8x0m.c | 2 +- > sound/pci/korg1212/korg1212.c | 2 +- > sound/pci/lola/lola.c | 2 +- > sound/pci/lx6464es/lx6464es.c | 2 +- > sound/pci/maestro3.c | 4 ++-- > sound/pci/mixart/mixart.c | 2 +- > sound/pci/nm256/nm256.c | 12 ++++++------ > sound/pci/oxygen/oxygen.c | 2 +- > sound/pci/oxygen/virtuoso.c | 2 +- > sound/pci/pcxhr/pcxhr.c | 4 ++-- > sound/pci/riptide/riptide.c | 2 +- > sound/pci/rme32.c | 4 ++-- > sound/pci/rme96.c | 2 +- > sound/pci/rme9652/hdsp.c | 2 +- > sound/pci/rme9652/hdspm.c | 2 +- > sound/pci/rme9652/rme9652.c | 4 ++-- > sound/pci/sis7019.c | 2 +- > sound/pci/sonicvibes.c | 6 +++--- > sound/pci/trident/trident.c | 2 +- > sound/pci/via82xx.c | 4 ++-- > sound/pci/via82xx_modem.c | 2 +- > sound/pci/vx222/vx222.c | 4 ++-- > sound/pci/ymfpci/ymfpci.c | 4 ++-- > sound/pcmcia/pdaudiocf/pdaudiocf.c | 2 +- > sound/pcmcia/vx/vxpocket.c | 2 +- > sound/ppc/powermac.c | 2 +- > sound/sh/aica.c | 2 +- > sound/sparc/amd7930.c | 2 +- > sound/sparc/cs4231.c | 2 +- > sound/sparc/dbri.c | 2 +- > sound/usb/6fire/chip.c | 2 +- > sound/usb/caiaq/device.c | 2 +- > sound/usb/card.c | 6 +++--- > sound/usb/misc/ua101.c | 2 +- > sound/usb/usx2y/us122l.c | 2 +- > sound/usb/usx2y/usbusx2y.c | 2 +- > 109 files changed, 172 insertions(+), 172 deletions(-) > > diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl > --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl > +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl > @@ -404,7 +404,7 @@ > /* SNDRV_CARDS: maximum number of cards supported by this module */ > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > - static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > + static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > /* definition of the chip-specific record */ > struct mychip { > diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c > --- a/sound/core/oss/pcm_oss.c > +++ b/sound/core/oss/pcm_oss.c > @@ -47,7 +47,7 @@ > > static int dsp_map[SNDRV_CARDS]; > static int adsp_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; > -static int nonblock_open = 1; > +static bool nonblock_open = 1; > > MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>, Abramo Bagnara <abramo(a)alsa-project.org>"); > MODULE_DESCRIPTION("PCM OSS emulation for ALSA."); > diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c > --- a/sound/core/seq/seq_dummy.c > +++ b/sound/core/seq/seq_dummy.c > @@ -65,7 +65,7 @@ MODULE_LICENSE("GPL"); > MODULE_ALIAS("snd-seq-client-" __stringify(SNDRV_SEQ_CLIENT_DUMMY)); > > static int ports = 1; > -static int duplex; > +static bool duplex; > > module_param(ports, int, 0444); > MODULE_PARM_DESC(ports, "number of ports to be created"); > diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c > --- a/sound/drivers/aloop.c > +++ b/sound/drivers/aloop.c > @@ -51,7 +51,7 @@ MODULE_SUPPORTED_DEVICE("{{ALSA,Loopback > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; > +static bool enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; > static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; > static int pcm_notify[SNDRV_CARDS]; > > diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c > --- a/sound/drivers/dummy.c > +++ b/sound/drivers/dummy.c > @@ -60,15 +60,15 @@ MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy so > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; > +static bool enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; > static char *model[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = NULL}; > static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; > //static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; > #ifdef CONFIG_HIGH_RES_TIMERS > -static int hrtimer = 1; > +static bool hrtimer = 1; > #endif > -static int fake_buffer = 1; > +static bool fake_buffer = 1; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for dummy soundcard."); > diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c > --- a/sound/drivers/ml403-ac97cr.c > +++ b/sound/drivers/ml403-ac97cr.c > @@ -73,7 +73,7 @@ MODULE_SUPPORTED_DEVICE("{{Xilinx,ML403 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for ML403 AC97 Controller Reference."); > diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c > --- a/sound/drivers/mpu401/mpu401.c > +++ b/sound/drivers/mpu401/mpu401.c > @@ -35,13 +35,13 @@ MODULE_LICENSE("GPL"); > > static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* exclude the first card */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > #ifdef CONFIG_PNP > -static int pnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool pnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > #endif > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* MPU-401 port number */ > static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* MPU-401 IRQ */ > -static int uart_enter[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool uart_enter[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for MPU-401 device."); > diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c > --- a/sound/drivers/mts64.c > +++ b/sound/drivers/mts64.c > @@ -36,7 +36,7 @@ > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > static struct platform_device *platform_devices[SNDRV_CARDS]; > static int device_count; > diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c > --- a/sound/drivers/opl3/opl3_midi.c > +++ b/sound/drivers/opl3/opl3_midi.c > @@ -27,7 +27,7 @@ > > extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; > > -extern int use_internal_drums; > +extern bool use_internal_drums; > > static void snd_opl3_note_off_unsafe(void *p, int note, int vel, > struct snd_midi_channel *chan); > diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c > --- a/sound/drivers/opl3/opl3_seq.c > +++ b/sound/drivers/opl3/opl3_seq.c > @@ -32,7 +32,7 @@ MODULE_AUTHOR("Uros Bizjak <uros@kss-lok > MODULE_LICENSE("GPL"); > MODULE_DESCRIPTION("ALSA driver for OPL3 FM synth"); > > -int use_internal_drums = 0; > +bool use_internal_drums = 0; > module_param(use_internal_drums, bool, 0444); > MODULE_PARM_DESC(use_internal_drums, "Enable internal OPL2/3 drums."); > > diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c > --- a/sound/drivers/pcsp/pcsp.c > +++ b/sound/drivers/pcsp/pcsp.c > @@ -25,8 +25,8 @@ MODULE_ALIAS("platform:pcspkr"); > > static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > -static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ > -static int nopcm; /* Disable PCM capability of the driver */ > +static bool enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ > +static bool nopcm; /* Disable PCM capability of the driver */ > > module_param(index, int, 0444); > MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); > diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c > --- a/sound/drivers/pcsp/pcsp_lib.c > +++ b/sound/drivers/pcsp/pcsp_lib.c > @@ -14,7 +14,7 @@ > #include <asm/io.h> > #include "pcsp.h" > > -static int nforce_wa; > +static bool nforce_wa; > module_param(nforce_wa, bool, 0444); > MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround " > "(expect bad sound)"); > diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c > --- a/sound/drivers/portman2x4.c > +++ b/sound/drivers/portman2x4.c > @@ -55,7 +55,7 @@ > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > static struct platform_device *platform_devices[SNDRV_CARDS]; > static int device_count; > diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c > --- a/sound/drivers/serial-u16550.c > +++ b/sound/drivers/serial-u16550.c > @@ -69,7 +69,7 @@ static char *adaptor_names[] = { > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x3f8,0x2f8,0x3e8,0x2e8 */ > static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 3,4,5,7,9,10,11,14,15 */ > static int speed[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 38400}; /* 9600,19200,38400,57600,115200 */ > @@ -77,7 +77,7 @@ static int base[SNDRV_CARDS] = {[0 ... ( > static int outs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; /* 1 to 16 */ > static int ins[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; /* 1 to 16 */ > static int adaptor[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = SNDRV_SERIAL_SOUNDCANVAS}; > -static int droponfull[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS -1)] = SNDRV_SERIAL_NORMALBUFF }; > +static bool droponfull[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS -1)] = SNDRV_SERIAL_NORMALBUFF }; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Serial MIDI."); > diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c > --- a/sound/drivers/virmidi.c > +++ b/sound/drivers/virmidi.c > @@ -63,7 +63,7 @@ MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; > +static bool enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; > static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4}; > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c > --- a/sound/isa/ad1816a/ad1816a.c > +++ b/sound/isa/ad1816a/ad1816a.c > @@ -44,7 +44,7 @@ MODULE_SUPPORTED_DEVICE("{{Highscreen,So > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 1-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c > --- a/sound/isa/ad1848/ad1848.c > +++ b/sound/isa/ad1848/ad1848.c > @@ -43,11 +43,11 @@ MODULE_SUPPORTED_DEVICE("{{Analog Device > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,11,12,15 */ > static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */ > -static int thinkpad[SNDRV_CARDS]; /* Thinkpad special case */ > +static bool thinkpad[SNDRV_CARDS]; /* Thinkpad special case */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); > diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c > --- a/sound/isa/adlib.c > +++ b/sound/isa/adlib.c > @@ -18,7 +18,7 @@ MODULE_LICENSE("GPL"); > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/isa/als100.c b/sound/isa/als100.c > --- a/sound/isa/als100.c > +++ b/sound/isa/als100.c > @@ -54,7 +54,7 @@ MODULE_LICENSE("GPL"); > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c > --- a/sound/isa/azt2320.c > +++ b/sound/isa/azt2320.c > @@ -55,7 +55,7 @@ MODULE_SUPPORTED_DEVICE("{{Aztech System > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c > --- a/sound/isa/cmi8330.c > +++ b/sound/isa/cmi8330.c > @@ -69,9 +69,9 @@ MODULE_SUPPORTED_DEVICE("{{C-Media,CMI83 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; > #ifdef CONFIG_PNP > -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > #endif > static long sbport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; > static int sbirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; > diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c > --- a/sound/isa/cs423x/cs4231.c > +++ b/sound/isa/cs423x/cs4231.c > @@ -41,7 +41,7 @@ MODULE_SUPPORTED_DEVICE("{{Crystal Semic > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,11,12,15 */ > diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c > --- a/sound/isa/cs423x/cs4236.c > +++ b/sound/isa/cs423x/cs4236.c > @@ -74,9 +74,9 @@ MODULE_ALIAS("snd_cs4232"); > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > #ifdef CONFIG_PNP > -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > #endif > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static long cport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c > --- a/sound/isa/es1688/es1688.c > +++ b/sound/isa/es1688/es1688.c > @@ -51,9 +51,9 @@ MODULE_ALIAS("snd_es968"); > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > #ifdef CONFIG_PNP > -static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; > +static bool isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; > #endif > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ > static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* Usually 0x388 */ > static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1}; > diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c > --- a/sound/isa/es18xx.c > +++ b/sound/isa/es18xx.c > @@ -1964,9 +1964,9 @@ MODULE_SUPPORTED_DEVICE("{{ESS,ES1868 Pn > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > #ifdef CONFIG_PNP > -static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; > +static bool isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; > #endif > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260,0x280 */ > #ifndef CONFIG_PNP > diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c > --- a/sound/isa/galaxy/galaxy.c > +++ b/sound/isa/galaxy/galaxy.c > @@ -35,7 +35,7 @@ MODULE_LICENSE("GPL"); > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); > diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c > --- a/sound/isa/gus/gusclassic.c > +++ b/sound/isa/gus/gusclassic.c > @@ -42,7 +42,7 @@ MODULE_SUPPORTED_DEVICE("{{Gravis,UltraS > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x230,0x240,0x250,0x260 */ > static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 3,5,9,11,12,15 */ > static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */ > diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c > --- a/sound/isa/gus/gusextreme.c > +++ b/sound/isa/gus/gusextreme.c > @@ -46,7 +46,7 @@ MODULE_SUPPORTED_DEVICE("{{Gravis,UltraS > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ > static long gf1_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS) - 1] = -1}; /* 0x210,0x220,0x230,0x240,0x250,0x260,0x270 */ > static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS) - 1] = -1}; /* 0x300,0x310,0x320 */ > diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c > --- a/sound/isa/gus/gusmax.c > +++ b/sound/isa/gus/gusmax.c > @@ -40,7 +40,7 @@ MODULE_SUPPORTED_DEVICE("{{Gravis,UltraS > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x230,0x240,0x250,0x260 */ > static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 2,3,5,9,11,12,15 */ > static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */ > diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c > --- a/sound/isa/gus/interwave.c > +++ b/sound/isa/gus/interwave.c > @@ -55,9 +55,9 @@ MODULE_SUPPORTED_DEVICE("{{AMD,InterWave > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > #ifdef CONFIG_PNP > -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > #endif > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x210,0x220,0x230,0x240,0x250,0x260 */ > #ifdef SNDRV_STB > diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c > --- a/sound/isa/msnd/msnd_pinnacle.c > +++ b/sound/isa/msnd/msnd_pinnacle.c > @@ -785,7 +785,7 @@ static int write_ndelay[SNDRV_CARDS] = { > static int calibrate_signal; > > #ifdef CONFIG_PNP > -static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > module_param_array(isapnp, bool, NULL, 0444); > MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); > #define has_isapnp(x) isapnp[x] > diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c > --- a/sound/isa/opl3sa2.c > +++ b/sound/isa/opl3sa2.c > @@ -46,9 +46,9 @@ MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF719 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > #ifdef CONFIG_PNP > -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > #endif > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0xf86,0x370,0x100 */ > static long sb_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ > diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c > --- a/sound/isa/opti9xx/miro.c > +++ b/sound/isa/opti9xx/miro.c > @@ -61,7 +61,7 @@ static int dma2 = SNDRV_DEFAULT_DMA1; / > static int wss; > static int ide; > #ifdef CONFIG_PNP > -static int isapnp = 1; /* Enable ISA PnP detection */ > +static bool isapnp = 1; /* Enable ISA PnP detection */ > #endif > > module_param(index, int, 0444); > diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c > --- a/sound/isa/opti9xx/opti92x-ad1848.c > +++ b/sound/isa/opti9xx/opti92x-ad1848.c > @@ -63,7 +63,7 @@ MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 ( > > static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > -//static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ > +//static bool enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ > #ifdef CONFIG_PNP > static int isapnp = 1; /* Enable ISA PnP detection */ > #endif > diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c > --- a/sound/isa/sb/jazz16.c > +++ b/sound/isa/sb/jazz16.c > @@ -36,7 +36,7 @@ MODULE_LICENSE("GPL"); > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static unsigned long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; > static unsigned long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; > static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; > diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c > --- a/sound/isa/sb/sb16.c > +++ b/sound/isa/sb/sb16.c > @@ -68,9 +68,9 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ > #ifdef CONFIG_PNP > -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > #endif > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260,0x280 */ > static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x330,0x300 */ > diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c > --- a/sound/isa/sb/sb8.c > +++ b/sound/isa/sb/sb8.c > @@ -36,7 +36,7 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ > static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */ > static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3 */ > diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c > --- a/sound/isa/sc6000.c > +++ b/sound/isa/sc6000.c > @@ -48,7 +48,7 @@ MODULE_SUPPORTED_DEVICE("{{Gallant, SC-6 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220, 0x240 */ > static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 11 */ > static long mss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530, 0xe80 */ > diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c > --- a/sound/isa/wavefront/wavefront.c > +++ b/sound/isa/wavefront/wavefront.c > @@ -38,9 +38,9 @@ MODULE_SUPPORTED_DEVICE("{{Turtle Beach, > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > #ifdef CONFIG_PNP > -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > #endif > static long cs4232_pcm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static int cs4232_pcm_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,11,12,15 */ > @@ -51,7 +51,7 @@ static int ics2115_irq[SNDRV_CARDS] = SN > static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ > static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */ > static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */ > -static int use_cs4232_midi[SNDRV_CARDS]; > +static bool use_cs4232_midi[SNDRV_CARDS]; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for WaveFront soundcard."); > diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c > --- a/sound/oss/ad1848.c > +++ b/sound/oss/ad1848.c > @@ -119,9 +119,9 @@ ad1848_port_info; > static struct address_info cfg; > static int nr_ad1848_devs; > > -static int deskpro_xl; > -static int deskpro_m; > -static int soundpro; > +static bool deskpro_xl; > +static bool deskpro_m; > +static bool soundpro; > > static volatile signed char irq2dev[17] = { > -1, -1, -1, -1, -1, -1, -1, -1, > @@ -177,7 +177,7 @@ static struct { > #ifdef CONFIG_PNP > static int isapnp = 1; > static int isapnpjump; > -static int reverse; > +static bool reverse; > > static int audio_activated; > #else > diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c > --- a/sound/oss/msnd_pinnacle.c > +++ b/sound/oss/msnd_pinnacle.c > @@ -1701,7 +1701,7 @@ static int joystick_io __initdata = CONF > #ifndef CONFIG_MSNDPIN_DIGITAL > # define CONFIG_MSNDPIN_DIGITAL 0 > #endif > -static int digital __initdata = CONFIG_MSNDPIN_DIGITAL; > +static bool digital __initdata = CONFIG_MSNDPIN_DIGITAL; > > #endif /* MSND_CLASSIC */ > > diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c > --- a/sound/oss/pas2_card.c > +++ b/sound/oss/pas2_card.c > @@ -41,19 +41,19 @@ static int pas_irq; > static int pas_sb_base; > DEFINE_SPINLOCK(pas_lock); > #ifndef CONFIG_PAS_JOYSTICK > -static int joystick; > +static bool joystick; > #else > -static int joystick = 1; > +static bool joystick = 1; > #endif > #ifdef SYMPHONY_PAS > -static int symphony = 1; > +static bool symphony = 1; > #else > -static int symphony; > +static bool symphony; > #endif > #ifdef BROKEN_BUS_CLOCK > -static int broken_bus_clock = 1; > +static bool broken_bus_clock = 1; > #else > -static int broken_bus_clock; > +static bool broken_bus_clock; > #endif > > static struct address_info cfg; > diff --git a/sound/oss/pss.c b/sound/oss/pss.c > --- a/sound/oss/pss.c > +++ b/sound/oss/pss.c > @@ -117,9 +117,9 @@ > > /* If compiled into kernel, it enable or disable pss mixer */ > #ifdef CONFIG_PSS_MIXER > -static int pss_mixer = 1; > +static bool pss_mixer = 1; > #else > -static int pss_mixer; > +static bool pss_mixer; > #endif > > > @@ -147,7 +147,7 @@ static DEFINE_SPINLOCK(lock); > static int pss_initialized; > static int nonstandard_microcode; > static int pss_cdrom_port = -1; /* Parameter for the PSS cdrom port */ > -static int pss_enable_joystick; /* Parameter for enabling the joystick */ > +static bool pss_enable_joystick; /* Parameter for enabling the joystick */ > static coproc_operations pss_coproc_operations; > > static void pss_write(pss_confdata *devc, int data) > @@ -1133,8 +1133,8 @@ static int mss_irq __initdata = -1; > static int mss_dma __initdata = -1; > static int mpu_io __initdata = -1; > static int mpu_irq __initdata = -1; > -static int pss_no_sound = 0; /* Just configure non-sound components */ > -static int pss_keep_settings = 1; /* Keep hardware settings at module exit */ > +static bool pss_no_sound = 0; /* Just configure non-sound components */ > +static bool pss_keep_settings = 1; /* Keep hardware settings at module exit */ > static char *pss_firmware = "/etc/sound/pss_synth"; > > module_param(pss_io, int, 0); > diff --git a/sound/oss/trix.c b/sound/oss/trix.c > --- a/sound/oss/trix.c > +++ b/sound/oss/trix.c > @@ -31,7 +31,7 @@ > > static int mpu; > > -static int joystick; > +static bool joystick; > > static unsigned char trix_read(int addr) > { > diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c > --- a/sound/pci/ac97/ac97_codec.c > +++ b/sound/pci/ac97/ac97_codec.c > @@ -42,7 +42,7 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@pe > MODULE_DESCRIPTION("Universal interface for Audio Codec '97"); > MODULE_LICENSE("GPL"); > > -static int enable_loopback; > +static bool enable_loopback; > > module_param(enable_loopback, bool, 0444); > MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control"); > diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c > --- a/sound/pci/ad1889.c > +++ b/sound/pci/ad1889.c > @@ -66,7 +66,7 @@ static char *id[SNDRV_CARDS] = SNDRV_DEF > module_param_array(id, charp, NULL, 0444); > MODULE_PARM_DESC(id, "ID string for the AD1889 soundcard."); > > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > module_param_array(enable, bool, NULL, 0444); > MODULE_PARM_DESC(enable, "Enable AD1889 soundcard."); > > diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c > --- a/sound/pci/ali5451/ali5451.c > +++ b/sound/pci/ali5451/ali5451.c > @@ -48,7 +48,7 @@ MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci > static int index = SNDRV_DEFAULT_IDX1; /* Index */ > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > static int pcm_channels = 32; > -static int spdif; > +static bool spdif; > > module_param(index, int, 0444); > MODULE_PARM_DESC(index, "Index value for ALI M5451 PCI Audio."); > @@ -60,7 +60,7 @@ module_param(spdif, bool, 0444); > MODULE_PARM_DESC(spdif, "Support SPDIF I/O"); > > /* just for backward compatibility */ > -static int enable; > +static bool enable; > module_param(enable, bool, 0444); > > > diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c > --- a/sound/pci/als4000.c > +++ b/sound/pci/als4000.c > @@ -90,7 +90,7 @@ MODULE_SUPPORTED_DEVICE("{{Avance Logic, > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > #ifdef SUPPORT_JOYSTICK > static int joystick_port[SNDRV_CARDS]; > #endif > diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c > --- a/sound/pci/asihpi/asihpi.c > +++ b/sound/pci/asihpi/asihpi.c > @@ -63,8 +63,8 @@ MODULE_DESCRIPTION("AudioScience ALSA AS > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > -static int enable_hpi_hwdep = 1; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable_hpi_hwdep = 1; > > module_param_array(index, int, NULL, S_IRUGO); > MODULE_PARM_DESC(index, "ALSA index value for AudioScience soundcard."); > diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c > --- a/sound/pci/atiixp.c > +++ b/sound/pci/atiixp.c > @@ -43,7 +43,7 @@ static int index = SNDRV_DEFAULT_IDX1; / > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > static int ac97_clock = 48000; > static char *ac97_quirk; > -static int spdif_aclink = 1; > +static bool spdif_aclink = 1; > static int ac97_codec = -1; > > module_param(index, int, 0444); > @@ -60,7 +60,7 @@ module_param(spdif_aclink, bool, 0444); > MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link."); > > /* just for backward compatibility */ > -static int enable; > +static bool enable; > module_param(enable, bool, 0444); > > > diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c > --- a/sound/pci/atiixp_modem.c > +++ b/sound/pci/atiixp_modem.c > @@ -51,7 +51,7 @@ module_param(ac97_clock, int, 0444); > MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz)."); > > /* just for backward compatibility */ > -static int enable; > +static bool enable; > module_param(enable, bool, 0444); > > > diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c > --- a/sound/pci/au88x0/au88x0.c > +++ b/sound/pci/au88x0/au88x0.c > @@ -26,7 +26,7 @@ > // module parameters (see "Module Parameters") > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > static int pcifix[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 255 }; > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c > --- a/sound/pci/aw2/aw2-alsa.c > +++ b/sound/pci/aw2/aw2-alsa.c > @@ -153,7 +153,7 @@ static int snd_aw2_control_switch_captur > ********************************/ > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Audiowerk2 soundcard."); > diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c > --- a/sound/pci/azt3328.c > +++ b/sound/pci/azt3328.c > @@ -301,7 +301,7 @@ static char *id[SNDRV_CARDS] = SNDRV_DEF > module_param_array(id, charp, NULL, 0444); > MODULE_PARM_DESC(id, "ID string for AZF3328 soundcard."); > > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > module_param_array(enable, bool, NULL, 0444); > MODULE_PARM_DESC(enable, "Enable AZF3328 soundcard."); > > diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c > --- a/sound/pci/bt87x.c > +++ b/sound/pci/bt87x.c > @@ -42,9 +42,9 @@ MODULE_SUPPORTED_DEVICE("{{Brooktree,Bt8 > > static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* Exclude the first card */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > static int digital_rate[SNDRV_CARDS]; /* digital input rate */ > -static int load_all; /* allow to load the non-whitelisted cards */ > +static bool load_all; /* allow to load the non-whitelisted cards */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Bt87x soundcard"); > diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c > --- a/sound/pci/ca0106/ca0106_main.c > +++ b/sound/pci/ca0106/ca0106_main.c > @@ -156,7 +156,7 @@ MODULE_SUPPORTED_DEVICE("{{Creative,SB C > // module parameters (see "Module Parameters") > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c > --- a/sound/pci/cmipci.c > +++ b/sound/pci/cmipci.c > @@ -54,10 +54,10 @@ MODULE_SUPPORTED_DEVICE("{{C-Media,CMI87 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > static long mpu_port[SNDRV_CARDS]; > static long fm_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1}; > -static int soft_ac3[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1}; > +static bool soft_ac3[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1}; > #ifdef SUPPORT_JOYSTICK > static int joystick_port[SNDRV_CARDS]; > #endif > diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c > --- a/sound/pci/cs4281.c > +++ b/sound/pci/cs4281.c > @@ -44,8 +44,8 @@ MODULE_SUPPORTED_DEVICE("{{Cirrus Logic, > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > -static int dual_codec[SNDRV_CARDS]; /* dual codec */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > +static bool dual_codec[SNDRV_CARDS]; /* dual codec */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for CS4281 soundcard."); > diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c > --- a/sound/pci/cs46xx/cs46xx.c > +++ b/sound/pci/cs46xx/cs46xx.c > @@ -46,10 +46,10 @@ MODULE_SUPPORTED_DEVICE("{{Cirrus Logic, > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > -static int external_amp[SNDRV_CARDS]; > -static int thinkpad[SNDRV_CARDS]; > -static int mmap_valid[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool external_amp[SNDRV_CARDS]; > +static bool thinkpad[SNDRV_CARDS]; > +static bool mmap_valid[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for the CS46xx soundcard."); > diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c > --- a/sound/pci/cs5535audio/cs5535audio.c > +++ b/sound/pci/cs5535audio/cs5535audio.c > @@ -57,7 +57,7 @@ static struct ac97_quirk ac97_quirks[] _ > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for " DRIVER_NAME); > diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c > --- a/sound/pci/ctxfi/cttimer.c > +++ b/sound/pci/ctxfi/cttimer.c > @@ -15,8 +15,8 @@ > #include "cthardware.h" > #include "cttimer.h" > > -static int use_system_timer; > -MODULE_PARM_DESC(use_system_timer, "Foce to use system-timer"); > +static bool use_system_timer; > +MODULE_PARM_DESC(use_system_timer, "Force to use system-timer"); > module_param(use_system_timer, bool, S_IRUGO); > > struct ct_timer_ops { > diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c > --- a/sound/pci/ctxfi/xfi.c > +++ b/sound/pci/ctxfi/xfi.c > @@ -32,7 +32,7 @@ module_param(multiple, uint, S_IRUGO); > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > static unsigned int subsystem[SNDRV_CARDS]; > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c > --- a/sound/pci/echoaudio/echoaudio.c > +++ b/sound/pci/echoaudio/echoaudio.c > @@ -26,7 +26,7 @@ MODULE_DEVICE_TABLE(pci, snd_echo_ids); > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for " ECHOCARD_NAME " soundcard."); > diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c > --- a/sound/pci/emu10k1/emu10k1.c > +++ b/sound/pci/emu10k1/emu10k1.c > @@ -44,13 +44,13 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > static int extin[SNDRV_CARDS]; > static int extout[SNDRV_CARDS]; > static int seq_ports[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4}; > static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64}; > static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128}; > -static int enable_ir[SNDRV_CARDS]; > +static bool enable_ir[SNDRV_CARDS]; > static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ > static uint delay_pcm_irq[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; > > diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c > --- a/sound/pci/emu10k1/emu10k1x.c > +++ b/sound/pci/emu10k1/emu10k1x.c > @@ -50,7 +50,7 @@ MODULE_SUPPORTED_DEVICE("{{Dell Creative > // module parameters (see "Module Parameters") > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for the EMU10K1X soundcard."); > diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c > --- a/sound/pci/ens1370.c > +++ b/sound/pci/ens1370.c > @@ -83,12 +83,12 @@ MODULE_SUPPORTED_DEVICE("{{Ensoniq,Audio > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > #ifdef SUPPORT_JOYSTICK > #ifdef CHIP1371 > static int joystick_port[SNDRV_CARDS]; > #else > -static int joystick[SNDRV_CARDS]; > +static bool joystick[SNDRV_CARDS]; > #endif > #endif > #ifdef CHIP1371 > diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c > --- a/sound/pci/es1938.c > +++ b/sound/pci/es1938.c > @@ -79,7 +79,7 @@ MODULE_SUPPORTED_DEVICE("{{ESS,ES1938}," > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for ESS Solo-1 soundcard."); > diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c > --- a/sound/pci/es1968.c > +++ b/sound/pci/es1968.c > @@ -132,7 +132,7 @@ MODULE_SUPPORTED_DEVICE("{{ESS,Maestro 2 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 1-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > static int total_bufsize[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1024 }; > static int pcm_substreams_p[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4 }; > static int pcm_substreams_c[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1 }; > @@ -140,7 +140,7 @@ static int clock[SNDRV_CARDS]; > static int use_pm[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; > static int enable_mpu[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; > #ifdef SUPPORT_JOYSTICK > -static int joystick[SNDRV_CARDS]; > +static bool joystick[SNDRV_CARDS]; > #endif > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c > --- a/sound/pci/fm801.c > +++ b/sound/pci/fm801.c > @@ -48,7 +48,7 @@ MODULE_SUPPORTED_DEVICE("{{ForteMedia,FM > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > /* > * Enable TEA575x tuner > * 1 = MediaForte 256-PCS > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -58,13 +58,13 @@ > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > static char *model[SNDRV_CARDS]; > static int position_fix[SNDRV_CARDS]; > static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; > static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; > static int probe_only[SNDRV_CARDS]; > -static int single_cmd; > +static bool single_cmd; > static int enable_msi = -1; > #ifdef CONFIG_SND_HDA_PATCH_LOADER > static char *patch[SNDRV_CARDS]; > @@ -116,12 +116,12 @@ MODULE_PARM_DESC(power_save, "Automatic > * this may give more power-saving, but will take longer time to > * wake up. > */ > -static int power_save_controller = 1; > +static bool power_save_controller = 1; > module_param(power_save_controller, bool, 0644); > MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); > #endif > > -static int align_buffer_size = 1; > +static bool align_buffer_size = 1; > module_param(align_buffer_size, bool, 0644); > MODULE_PARM_DESC(align_buffer_size, > "Force buffer and period sizes to be multiple of 128 bytes."); > diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c > --- a/sound/pci/ice1712/ice1712.c > +++ b/sound/pci/ice1712/ice1712.c > @@ -84,9 +84,9 @@ MODULE_SUPPORTED_DEVICE("{" > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ > static char *model[SNDRV_CARDS]; > -static int omni[SNDRV_CARDS]; /* Delta44 & 66 Omni I/O support */ > +static bool omni[SNDRV_CARDS]; /* Delta44 & 66 Omni I/O support */ > static int cs8427_timeout[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 500}; /* CS8427 S/PDIF transceiver reset timeout value in msec */ > static int dxr_enable[SNDRV_CARDS]; /* DXR enable for DMX6FIRE */ > > diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c > --- a/sound/pci/ice1712/ice1724.c > +++ b/sound/pci/ice1712/ice1724.c > @@ -80,7 +80,7 @@ MODULE_SUPPORTED_DEVICE("{" > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > static char *model[SNDRV_CARDS]; > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c > --- a/sound/pci/intel8x0.c > +++ b/sound/pci/intel8x0.c > @@ -79,9 +79,9 @@ static int index = SNDRV_DEFAULT_IDX1; / > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > static int ac97_clock; > static char *ac97_quirk; > -static int buggy_semaphore; > +static bool buggy_semaphore; > static int buggy_irq = -1; /* auto-check */ > -static int xbox; > +static bool xbox; > static int spdif_aclink = -1; > static int inside_vm = -1; > > @@ -105,7 +105,7 @@ module_param(inside_vm, bint, 0444); > MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization."); > > /* just for backward compatibility */ > -static int enable; > +static bool enable; > module_param(enable, bool, 0444); > static int joystick; > module_param(joystick, int, 0444); > diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c > --- a/sound/pci/intel8x0m.c > +++ b/sound/pci/intel8x0m.c > @@ -68,7 +68,7 @@ module_param(ac97_clock, int, 0444); > MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (0 = auto-detect)."); > > /* just for backward compatibility */ > -static int enable; > +static bool enable; > module_param(enable, bool, 0444); > > /* > diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c > --- a/sound/pci/korg1212/korg1212.c > +++ b/sound/pci/korg1212/korg1212.c > @@ -408,7 +408,7 @@ MODULE_FIRMWARE("korg/k1212.dsp"); > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Korg 1212 soundcard."); > diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c > --- a/sound/pci/lola/lola.c > +++ b/sound/pci/lola/lola.c > @@ -35,7 +35,7 @@ > /* Standard options */ > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Digigram Lola driver."); > diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c > --- a/sound/pci/lx6464es/lx6464es.c > +++ b/sound/pci/lx6464es/lx6464es.c > @@ -42,7 +42,7 @@ MODULE_SUPPORTED_DEVICE("{digigram lx646 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Digigram LX6464ES interface."); > diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c > --- a/sound/pci/maestro3.c > +++ b/sound/pci/maestro3.c > @@ -64,8 +64,8 @@ MODULE_FIRMWARE("ess/maestro3_assp_minis > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* all enabled */ > -static int external_amp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* all enabled */ > +static bool external_amp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; > static int amp_gpio[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1}; > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c > --- a/sound/pci/mixart/mixart.c > +++ b/sound/pci/mixart/mixart.c > @@ -49,7 +49,7 @@ MODULE_SUPPORTED_DEVICE("{{Digigram," CA > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Digigram " CARD_NAME " soundcard."); > diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c > --- a/sound/pci/nm256/nm256.c > +++ b/sound/pci/nm256/nm256.c > @@ -57,12 +57,12 @@ static int index = SNDRV_DEFAULT_IDX1; / > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > static int playback_bufsize = 16; > static int capture_bufsize = 16; > -static int force_ac97; /* disabled as default */ > +static bool force_ac97; /* disabled as default */ > static int buffer_top; /* not specified */ > -static int use_cache; /* disabled */ > -static int vaio_hack; /* disabled */ > -static int reset_workaround; > -static int reset_workaround_2; > +static bool use_cache; /* disabled */ > +static bool vaio_hack; /* disabled */ > +static bool reset_workaround; > +static bool reset_workaround_2; > > module_param(index, int, 0444); > MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); > @@ -86,7 +86,7 @@ module_param(reset_workaround_2, bool, 0 > MODULE_PARM_DESC(reset_workaround_2, "Enable extended AC97 RESET workaround for some other laptops."); > > /* just for backward compatibility */ > -static int enable; > +static bool enable; > module_param(enable, bool, 0444); > > > diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c > --- a/sound/pci/oxygen/oxygen.c > +++ b/sound/pci/oxygen/oxygen.c > @@ -74,7 +74,7 @@ MODULE_SUPPORTED_DEVICE("{{C-Media,CMI87 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "card index"); > diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c > --- a/sound/pci/oxygen/virtuoso.c > +++ b/sound/pci/oxygen/virtuoso.c > @@ -32,7 +32,7 @@ MODULE_SUPPORTED_DEVICE("{{Asus,AV66},{A > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "card index"); > diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c > --- a/sound/pci/pcxhr/pcxhr.c > +++ b/sound/pci/pcxhr/pcxhr.c > @@ -52,8 +52,8 @@ MODULE_SUPPORTED_DEVICE("{{Digigram," DR > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ > -static int mono[SNDRV_CARDS]; /* capture mono only */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ > +static bool mono[SNDRV_CARDS]; /* capture mono only */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Digigram " DRIVER_NAME " soundcard"); > diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c > --- a/sound/pci/riptide/riptide.c > +++ b/sound/pci/riptide/riptide.c > @@ -122,7 +122,7 @@ MODULE_FIRMWARE("riptide.hex"); > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; > > #ifdef SUPPORT_JOYSTICK > static int joystick_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x200 }; > diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c > --- a/sound/pci/rme32.c > +++ b/sound/pci/rme32.c > @@ -89,8 +89,8 @@ > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > -static int fullduplex[SNDRV_CARDS]; // = {[0 ... (SNDRV_CARDS - 1)] = 1}; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool fullduplex[SNDRV_CARDS]; // = {[0 ... (SNDRV_CARDS - 1)] = 1}; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for RME Digi32 soundcard."); > diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c > --- a/sound/pci/rme96.c > +++ b/sound/pci/rme96.c > @@ -53,7 +53,7 @@ MODULE_SUPPORTED_DEVICE("{{RME,Digi96}," > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for RME Digi96 soundcard."); > diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c > --- a/sound/pci/rme9652/hdsp.c > +++ b/sound/pci/rme9652/hdsp.c > @@ -45,7 +45,7 @@ > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for RME Hammerfall DSP interface."); > diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c > --- a/sound/pci/rme9652/hdspm.c > +++ b/sound/pci/rme9652/hdspm.c > @@ -61,7 +61,7 @@ > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for RME HDSPM interface."); > diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c > --- a/sound/pci/rme9652/rme9652.c > +++ b/sound/pci/rme9652/rme9652.c > @@ -38,8 +38,8 @@ > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > -static int precise_ptr[SNDRV_CARDS]; /* Enable precise pointer */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool precise_ptr[SNDRV_CARDS]; /* Enable precise pointer */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for RME Digi9652 (Hammerfall) soundcard."); > diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c > --- a/sound/pci/sis7019.c > +++ b/sound/pci/sis7019.c > @@ -40,7 +40,7 @@ MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 A > > static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > -static int enable = 1; > +static bool enable = 1; > > module_param(index, int, 0444); > MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); > diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c > --- a/sound/pci/sonicvibes.c > +++ b/sound/pci/sonicvibes.c > @@ -52,9 +52,9 @@ MODULE_SUPPORTED_DEVICE("{{S3,SonicVibes > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > -static int reverb[SNDRV_CARDS]; > -static int mge[SNDRV_CARDS]; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool reverb[SNDRV_CARDS]; > +static bool mge[SNDRV_CARDS]; > static unsigned int dmaio = 0x7a00; /* DDMA i/o address */ > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c > --- a/sound/pci/trident/trident.c > +++ b/sound/pci/trident/trident.c > @@ -47,7 +47,7 @@ MODULE_SUPPORTED_DEVICE("{{Trident,4DWav > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 32}; > static int wavetable_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8192}; > > diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c > --- a/sound/pci/via82xx.c > +++ b/sound/pci/via82xx.c > @@ -80,7 +80,7 @@ static int index = SNDRV_DEFAULT_IDX1; / > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > static long mpu_port; > #ifdef SUPPORT_JOYSTICK > -static int joystick; > +static bool joystick; > #endif > static int ac97_clock = 48000; > static char *ac97_quirk; > @@ -110,7 +110,7 @@ module_param(nodelay, int, 0444); > MODULE_PARM_DESC(nodelay, "Disable 500ms init delay"); > > /* just for backward compatibility */ > -static int enable; > +static bool enable; > module_param(enable, bool, 0444); > > > diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c > --- a/sound/pci/via82xx_modem.c > +++ b/sound/pci/via82xx_modem.c > @@ -66,7 +66,7 @@ module_param(ac97_clock, int, 0444); > MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz)."); > > /* just for backward compatibility */ > -static int enable; > +static bool enable; > module_param(enable, bool, 0444); > > > diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c > --- a/sound/pci/vx222/vx222.c > +++ b/sound/pci/vx222/vx222.c > @@ -37,8 +37,8 @@ MODULE_SUPPORTED_DEVICE("{{Digigram," CA > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > -static int mic[SNDRV_CARDS]; /* microphone */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool mic[SNDRV_CARDS]; /* microphone */ > static int ibl[SNDRV_CARDS]; /* microphone */ > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c > --- a/sound/pci/ymfpci/ymfpci.c > +++ b/sound/pci/ymfpci/ymfpci.c > @@ -41,13 +41,13 @@ MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF724 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > static long fm_port[SNDRV_CARDS]; > static long mpu_port[SNDRV_CARDS]; > #ifdef SUPPORT_JOYSTICK > static long joystick_port[SNDRV_CARDS]; > #endif > -static int rear_switch[SNDRV_CARDS]; > +static bool rear_switch[SNDRV_CARDS]; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for the Yamaha DS-1 PCI soundcard."); > diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c > --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c > +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c > @@ -39,7 +39,7 @@ MODULE_SUPPORTED_DEVICE("{{Sound Core," > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); > diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c > --- a/sound/pcmcia/vx/vxpocket.c > +++ b/sound/pcmcia/vx/vxpocket.c > @@ -39,7 +39,7 @@ MODULE_SUPPORTED_DEVICE("{{Digigram,VXPo > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ > static int ibl[SNDRV_CARDS]; > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c > --- a/sound/ppc/powermac.c > +++ b/sound/ppc/powermac.c > @@ -36,7 +36,7 @@ MODULE_LICENSE("GPL"); > > static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ > static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ > -static int enable_beep = 1; > +static bool enable_beep = 1; > > module_param(index, int, 0444); > MODULE_PARM_DESC(index, "Index value for " CHIP_NAME " soundchip."); > diff --git a/sound/sh/aica.c b/sound/sh/aica.c > --- a/sound/sh/aica.c > +++ b/sound/sh/aica.c > @@ -55,7 +55,7 @@ MODULE_FIRMWARE("aica_firmware.bin"); > #define CARD_NAME "AICA" > static int index = -1; > static char *id; > -static int enable = 1; > +static bool enable = 1; > module_param(index, int, 0444); > MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); > module_param(id, charp, 0444); > diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c > --- a/sound/sparc/amd7930.c > +++ b/sound/sparc/amd7930.c > @@ -50,7 +50,7 @@ > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Sun AMD7930 soundcard."); > diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c > --- a/sound/sparc/cs4231.c > +++ b/sound/sparc/cs4231.c > @@ -40,7 +40,7 @@ > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > /* Enable this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Sun CS4231 soundcard."); > diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c > --- a/sound/sparc/dbri.c > +++ b/sound/sparc/dbri.c > @@ -80,7 +80,7 @@ MODULE_SUPPORTED_DEVICE("{{Sun,DBRI}}"); > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > /* Enable this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for Sun DBRI soundcard."); > diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c > --- a/sound/usb/6fire/chip.c > +++ b/sound/usb/6fire/chip.c > @@ -35,7 +35,7 @@ MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable card */ > static struct sfire_chip *chips[SNDRV_CARDS] = SNDRV_DEFAULT_PTR; > static struct usb_device *devices[SNDRV_CARDS] = SNDRV_DEFAULT_PTR; > > diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c > --- a/sound/usb/caiaq/device.c > +++ b/sound/usb/caiaq/device.c > @@ -55,7 +55,7 @@ MODULE_SUPPORTED_DEVICE("{{Native Instru > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ > static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > static int snd_card_used[SNDRV_CARDS]; > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/usb/card.c b/sound/usb/card.c > --- a/sound/usb/card.c > +++ b/sound/usb/card.c > @@ -78,14 +78,14 @@ MODULE_SUPPORTED_DEVICE("{{Generic,USB A > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ > /* Vendor/product IDs for this card */ > static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; > static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; > static int nrpacks = 8; /* max. number of packets per urb */ > -static int async_unlink = 1; > +static bool async_unlink = 1; > static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ > -static int ignore_ctl_error; > +static bool ignore_ctl_error; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); > diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c > --- a/sound/usb/misc/ua101.c > +++ b/sound/usb/misc/ua101.c > @@ -52,7 +52,7 @@ MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > static unsigned int queue_length = 21; > > module_param_array(index, int, NULL, 0444); > diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c > --- a/sound/usb/usx2y/us122l.c > +++ b/sound/usb/usx2y/us122l.c > @@ -37,7 +37,7 @@ MODULE_LICENSE("GPL"); > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ > static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ > /* Enable this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for "NAME_ALLCAPS"."); > diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c > --- a/sound/usb/usx2y/usbusx2y.c > +++ b/sound/usb/usx2y/usbusx2y.c > @@ -154,7 +154,7 @@ MODULE_SUPPORTED_DEVICE("{{TASCAM(0x1604 > > static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ > static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ > -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ > > module_param_array(index, int, NULL, 0444); > MODULE_PARM_DESC(index, "Index value for "NAME_ALLCAPS"."); >

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[alsa-devel] [PATCH 0/5] snd-usb endpoint logic rework, version 3
by Daniel Mack 19 Dec '11

19 Dec '11

This is the 3rd round, and I didn't mark it RFC anymore as Clemens and Takashi agreed on the way of implementation in general. So in comparison to the last version, this round incorporates some spinlock fixes, a lot less debugging output and my S-o-b lines. Please test this extensively with different sound cards. I didn't experience a single problem over the last hours of testing, but the more testing the better. Thanks, Daniel Daniel Mack (5): ALSA: snd-usb: add snd_usb_audio-wide mutex ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: add support for implicit feedback sound/usb/card.c | 8 +- sound/usb/card.h | 65 +++ sound/usb/endpoint.c | 1427 +++++++++++++++++++++++++------------------------- sound/usb/endpoint.h | 31 +- sound/usb/pcm.c | 428 +++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 2 + 7 files changed, 1191 insertions(+), 801 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH 0/5] RFC v2 for snd-usb endpoint logic rework
by Daniel Mack 19 Dec '11

19 Dec '11

This is v2 of the patch bulk I posted some weeks ago, which incoporates the points that were discussed. Thanks to everyone who commented and helped with sugesstions. This version work really well here, but as last time, it needs a lot more testing. What's still missing are some cleanups and more documentation, I'll do that once the acutal content is agreed upon. Patches should apply cleanly to the latest mainline git + Takashi's sound tree. Thanks for any feedback for this next round, Daniel Daniel Mack (5): ALSA: snd-usb: add snd_usb_audio-wide mutex ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: add support for implicit feedback sound/usb/card.c | 8 +- sound/usb/card.h | 65 +++ sound/usb/endpoint.c | 1434 +++++++++++++++++++++++++------------------------- sound/usb/endpoint.h | 31 +- sound/usb/pcm.c | 444 +++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 2 + 7 files changed, 1214 insertions(+), 801 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH 1/4] ASoC: Convert WM8960 to devm_kzalloc()
by Mark Brown 17 Dec '11

17 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8960.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 2315b86..e5caae3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -973,7 +973,8 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, struct wm8960_priv *wm8960; int ret; - wm8960 = kzalloc(sizeof(struct wm8960_priv), GFP_KERNEL); + wm8960 = devm_kzalloc(&i2c->dev, sizeof(struct wm8960_priv), + GFP_KERNEL); if (wm8960 == NULL) return -ENOMEM; @@ -982,15 +983,13 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); - if (ret < 0) - kfree(wm8960); + return ret; } static __devexit int wm8960_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.7.3

1 3

[alsa-devel] [PATCH 0/2] OMAP4: McBSP: ASoC: Fix FIFO configuration
by Peter Ujfalusi 17 Dec '11

17 Dec '11

Hello, All McBSP ports in OMAP4 has 128 word long FIFO which is not taken into account. Fix the configuration so we can configure the FIFO for McBSP in OMAP4. Regards, Peter --- Peter Ujfalusi (2): OMAP4: mcbsp: Enable FIFO usage ASoC: omap-mcbsp: Enable FIFO usage on OMAP4 arch/arm/mach-omap2/mcbsp.c | 3 +++ sound/soc/omap/omap-mcbsp.c | 2 +- 2 files changed, 4 insertions(+), 1 deletions(-) -- 1.7.8

3 5

[alsa-devel] [PATCH 0/5] ASoC: Convert pxa machine drivers to use use snd_soc_register_card()
by Axel Lin 17 Dec '11

17 Dec '11

This serial converts some pxa machine drivers to use use snd_soc_register_card(). Note: I don't have these hardware, so I appreciate if someone can test these patches. Axel Lin (5): ASoC: pxa: Convert e740_wm9705 to use snd_soc_register_card() ASoC: pxa: Convert e750_wm9705 to use snd_soc_register_card() ASoC: pxa: Convert e800_wm9712 to use snd_soc_register_card() ASoC: pxa: Convert imote2 to use snd_soc_register_card() ASoC: pxa: Convert tosa to use snd_soc_register_card() arch/arm/mach-pxa/eseries.c | 18 +++++++++ arch/arm/mach-pxa/stargate2.c | 6 +++ arch/arm/mach-pxa/tosa.c | 6 +++ sound/soc/pxa/e740_wm9705.c | 75 +++++++++++++++++----------------------- sound/soc/pxa/e750_wm9705.c | 66 ++++++++++++++++------------------ sound/soc/pxa/e800_wm9712.c | 66 ++++++++++++++++------------------ sound/soc/pxa/imote2.c | 41 +++++++++++++--------- sound/soc/pxa/tosa.c | 77 ++++++++++++++++------------------------- 8 files changed, 178 insertions(+), 177 deletions(-) -- 1.7.5.4

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[alsa-devel] Alsa-lib version info
by Alan Horstmann 16 Dec '11

16 Dec '11

Hi, I have read in the docs about snd_asoundlib_version() to get a version string, but is there a way to get the SND_LIB_VERSION number from a runtime Alsa-lib, rather than from the header included at application build time, or would it have to be constructed from the string? Thanks Alan

2 2

[alsa-devel] Jabra SPEAK 410 USB - no audio playback
by Richard W.M. Jones 16 Dec '11

16 Dec '11

I got one of these rather nice USB conference phones. It works fine with Windows/Mac. ALSA gives the following errors: [336517.162535] usb 1-1.1: SerialNumber: 0023781E0530x010100 [336517.172419] ALSA sound/usb/stream.c:176 10:1:1 : no or invalid class specific endpoint descriptor [336517.177029] ALSA sound/usb/stream.c:176 10:2:1 : no or invalid class specific endpoint descriptor [336517.183248] ALSA sound/usb/mixer.c:795 2:0: cannot get min/max values for control 2 (id 2) [336517.183350] ALSA sound/usb/mixer.c:795 5:0: cannot get min/max values for control 2 (id 5) [336517.236151] input: Jabra SPEAK 410 USB as /devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.1/1-1.1:1.3/input/input13 The microphone and other features are fine, but there is no playback. I hope I've attached all possible details about this device, but if there is anything I've missed out, please let me know. Rich. -- Richard Jones, Virtualization Group, Red Hat http://people.redhat.com/~rjones libguestfs lets you edit virtual machines. Supports shell scripting, bindings from many languages. http://libguestfs.org

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[alsa-devel] [PATCH v4 0/5] Support for OMAP4 Digital Microphone interface
by Peter Ujfalusi 16 Dec '11

16 Dec '11

Hello, The following series will add support for OMAP4 DMIC interface, and enable them on sdp4430/Blaze boards. Changes since v3: - dmic clk divider selection moved to hw_params phase to allow fixed rate configuration from machine driver init. Changes since v2: - Use module_platform_driver - convert to use devm_ - Add clk_id for the output clock - Extend the comment to explain why the driven can change the divider for 192KHz sampling rate. Changes since v1: - 192KHz rate support fixed (need to use different divider) - Hold reference for dmic_fclk over the lifetime of the driver - Removed the clkdiv callback, and replaced with frequency based configuration for the external components (dmic frequency) Regards, Peter --- Peter Ujfalusi (5): OMAP4: hwmod: Add names for DMIC memory address space ASoC: OMAP4: omap-dmic: Initial support for OMAP DMIC OMAP4: devices: Register OMAP4 DMIC platform device OMAP4: board-4430sdp: Register platform device for digimic codec ASoC: sdp4430: Add support for digital microphones arch/arm/mach-omap2/board-4430sdp.c | 6 + arch/arm/mach-omap2/devices.c | 22 ++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + sound/soc/omap/Kconfig | 5 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-dmic.c | 549 ++++++++++++++++++++++++++++ sound/soc/omap/omap-dmic.h | 69 ++++ sound/soc/omap/sdp4430.c | 85 ++++- 8 files changed, 729 insertions(+), 11 deletions(-) create mode 100644 sound/soc/omap/omap-dmic.c create mode 100644 sound/soc/omap/omap-dmic.h -- 1.7.8.rc3

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[alsa-devel] [PATCH] Fix compile error in sound/soc/mid-x86/sst_platform.c
by Joerg Roedel 16 Dec '11

16 Dec '11

(again with mailing lists on Cc, sorry) Hi, I tested current linux-next against my iommu-configs and found a compile error in sound/soc/mid-x86/sst_platform.c. The variable ret_val in the sst_platform_open() function is used but not declared. Attached patch fixes this.

4 4

[alsa-devel] [PATCH] ASoC: Remove unused variables in wm9090.c
by Axel Lin 16 Dec '11

16 Dec '11

Fix below build warnings: CC sound/soc/codecs/wm9090.o sound/soc/codecs/wm9090.c: In function 'wm9090_set_bias_level': sound/soc/codecs/wm9090.c:496: warning: unused variable 'ret' sound/soc/codecs/wm9090.c:496: warning: unused variable 'i' sound/soc/codecs/wm9090.c:495: warning: unused variable 'reg_cache' sound/soc/codecs/wm9090.c: In function 'wm9090_i2c_remove': sound/soc/codecs/wm9090.c:658: warning: unused variable 'wm9090' Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm9090.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 41ebe0d..acac7d6 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -492,9 +492,6 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) static int wm9090_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 *reg_cache = codec->reg_cache; - int i, ret; - switch (level) { case SND_SOC_BIAS_ON: break; @@ -655,8 +652,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) { - struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); - snd_soc_unregister_codec(&i2c->dev); return 0; -- 1.7.5.4

2 1

[alsa-devel] [RFC PATCH 0/1] ASoC: dsd1791: Introduce driver for TI DSD1791 stereo codec
by Michael Williamson 16 Dec '11

16 Dec '11

The following patch is an introduction of a codec driver for the TI DSD1791 audio part. This patch adds basic support and has been tested using a MityDSP-L138F SOM sitting on a Critical Link Industrial I/O evaluation board in 16 bit and 24 bit I2S mode. This is an RFC, as I'm still wandering around the ALSA framework a bit. Comments and guidance greatly appreciated. I can follow with control registers (gain, etc.) if this stuff looks OK. Thanks. -Mike Michael Williamson (1): ASoC: dsd1791: Introduce driver for TI DSD1791 stereo codec sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dsd1791.c | 282 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 288 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/dsd1791.c -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Fix build error in sst_platform.c
by Axel Lin 16 Dec '11

16 Dec '11

Fix below build error: CC sound/soc/mid-x86/sst_platform.o sound/soc/mid-x86/sst_platform.c: In function 'sst_platform_open': sound/soc/mid-x86/sst_platform.c:274:2: error: 'ret_val' undeclared (first use in this function) sound/soc/mid-x86/sst_platform.c:274:2: note: each undeclared identifier is reported only once for each function it appears in make[3]: *** [sound/soc/mid-x86/sst_platform.o] Error 1 make[2]: *** [sound/soc/mid-x86] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/mid-x86/sst_platform.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 24f9471..11c39c5 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -267,6 +267,7 @@ static int sst_platform_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; + int ret_val; pr_debug("sst_platform_open called\n"); -- 1.7.5.4

1 0

Re: [alsa-devel] [PATCH 2/15] module_param: avoid bool abuse, add bint for special cases.
by Guenter Roeck 15 Dec '11

15 Dec '11

On Wed, Dec 14, 2011 at 10:02:26PM -0500, Rusty Russell wrote: > For historical reasons, we allow module_param(bool) to take an int (or > an unsigned int). That's going away. > > A few drivers really want an int: they set it to -1 and a parameter > will set it to 0 or 1. This sucks: reading them from sysfs will give > 'Y' for both -1 and 1, but if we change it to an int, then the users > might be broken (if they did "param" instead of "param=1"). > > Use a new 'bint' parser for them. > > (ntfs has a different problem: it needs an int for debug_msgs because > it's also exposed via sysctl.) > > Cc: Steve Glendinning <steve.glendinning(a)smsc.com> > Cc: Jean Delvare <khali(a)linux-fr.org> > Cc: Guenter Roeck <guenter.roeck(a)ericsson.com> > Cc: Hoang-Nam Nguyen <hnguyen(a)de.ibm.com> > Cc: Christoph Raisch <raisch(a)de.ibm.com> > Cc: Roland Dreier <roland(a)kernel.org> > Cc: Sean Hefty <sean.hefty(a)intel.com> > Cc: Hal Rosenstock <hal.rosenstock(a)gmail.com> > Cc: linux390(a)de.ibm.com > Cc: Anton Altaparmakov <anton(a)tuxera.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: lm-sensors(a)lm-sensors.org > Cc: linux-rdma(a)vger.kernel.org > Cc: linux-s390(a)vger.kernel.org > Cc: linux-ntfs-dev(a)lists.sourceforge.net > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Rusty Russell <rusty(a)rustcorp.com.au> > --- > drivers/hwmon/emc2103.c | 2 +- For the hwmon driver: Acked-by: Guenter Roeck <guenter.roeck(a)ericsson.com> > drivers/infiniband/hw/ehca/ehca_main.c | 2 +- > drivers/s390/cio/cmf.c | 2 +- > fs/ntfs/super.c | 2 +- > include/linux/moduleparam.h | 6 ++++++ > kernel/params.c | 24 ++++++++++++++++++++++++ > sound/pci/intel8x0.c | 4 ++-- > 7 files changed, 36 insertions(+), 6 deletions(-) > > diff --git a/drivers/hwmon/emc2103.c b/drivers/hwmon/emc2103.c > --- a/drivers/hwmon/emc2103.c > +++ b/drivers/hwmon/emc2103.c > @@ -55,7 +55,7 @@ static const u8 REG_TEMP_MAX[4] = { 0x34 > * it. Default is to leave the device in the state it's already in (-1). > * This parameter allows APD mode to be optionally forced on or off */ > static int apd = -1; > -module_param(apd, bool, 0); > +module_param(apd, bint, 0); > MODULE_PARM_DESC(init, "Set to zero to disable anti-parallel diode mode"); > > struct temperature { > diff --git a/drivers/infiniband/hw/ehca/ehca_main.c b/drivers/infiniband/hw/ehca/ehca_main.c > --- a/drivers/infiniband/hw/ehca/ehca_main.c > +++ b/drivers/infiniband/hw/ehca/ehca_main.c > @@ -82,7 +82,7 @@ module_param_named(port_act_time, ehca_p > module_param_named(poll_all_eqs, ehca_poll_all_eqs, bool, S_IRUGO); > module_param_named(static_rate, ehca_static_rate, int, S_IRUGO); > module_param_named(scaling_code, ehca_scaling_code, bool, S_IRUGO); > -module_param_named(lock_hcalls, ehca_lock_hcalls, bool, S_IRUGO); > +module_param_named(lock_hcalls, ehca_lock_hcalls, bint, S_IRUGO); > module_param_named(number_of_cqs, ehca_max_cq, int, S_IRUGO); > module_param_named(number_of_qps, ehca_max_qp, int, S_IRUGO); > > diff --git a/drivers/s390/cio/cmf.c b/drivers/s390/cio/cmf.c > --- a/drivers/s390/cio/cmf.c > +++ b/drivers/s390/cio/cmf.c > @@ -98,7 +98,7 @@ enum cmb_format { > * enum cmb_format. > */ > static int format = CMF_AUTODETECT; > -module_param(format, bool, 0444); > +module_param(format, bint, 0444); > > /** > * struct cmb_operations - functions to use depending on cmb_format > diff --git a/fs/ntfs/super.c b/fs/ntfs/super.c > --- a/fs/ntfs/super.c > +++ b/fs/ntfs/super.c > @@ -3198,7 +3198,7 @@ MODULE_DESCRIPTION("NTFS 1.2/3.x driver > MODULE_VERSION(NTFS_VERSION); > MODULE_LICENSE("GPL"); > #ifdef DEBUG > -module_param(debug_msgs, bool, 0); > +module_param(debug_msgs, bint, 0); > MODULE_PARM_DESC(debug_msgs, "Enable debug messages."); > #endif > > diff --git a/include/linux/moduleparam.h b/include/linux/moduleparam.h > --- a/include/linux/moduleparam.h > +++ b/include/linux/moduleparam.h > @@ -367,6 +367,12 @@ extern int param_set_invbool(const char > extern int param_get_invbool(char *buffer, const struct kernel_param *kp); > #define param_check_invbool(name, p) __param_check(name, p, bool) > > +/* An int, which can only be set like a bool (though it shows as an int). */ > +extern struct kernel_param_ops param_ops_bint; > +extern int param_set_bint(const char *val, const struct kernel_param *kp); > +#define param_get_bint param_get_int > +#define param_check_bint param_check_int > + > /** > * module_param_array - a parameter which is an array of some type > * @name: the name of the array variable > diff --git a/kernel/params.c b/kernel/params.c > --- a/kernel/params.c > +++ b/kernel/params.c > @@ -369,6 +369,30 @@ struct kernel_param_ops param_ops_invboo > }; > EXPORT_SYMBOL(param_ops_invbool); > > +int param_set_bint(const char *val, const struct kernel_param *kp) > +{ > + struct kernel_param boolkp; > + bool v; > + int ret; > + > + /* Match bool exactly, by re-using it. */ > + boolkp = *kp; > + boolkp.arg = &v; > + boolkp.flags |= KPARAM_ISBOOL; > + > + ret = param_set_bool(val, &boolkp); > + if (ret == 0) > + *(int *)kp->arg = v; > + return ret; > +} > +EXPORT_SYMBOL(param_set_bint); > + > +struct kernel_param_ops param_ops_bint = { > + .set = param_set_bint, > + .get = param_get_int, > +}; > +EXPORT_SYMBOL(param_ops_bint); > + > /* We break the rule and mangle the string. */ > static int param_array(const char *name, > const char *val, > diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c > --- a/sound/pci/intel8x0.c > +++ b/sound/pci/intel8x0.c > @@ -95,13 +95,13 @@ module_param(ac97_quirk, charp, 0444); > MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware."); > module_param(buggy_semaphore, bool, 0444); > MODULE_PARM_DESC(buggy_semaphore, "Enable workaround for hardwares with problematic codec semaphores."); > -module_param(buggy_irq, bool, 0444); > +module_param(buggy_irq, bint, 0444); > MODULE_PARM_DESC(buggy_irq, "Enable workaround for buggy interrupts on some motherboards."); > module_param(xbox, bool, 0444); > MODULE_PARM_DESC(xbox, "Set to 1 for Xbox, if you have problems with the AC'97 codec detection."); > module_param(spdif_aclink, int, 0444); > MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link."); > -module_param(inside_vm, bool, 0444); > +module_param(inside_vm, bint, 0444); > MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization."); > > /* just for backward compatibility */ -- Guenter Roeck Distinguished Engineer PDU IP Systems

1 0

[PATCH] Fix compile error in sound/soc/mid-x86/sst_platform.c
by Joerg Roedel 15 Dec '11

15 Dec '11

The variable ret_val is used but not declared. Fix this. Signed-off-by: Joerg Roedel <joerg.roedel(a)amd.com> --- sound/soc/mid-x86/sst_platform.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 24f9471..11c39c5 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -267,6 +267,7 @@ static int sst_platform_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; + int ret_val; pr_debug("sst_platform_open called\n"); -- 1.7.5.4 -- AMD Operating System Research Center Advanced Micro Devices GmbH Einsteinring 24 85609 Dornach General Managers: Alberto Bozzo, Andrew Bowd Registration: Dornach, Landkr. Muenchen; Registerger. Muenchen, HRB Nr. 43632

1 0

[PATCH] Fix compile error in sound/soc/mid-x86/sst_platform.c
by Joerg Roedel 15 Dec '11

15 Dec '11

The variable ret_val is used but not declared. This causes the following compile error: sound/soc/mid-x86/sst_platform.c: In function =E2=80=98sst_platform_open=E2= =80=99: sound/soc/mid-x86/sst_platform.c:274:2: error: =E2=80=98ret_val=E2=80=99 = undeclared (first use in this function) sound/soc/mid-x86/sst_platform.c:274:2: note: each undeclared identifier = is reported only once for each function it appears in make[1]: *** [sound/soc/mid-x86/sst_platform.o] Error 1 Fix this. Signed-off-by: Joerg Roedel <joerg.roedel(a)amd.com> --- sound/soc/mid-x86/sst_platform.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_pla= tform.c index 24f9471..11c39c5 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -267,6 +267,7 @@ static int sst_platform_open(struct snd_pcm_substream= *substream) { struct snd_pcm_runtime *runtime =3D substream->runtime; struct sst_runtime_stream *stream; + int ret_val; =20 pr_debug("sst_platform_open called\n"); =20 --=20 1.7.5.4 --=20 AMD Operating System Research Center Advanced Micro Devices GmbH Einsteinring 24 85609 Dornach General Managers: Alberto Bozzo, Andrew Bowd Registration: Dornach, Landkr. Muenchen; Registerger. Muenchen, HRB Nr. 4= 3632

1 0

[alsa-devel] [PATCH] ALSA: hda: invert mute led polarity for HP laptops
by Gustavo Maciel Dias Vieira 15 Dec '11

15 Dec '11

At least two HP laptop models with the IDT 92HD81B1X5 codec have the polarity of the mute led inverted with respect to the current driver default. Change the polarity for these machines only. Signed-off-by: Gustavo Maciel Dias Vieira <gustavo(a)sagui.org> --- sound/pci/hda/patch_sigmatel.c | 12 +++++++++++- 1 files changed, 11 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d8d2f9d..c920a27 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5601,6 +5601,16 @@ static void stac92hd8x_fill_auto_spec(struct hda_codec *codec) spec->num_dmics = spec->auto_dmic_cnt; } +static int stac92hd8x_hp_mute_led_polarity(struct hda_codec *codec) +{ + switch (codec->subsystem_id) { + case 0x103c1650: + return 1; + } + return 0; +} + + static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5642,7 +5652,7 @@ again: codec->patch_ops = stac92xx_patch_ops; - if (find_mute_led_gpio(codec, 0)) + if (find_mute_led_gpio(codec, stac92hd8x_hp_mute_led_polarity(codec))) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); -- 1.7.7.3

2 6

Re: [alsa-devel] [PATCH 2/15] module_param: avoid bool abuse, add bint for special cases.
by Takashi Iwai 15 Dec '11

15 Dec '11

At Thu, 15 Dec 2011 13:32:26 +1030, Rusty Russell wrote: > > For historical reasons, we allow module_param(bool) to take an int (or > an unsigned int). That's going away. > > A few drivers really want an int: they set it to -1 and a parameter > will set it to 0 or 1. This sucks: reading them from sysfs will give > 'Y' for both -1 and 1, but if we change it to an int, then the users > might be broken (if they did "param" instead of "param=1"). > > Use a new 'bint' parser for them. > > (ntfs has a different problem: it needs an int for debug_msgs because > it's also exposed via sysctl.) > > Cc: Steve Glendinning <steve.glendinning(a)smsc.com> > Cc: Jean Delvare <khali(a)linux-fr.org> > Cc: Guenter Roeck <guenter.roeck(a)ericsson.com> > Cc: Hoang-Nam Nguyen <hnguyen(a)de.ibm.com> > Cc: Christoph Raisch <raisch(a)de.ibm.com> > Cc: Roland Dreier <roland(a)kernel.org> > Cc: Sean Hefty <sean.hefty(a)intel.com> > Cc: Hal Rosenstock <hal.rosenstock(a)gmail.com> > Cc: linux390(a)de.ibm.com > Cc: Anton Altaparmakov <anton(a)tuxera.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> For the sound part, Acked-by: Takashi Iwai <tiwai(a)suse.de> It's a nice feature. I know many drivers can be replaced with bint gracefully now (e.g. i915 driver). thanks, Takashi > Cc: lm-sensors(a)lm-sensors.org > Cc: linux-rdma(a)vger.kernel.org > Cc: linux-s390(a)vger.kernel.org > Cc: linux-ntfs-dev(a)lists.sourceforge.net > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Rusty Russell <rusty(a)rustcorp.com.au> > --- > drivers/hwmon/emc2103.c | 2 +- > drivers/infiniband/hw/ehca/ehca_main.c | 2 +- > drivers/s390/cio/cmf.c | 2 +- > fs/ntfs/super.c | 2 +- > include/linux/moduleparam.h | 6 ++++++ > kernel/params.c | 24 ++++++++++++++++++++++++ > sound/pci/intel8x0.c | 4 ++-- > 7 files changed, 36 insertions(+), 6 deletions(-) > > diff --git a/drivers/hwmon/emc2103.c b/drivers/hwmon/emc2103.c > --- a/drivers/hwmon/emc2103.c > +++ b/drivers/hwmon/emc2103.c > @@ -55,7 +55,7 @@ static const u8 REG_TEMP_MAX[4] = { 0x34 > * it. Default is to leave the device in the state it's already in (-1). > * This parameter allows APD mode to be optionally forced on or off */ > static int apd = -1; > -module_param(apd, bool, 0); > +module_param(apd, bint, 0); > MODULE_PARM_DESC(init, "Set to zero to disable anti-parallel diode mode"); > > struct temperature { > diff --git a/drivers/infiniband/hw/ehca/ehca_main.c b/drivers/infiniband/hw/ehca/ehca_main.c > --- a/drivers/infiniband/hw/ehca/ehca_main.c > +++ b/drivers/infiniband/hw/ehca/ehca_main.c > @@ -82,7 +82,7 @@ module_param_named(port_act_time, ehca_p > module_param_named(poll_all_eqs, ehca_poll_all_eqs, bool, S_IRUGO); > module_param_named(static_rate, ehca_static_rate, int, S_IRUGO); > module_param_named(scaling_code, ehca_scaling_code, bool, S_IRUGO); > -module_param_named(lock_hcalls, ehca_lock_hcalls, bool, S_IRUGO); > +module_param_named(lock_hcalls, ehca_lock_hcalls, bint, S_IRUGO); > module_param_named(number_of_cqs, ehca_max_cq, int, S_IRUGO); > module_param_named(number_of_qps, ehca_max_qp, int, S_IRUGO); > > diff --git a/drivers/s390/cio/cmf.c b/drivers/s390/cio/cmf.c > --- a/drivers/s390/cio/cmf.c > +++ b/drivers/s390/cio/cmf.c > @@ -98,7 +98,7 @@ enum cmb_format { > * enum cmb_format. > */ > static int format = CMF_AUTODETECT; > -module_param(format, bool, 0444); > +module_param(format, bint, 0444); > > /** > * struct cmb_operations - functions to use depending on cmb_format > diff --git a/fs/ntfs/super.c b/fs/ntfs/super.c > --- a/fs/ntfs/super.c > +++ b/fs/ntfs/super.c > @@ -3198,7 +3198,7 @@ MODULE_DESCRIPTION("NTFS 1.2/3.x driver > MODULE_VERSION(NTFS_VERSION); > MODULE_LICENSE("GPL"); > #ifdef DEBUG > -module_param(debug_msgs, bool, 0); > +module_param(debug_msgs, bint, 0); > MODULE_PARM_DESC(debug_msgs, "Enable debug messages."); > #endif > > diff --git a/include/linux/moduleparam.h b/include/linux/moduleparam.h > --- a/include/linux/moduleparam.h > +++ b/include/linux/moduleparam.h > @@ -367,6 +367,12 @@ extern int param_set_invbool(const char > extern int param_get_invbool(char *buffer, const struct kernel_param *kp); > #define param_check_invbool(name, p) __param_check(name, p, bool) > > +/* An int, which can only be set like a bool (though it shows as an int). */ > +extern struct kernel_param_ops param_ops_bint; > +extern int param_set_bint(const char *val, const struct kernel_param *kp); > +#define param_get_bint param_get_int > +#define param_check_bint param_check_int > + > /** > * module_param_array - a parameter which is an array of some type > * @name: the name of the array variable > diff --git a/kernel/params.c b/kernel/params.c > --- a/kernel/params.c > +++ b/kernel/params.c > @@ -369,6 +369,30 @@ struct kernel_param_ops param_ops_invboo > }; > EXPORT_SYMBOL(param_ops_invbool); > > +int param_set_bint(const char *val, const struct kernel_param *kp) > +{ > + struct kernel_param boolkp; > + bool v; > + int ret; > + > + /* Match bool exactly, by re-using it. */ > + boolkp = *kp; > + boolkp.arg = &v; > + boolkp.flags |= KPARAM_ISBOOL; > + > + ret = param_set_bool(val, &boolkp); > + if (ret == 0) > + *(int *)kp->arg = v; > + return ret; > +} > +EXPORT_SYMBOL(param_set_bint); > + > +struct kernel_param_ops param_ops_bint = { > + .set = param_set_bint, > + .get = param_get_int, > +}; > +EXPORT_SYMBOL(param_ops_bint); > + > /* We break the rule and mangle the string. */ > static int param_array(const char *name, > const char *val, > diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c > --- a/sound/pci/intel8x0.c > +++ b/sound/pci/intel8x0.c > @@ -95,13 +95,13 @@ module_param(ac97_quirk, charp, 0444); > MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware."); > module_param(buggy_semaphore, bool, 0444); > MODULE_PARM_DESC(buggy_semaphore, "Enable workaround for hardwares with problematic codec semaphores."); > -module_param(buggy_irq, bool, 0444); > +module_param(buggy_irq, bint, 0444); > MODULE_PARM_DESC(buggy_irq, "Enable workaround for buggy interrupts on some motherboards."); > module_param(xbox, bool, 0444); > MODULE_PARM_DESC(xbox, "Set to 1 for Xbox, if you have problems with the AC'97 codec detection."); > module_param(spdif_aclink, int, 0444); > MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link."); > -module_param(inside_vm, bool, 0444); > +module_param(inside_vm, bint, 0444); > MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization."); > > /* just for backward compatibility */ >

1 0

[alsa-devel] [PATCH] ASoC: Staticise mfld_msic_dailink
by Axel Lin 15 Dec '11

15 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/mid-x86/mfld_machine.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index e53f8e4..8ae0574 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -281,7 +281,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } -struct snd_soc_dai_link mfld_msic_dailink[] = { +static struct snd_soc_dai_link mfld_msic_dailink[] = { { .name = "Medfield Headset", .stream_name = "Headset", -- 1.7.5.4

3 2

Re: [alsa-devel] bugs in "ASoC: core - Optimise and refactor pcm_new() to pass only rtd"?
by Mark Brown 14 Dec '11

14 Dec '11

On Wed, Dec 14, 2011 at 12:43:50PM +0000, James Hogan wrote: > I've bisected a problem with an out of tree set of asoc drivers down to commit > 552d1ef6b5a98d7b95959d5b139071e3c90cebf1 ("ASoC: core - Optimise and refactor > pcm_new() to pass only rtd"). By the looks of it this change was intended not > to change the workings of the code, however the dai parameter of pcm_new used > to be passed codec_dai in soc_new_pcm() (initialised from rtd->codec_dai), but > each driver now sets it's dai to rtd->cpu_dai instead. This seems to make the > driver check the wrong dai's driver->capture.channels_min in the pcm_new > callback, which makes it dereference > pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream in the driver's > preallocate_dma_buffer() function which can be NULL. Is this a bug (in every > pcm asoc driver) that just happens not to have been noticed, or have I > misunderstood it? Your drivers should never have beenn relying on this behaviour of the core - see changes in the pxa and fsl drivers which remove similar assumptions. The CPU driver should be checking for the substream if it's looking to see if a particular substream is present, not peering inside the DAI for a random parameter.

1 0

[alsa-devel] [PATCH] ASoC: Tune the accessory detection rates for WM8996
by Mark Brown 14 Dec '11

14 Dec '11

Use longer intervals when the microphone is not inserted to increase robustness against leisurely insertion. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 18 ++++++++++++------ 1 files changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8f88f5a..da7acae 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2572,8 +2572,10 @@ static void wm8996_micd(struct snd_soc_codec *codec) SND_JACK_BTN_0); snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - WM8996_MICD_RATE_MASK); + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + WM8996_MICD_RATE_MASK | + 9 << WM8996_MICD_BIAS_STARTTIME_SHIFT); return; } @@ -2590,8 +2592,10 @@ static void wm8996_micd(struct snd_soc_codec *codec) /* Increase poll rate to give better responsiveness * for buttons */ snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 5 << WM8996_MICD_RATE_SHIFT); + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + 5 << WM8996_MICD_RATE_SHIFT | + 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); } else { dev_dbg(codec->dev, "Mic button up\n"); snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0); @@ -2639,8 +2643,10 @@ static void wm8996_micd(struct snd_soc_codec *codec) * responsiveness. */ snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 7 << WM8996_MICD_RATE_SHIFT); + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + 7 << WM8996_MICD_RATE_SHIFT | + 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); } } } -- 1.7.7.3

1 0

Re: [alsa-devel] MAYA44 USB audio device no inputs control
by Kamil Sobolak 14 Dec '11

14 Dec '11

Hi! I attached two logs in zip folder. Please check this. Thanks! Kamil Sobolak

2 1

[alsa-devel] [PATCH 1/8] ASoC: Remove cache default for volatile wm9081 reset register
by Mark Brown 14 Dec '11

14 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 1f2672b..a6bab39 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -31,7 +31,6 @@ #include "wm9081.h" static struct reg_default wm9081_reg[] = { - { 0, 0x9081 }, /* R0 - Software Reset */ { 2, 0x00B9 }, /* R2 - Analogue Lineout */ { 3, 0x00B9 }, /* R3 - Analogue Speaker PGA */ { 4, 0x0001 }, /* R4 - VMID Control */ -- 1.7.7.3

1 7

Re: [alsa-devel] [PATCH 5/5] ASoC: mxs: correct 'direction' of device_prep_dma_cyclic
by Dong Aisheng-B29396 14 Dec '11

14 Dec '11

> -----Original Message----- > From: linux-kernel-owner(a)vger.kernel.org [mailto:linux-kernel- > owner(a)vger.kernel.org] On Behalf Of Shawn Guo > Sent: Wednesday, December 14, 2011 3:48 PM > To: Liam Girdwood; Mark Brown > Cc: Dong Aisheng; alsa-devel(a)alsa-project.org; linux- > kernel(a)vger.kernel.org; linux-arm-kernel(a)lists.infradead.org; Shawn Guo > Subject: [PATCH 5/5] ASoC: mxs: correct 'direction' of > device_prep_dma_cyclic > > The commit 49920bc (dmaengine: add new enum dma_transfer_direction) > changes the type of parameter 'direction' of device_prep_dma_cyclic from > dma_data_direction to dma_transfer_direction. > > Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> > --- > sound/soc/mxs/mxs-pcm.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Regards Dong Aisheng

1 0

[alsa-devel] [PATCH] ALSA: hda: remove unused quirk for inverted mute led
by Gustavo Maciel Dias Vieira 14 Dec '11

14 Dec '11

Commit b99a776d0b17ae0f3a54e86009887a00ac4889d0 removed all effects of the STAC92HD83* model quirk "hp". However, it left the model selection and documentation behind, confusing users with inverted mute leds. Completely remove this quirk and its documentation. Signed-off-by: Gustavo Maciel Dias Vieira <gustavo(a)sagui.org> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 - sound/pci/hda/patch_sigmatel.c | 4 ---- 2 files changed, 0 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index bda5469..c8c5454 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -338,7 +338,6 @@ STAC92HD83* mic-ref Reference board with power management for ports dell-s14 Dell laptop dell-vostro-3500 Dell Vostro 3500 laptop - hp HP laptops with (inverted) mute-LED hp-dv7-4000 HP dv-7 4000 auto BIOS setup (default) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d8d2f9d..2034f33 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -96,7 +96,6 @@ enum { STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, STAC_DELL_VOSTRO_3500, - STAC_92HD83XXX_HP, STAC_92HD83XXX_HP_cNB11_INTQUAD, STAC_HP_DV7_4000, STAC_92HD83XXX_MODELS @@ -1691,7 +1690,6 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", [STAC_DELL_VOSTRO_3500] = "dell-vostro-3500", - [STAC_92HD83XXX_HP] = "hp", [STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad", [STAC_HP_DV7_4000] = "hp-dv7-4000", }; @@ -1706,8 +1704,6 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "unknown Dell", STAC_DELL_S14), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x1028, "Dell Vostro 3500", STAC_DELL_VOSTRO_3500), - SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600, - "HP", STAC_92HD83XXX_HP), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1657, -- 1.7.7.3

2 1

[alsa-devel] Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again)
by Colomban Wendling 14 Dec '11

14 Dec '11

Hi, Back to 3.0, the sound on my soundcard [1] started to be choppy, I reported it [2] and it got fixed (thanks to Takashi Iwai!). However, the story repeated with 3.1 (and probably 3.0.8 or before [3]): I've got similar choppy sound again. I bisected the few commits that happened on sound/pci/hda/patch_realtek.c, and finally found the villain: "8974bd51 ALSA: hda/realtek - Fix auto-mute with HP+LO configuration" Reverting it from v3.1 fixes the problem. master (94956ee) also suffers of the problem, but I couldn't test without that commit because reverting it fails, too much changes happened. Thanks for your attention, and if you need any extra information (although many are already available in my previous report [2]) or testing, please let me know. Regards, Colomban [1] Realtek ALC889 on an MSI H55M-E33 [2] bugs.debian.org/cgi-bin/bugreport.cgi?bug=635362, http://mailman.alsa-project.org/pipermail/alsa-devel/2011-July/042250.html (though I can't find my original message here) [3] I first saw the problem in Debian's 3.0.0-6 which includes 3.0 up to .8 -- but the debugging I did was on vanilla kernel from git://github.com/torvalds/linux.git.

2 4

Re: [alsa-devel] [PATCH] ALSA HDA: Add support for new AMD products
by Takashi Iwai 14 Dec '11

14 Dec '11

At Wed, 14 Dec 2011 16:10:27 +0800, Andiry Xu wrote: > > This patch adds HDMI audio support for new AMD products. As HW default > disable snoop, force non-snoop mode in HD audio driver. > > Signed-off-by: Andiry Xu <andiry.xu(a)amd.com> Thanks, applied now. Also I committed a clean up patch to merge both VIA and AMD-HDMI non-snoop check in a single place. Takashi > --- > sound/pci/hda/hda_intel.c | 13 +++++++++++++ > 1 files changed, 13 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index d1582dd..9f7c901a 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -487,6 +487,7 @@ enum { > AZX_DRIVER_SCH, > AZX_DRIVER_ATI, > AZX_DRIVER_ATIHDMI, > + AZX_DRIVER_ATIHDMI_NS, > AZX_DRIVER_VIA, > AZX_DRIVER_SIS, > AZX_DRIVER_ULI, > @@ -533,6 +534,7 @@ static char *driver_short_names[] __devinitdata = { > [AZX_DRIVER_SCH] = "HDA Intel MID", > [AZX_DRIVER_ATI] = "HDA ATI SB", > [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", > + [AZX_DRIVER_ATIHDMI_NS] = "HDA ATI HDMI", > [AZX_DRIVER_VIA] = "HDA VIA VT82xx", > [AZX_DRIVER_SIS] = "HDA SIS966", > [AZX_DRIVER_ULI] = "HDA ULI M5461", > @@ -2678,6 +2680,8 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, > > chip->single_cmd = single_cmd; > chip->snoop = hda_snoop; > + if (chip->driver_type == AZX_DRIVER_ATIHDMI_NS) > + chip->snoop = 0; > > if (bdl_pos_adj[dev] < 0) { > switch (chip->driver_type) { > @@ -2776,6 +2780,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, > chip->capture_streams = ULI_NUM_CAPTURE; > break; > case AZX_DRIVER_ATIHDMI: > + case AZX_DRIVER_ATIHDMI_NS: > chip->playback_streams = ATIHDMI_NUM_PLAYBACK; > chip->capture_streams = ATIHDMI_NUM_CAPTURE; > break; > @@ -3037,6 +3042,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > { PCI_DEVICE(0x1002, 0xaa48), > .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0x9902), > + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0xaaa0), > + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0xaaa8), > + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0xaab0), > + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, > /* VIA VT8251/VT8237A */ > { PCI_DEVICE(0x1106, 0x3288), > .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, > -- > 1.7.4.1 > >

1 0

[alsa-devel] start_threshold and stream controls for PCM streams
by Nallasellan, Singaravelan 14 Dec '11

14 Dec '11

Hi, Would like to know how start_threshold is expected to be used in single_threaded audio player? I assume that in blocking implementation, the write call will block till the ring buffer becomes empty. Is it right? If the audio player has to wait for the write call to be blocked, I guess it is possible that the delay to service the stream controls will be delayed as long as the buffer duration. If so, how can I send stream control operations for the stream when the stream is blocked for the buffer? What are the option to service stream controls immediately? Thanks in advance. Thanks -Sing

3 2

[alsa-devel] [PATCH] ALSA: HDA: Use LPIB Position fix for Intel SCH Poulsbo
by David Henningsson 14 Dec '11

14 Dec '11

Hi Takashi, I just got another position_fix=1 report for the same vendor/deviceID [1] as I sent you the other day, and I assume Eric Piel (a month ago on this list) have the same chipset as well, although I cannot see his alsa-info. I think you have it, can you verify? So...can we consider quirking the entire chip to use LPIB, like in the attached patch (untested)? If so I'll send another patch later to remove existing quirks for this chipset. // David [1] http://www.alsa-project.org/db/?f=b7421521b8d8a4db3842ee14cdb54b4f440cae5e

2 1

[alsa-devel] [PATCH] ASoC: Remove unused extern declarations for sh4_hac_dai and sh7760_soc_platform
by Axel Lin 14 Dec '11

14 Dec '11

Both sh4_hac_dai and sh7760_soc_platform are changed to static by multi-component patch and they are not used in sh7760-ac97.c now. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/sh/sh7760-ac97.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index c62ae68..df651e8 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -16,10 +16,6 @@ #define IPSEL 0xFE400034 -/* platform specific structs can be declared here */ -extern struct snd_soc_dai_driver sh4_hac_dai[2]; -extern struct snd_soc_platform_driver sh7760_soc_platform; - static struct snd_soc_dai_link sh7760_ac97_dai = { .name = "AC97", .stream_name = "AC97 HiFi", -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Staticise sst_pcm_new and sst_soc_platform_drv
by Axel Lin 14 Dec '11

14 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/mid-x86/sst_platform.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 24f9471..5b936d5 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -443,7 +443,7 @@ static void sst_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) +static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; @@ -463,7 +463,7 @@ int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) } return retval; } -struct snd_soc_platform_driver sst_soc_platform_drv = { +static struct snd_soc_platform_driver sst_soc_platform_drv = { .ops = &sst_platform_ops, .pcm_new = sst_pcm_new, .pcm_free = sst_pcm_free, -- 1.7.5.4

3 2

[PATCH] ALSA: HDA: Use LPIB Position fix for Intel SCH Poulsbo
by David Henningsson 14 Dec '11

14 Dec '11

Several people with this chipset have reported inconsistent/sloppy values for position reporting when the DMA position buffer is used, and that setting position_fix=1 have fixed their problems. BugLink: https://bugs.launchpad.net/bugs/825709 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0746ab4..f19d7fa 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2971,7 +2971,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */ + /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_BUFSIZE }, /* ICH6 */ -- 1.7.5.4 --------------050108040603000000050605--

1 0

[alsa-devel] [PATCH] ASoC: Complete initialisation before registering Samsung PCM DAI
by Mark Brown 14 Dec '11

14 Dec '11

Otherwise there's a race where the DAI might get used without everything having been set up. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/pcm.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index beef63f..3a29c26 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -570,12 +570,6 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) } clk_enable(pcm->pclk); - ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); - if (ret != 0) { - dev_err(&pdev->dev, "failed to get pcm_clock\n"); - goto err5; - } - s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + S3C_PCM_RXFIFO; s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start @@ -587,6 +581,12 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); + if (ret != 0) { + dev_err(&pdev->dev, "failed to get register DAI: %d\n", ret); + goto err5; + } + return 0; err5: -- 1.7.7.3

2 1

Re: [alsa-devel] _mmap_playback_avail() short (was: Default device XRUNs with perriods=2)
by Raymond Yau 14 Dec '11

14 Dec '11

>> Do you require the sound card give you accurate playback position ? >No. Just accurate playback (or capture). But reliably, on any system, >without hardware-specific tweaks, 'out-of-the-box'. Ideally with the >potential for reasonable latency if the user requests that. And not >necessarily only on the 'latest' Alsa versions. at 48000Hz, the hwptr of snd-ymfpci is increase to next boundary (256 frames). so you cannot get accurate playback position. this situation can be emulated with snd-hda-intel by masking the value return by pointer callback with 0xffffff00 >> snd-ymfpci only update hwptr in 5.3ms intervals >> >> it seem that snd-usb-audio update hwptr in 1ms >> >> > It seems the options are:- >> > >> > 1. Use 3 periods and accept worse latency, always filling with >> > avail_frames >> >> snd-ymfpci has periods_min 3 >> and there are sound cards with periods_max = 2 >Yes, this is the sort of thing that makes it more difficult to create a good >workaround. seem only two drivers have this limitation , so you may ask the author of those drivers whether it is feasible to change periods_max to a higher value drivers/dummy.c: .periods_max = 2, soc/samsung/idma.c: .periods_max = 2,

1 0

[alsa-devel] [PATCH] ASoC: Staticise asoc_idma_platform
by Mark Brown 14 Dec '11

14 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/idma.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index baf97eb..2bcf758 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -410,7 +410,7 @@ void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) idma.lp_tx_addr = addr; } -struct snd_soc_platform_driver asoc_idma_platform = { +static struct snd_soc_platform_driver asoc_idma_platform = { .ops = &idma_ops, .pcm_new = idma_new, .pcm_free = idma_free, -- 1.7.7.3

2 1

[alsa-devel] [PATCH v2] ASoC: Tegra machine ASoC driver for boards using ALC5332 codec
by Leon Romanovsky 13 Dec '11

13 Dec '11

At this stage only Toshiba AC100/Dynabook supported. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> -- Changes since v1: * Use module_platform_driver. * Move controls, dapm_widgets and dapm_routes to snd_soc_card structure. * Set dai_fmt in snd_soc_dai_link structure instead probe function. * Remove check board in probe function * Fix warning about unused codec_dai. * Use devm_kzalloc to manage memory allocation. --- sound/soc/tegra/Kconfig | 9 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_alc5632.c | 233 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 244 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/tegra_alc5632.c diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index c6af1fd..ce1b773 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -47,3 +47,12 @@ config SND_SOC_TEGRA_TRIMSLICE help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. + +config SND_SOC_TEGRA_ALC5632 + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 4d943b3..8e584b8 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -14,6 +14,8 @@ obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o +snd-soc-tegra-alc5632-objs := tegra_alc5632.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o +obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c new file mode 100644 index 0000000..89697c6 --- /dev/null +++ b/sound/soc/tegra/tegra_alc5632.c @@ -0,0 +1,233 @@ +/* +* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Marc Dietrich <marvin24(a)gmx.de> +* +* 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. +*/ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/alc5632.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-alc5632" + +struct tegra_alc5632 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_alc5632_asoc_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->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + switch (srate) { + case 64000: + case 88200: + case 96000: + mclk = 128 * srate; + break; + default: + mclk = 512 * srate; + break; + } + /* FIXME: Codec only requires >= 3MHz if OSR==0 */ + while (mclk < 6000000) + mclk *= 2; + + err = tegra_asoc_utils_set_rate(&alc5632->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_alc5632_asoc_ops = { + .hw_params = tegra_alc5632_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_alc5632_hs_jack; + +static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_alc5632_hs_jack_gpio = { + .name = "Headset Detect", + .report = SND_JACK_HEADSET, + .debounce_time = 150, +}; + +static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Int Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = { + /* Internal Speaker */ + {"Int Spk", NULL, "SPKOUT"}, + {"Int Spk", NULL, "SPKOUTN"}, + + /* Headset Mic */ + {"MIC1", NULL, "MICBIAS1"}, + {"MICBIAS1", NULL, "Headset Mic"}, + + /* Headset Stereophone */ + {"Headset Stereophone", NULL, "HPR"}, + {"Headset Stereophone", NULL, "HPL"}, +}; + +static const struct snd_kcontrol_new tegra_alc5632_controls[] = { + SOC_DAPM_PIN_SWITCH("Int Spk"), +}; + +static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack); + snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins), + tegra_alc5632_hs_jack_pins); + snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, + 1, &tegra_alc5632_hs_jack_gpio); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static struct snd_soc_dai_link tegra_alc5632_dai = { + .name = "ALC5632", + .stream_name = "ALC5632 PCM", + .codec_name = "alc5632.0-001e", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "alc5632-hifi", + .init = tegra_alc5632_asoc_init, + .ops = &tegra_alc5632_asoc_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_alc5632 = { + .name = "tegra-alc5632", + .dai_link = &tegra_alc5632_dai, + .num_links = 1, + .controls = tegra_alc5632_controls, + .num_controls = ARRAY_SIZE(tegra_alc5632_controls), + .dapm_widgets = tegra_alc5632_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_alc5632_dapm_widgets), + .dapm_routes = tegra_alc5632_audio_map, + .num_dapm_routes = ARRAY_SIZE(tegra_alc5632_audio_map), + .fully_routed = true, +}; + +static __devinit int tegra_alc5632_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_alc5632; + struct tegra_alc5632 *alc5632; + int ret; + + alc5632 = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_alc5632), GFP_KERNEL); + if (!alc5632) { + dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + return ret; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, alc5632); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + tegra_asoc_utils_fini(&alc5632->util_data); + return ret; + } + + return 0; +} + +static int __devexit tegra_alc5632_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&alc5632->util_data); + + return 0; +} + +static struct platform_driver tegra_alc5632_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tegra_alc5632_probe, + .remove = __devexit_p(tegra_alc5632_remove), +}; +module_platform_driver(tegra_alc5632_driver); + +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.3.4

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Re: [alsa-devel] [PATCH v4 4/6] core: add API header and driver header files
by Nallasellan, Singaravelan 13 Dec '11

13 Dec '11

> +struct snd_compr_runtime { > + snd_pcm_state_t state; > + struct snd_compr_ops *ops; > + void *buffer; Can we define buffer as char *? > + size_t buffer_size; > + size_t fragment_size; Can we define buffer_size and fragment_size as unsigned items? Can we have negative size here? > + unsigned int fragments; > + size_t hw_pointer; > + size_t app_pointer; > + size_t bytes_written; > + size_t bytes_copied; > + wait_queue_head_t sleep; > +}; > +

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[alsa-devel] [PATCH v2] ALSA: hda: fix mute led polarity for HP laptops with buggy BIOS
by Gustavo Maciel Dias Vieira 13 Dec '11

13 Dec '11

Some HP laptop models do not have a properly filled OEM string used to set the gpio and polarity of the mute led. Make the mute led configuration work for this case. Signed-off-by: Gustavo Maciel Dias Vieira <gustavo(a)sagui.org> --- Em Seg, 2011-12-12 às 17:10 +0100, Takashi Iwai escreveu: > > Yes, subsytem_id can be used. OTOH, I guess HP_Mute_LED_P_G check > would be easier and may cover more machines. So, I prefer the > latter. > Is this ok? Instead of setting polarity to 1, I can select it by subsystem_id list. Abraços, Gustavo sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index eeb25d52..616678f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4929,6 +4929,12 @@ static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity) set_hp_led_gpio(codec); return 1; } + /* BIOS bug: unfilled OEM string */ + if (strstr(dev->name, "HP_Mute_LED_P_G")) { + set_hp_led_gpio(codec); + spec->gpio_led_polarity = 1; + return 1; + } } /* -- 1.7.7.3

2 1

Re: [alsa-devel] ASoC updates for 3.3
by Takashi Iwai 13 Dec '11

13 Dec '11

At Tue, 13 Dec 2011 15:12:54 +0800, Mark Brown wrote: > > The following changes since commit 102477b6380978b1532e5d1f769155026a332aff: > > Merge branch 'for-3.2' into for-3.3 (2011-12-06 10:09:55 +0000) > > are available in the git repository at: > > git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 > > There's a merge up of 3.2-rc3 in here so the diffstat and log are bigger > than they might be otherwise - this is due to the cross tree merges with > the WM8994 MFD side which allow us to eliminate the ASoC version of the > WM8994 register access and cache code, using the much more efficient > regmap version instead. Thanks, pulled now. Though, I've got a few new compile warnings. sound/soc/codecs/wm8903.c:2036:2: warning: initialization discards ‘const’ qualifier from pointer target type sound/soc/codecs/wm9090.c:669:22: warning: unused variable ‘wm9090’ Takashi

2 1

[alsa-devel] [PATCH] ASoC: OMAP: Kconfig string cleanup
by Peter Ujfalusi 13 Dec '11

13 Dec '11

User visible string cleanup in the config file. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/Kconfig | 30 +++++++++++++++--------------- 1 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index fb1bf25..9329810 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,5 +1,5 @@ config SND_OMAP_SOC - tristate "SoC Audio for the Texas Instruments OMAP chips" + tristate "SoC Audio for Texas Instruments OMAP based boards" depends on ARCH_OMAP config SND_OMAP_SOC_DMIC @@ -16,7 +16,7 @@ config SND_OMAP_SOC_HDMI tristate config SND_OMAP_SOC_N810 - tristate "SoC Audio support for Nokia N810" + tristate "Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C depends on OMAP_MUX select SND_OMAP_SOC_MCBSP @@ -25,7 +25,7 @@ config SND_OMAP_SOC_N810 Say Y if you want to add support for SoC audio on Nokia N810. config SND_OMAP_SOC_RX51 - tristate "SoC Audio support for Nokia RX-51" + tristate "Audio support for Nokia RX-51" depends on SND_OMAP_SOC && MACH_NOKIA_RX51 select OMAP_MCBSP select SND_OMAP_SOC_MCBSP @@ -36,7 +36,7 @@ config SND_OMAP_SOC_RX51 hardware. This is also known as Nokia N900 product. config SND_OMAP_SOC_AMS_DELTA - tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + tristate "Audio support for Amstrad E3 (Delta) videophone" depends on SND_OMAP_SOC && MACH_AMS_DELTA select SND_OMAP_SOC_MCBSP select SND_SOC_CX20442 @@ -55,7 +55,7 @@ config SND_OMAP_SOC_AMS_DELTA RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" config SND_OMAP_SOC_OSK5912 - tristate "SoC Audio support for omap osk5912" + tristate "Audio support for omap osk5912" depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC23 @@ -63,7 +63,7 @@ config SND_OMAP_SOC_OSK5912 Say Y if you want to add support for SoC audio on osk5912. config SND_OMAP_SOC_OVERO - tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35" + tristate "Audio support for Gumstix Overo and CompuLab CM-T35" depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -72,7 +72,7 @@ config SND_OMAP_SOC_OVERO Gumstix Overo or CompuLab CM-T35 config SND_OMAP_SOC_OMAP3EVM - tristate "SoC Audio support for OMAP3EVM board" + tristate "Audio support for OMAP3EVM board" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -80,7 +80,7 @@ config SND_OMAP_SOC_OMAP3EVM Say Y if you want to add support for SoC audio on the omap3evm board. config SND_OMAP_SOC_AM3517EVM - tristate "SoC Audio support for OMAP3517 / AM3517 EVM" + tristate "Audio support for OMAP3517 / AM3517 EVM" depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC23 @@ -89,7 +89,7 @@ config SND_OMAP_SOC_AM3517EVM EVM. config SND_OMAP_SOC_SDP3430 - tristate "SoC Audio support for Texas Instruments SDP3430" + tristate "Audio support for Texas Instruments SDP3430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -98,7 +98,7 @@ config SND_OMAP_SOC_SDP3430 SDP3430. config SND_OMAP_SOC_SDP4430 - tristate "SoC Audio support for Texas Instruments SDP4430" + tristate "Audio support for Texas Instruments SDP4430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM @@ -109,7 +109,7 @@ config SND_OMAP_SOC_SDP4430 SDP4430. config SND_OMAP_SOC_OMAP4_HDMI - tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" + tristate "Audio support for Texas Instruments OMAP4 HDMI" depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4 select SND_OMAP_SOC_HDMI help @@ -117,7 +117,7 @@ config SND_OMAP_SOC_OMAP4_HDMI OMAP4 chips config SND_OMAP_SOC_OMAP3_PANDORA - tristate "SoC Audio support for OMAP3 Pandora" + tristate "Audio support for OMAP3 Pandora" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -125,7 +125,7 @@ config SND_OMAP_SOC_OMAP3_PANDORA Say Y if you want to add support for SoC audio on the OMAP3 Pandora. config SND_OMAP_SOC_OMAP3_BEAGLE - tristate "SoC Audio support for OMAP3 Beagle and Devkit8000" + tristate "Audio support for OMAP3 Beagle and Devkit8000" depends on TWL4030_CORE && SND_OMAP_SOC depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000) select SND_OMAP_SOC_MCBSP @@ -135,7 +135,7 @@ config SND_OMAP_SOC_OMAP3_BEAGLE the clone Devkit8000. config SND_OMAP_SOC_ZOOM2 - tristate "SoC Audio support for Zoom2" + tristate "Audio support for Zoom2" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 @@ -143,7 +143,7 @@ config SND_OMAP_SOC_ZOOM2 Say Y if you want to add support for Soc audio on Zoom2 board. config SND_OMAP_SOC_IGEP0020 - tristate "SoC Audio support for IGEP v2" + tristate "Audio support for IGEP v2" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020 select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 -- 1.7.8

2 2

Re: [alsa-devel] [PATCH] ALSA HDA: add support for AMD SI audio
by Takashi Iwai 13 Dec '11

13 Dec '11

At Tue, 13 Dec 2011 16:42:12 +0800, Andiry Xu wrote: > > This patch adds support for AMD Southern Island HD audio. Could you give some more background for this patch? That is, it's enabling non-snooping mode for which reason, and why new driver_type is needed, etc. thanks, Takashi > > Signed-off-by: Andiry Xu <andiry.xu(a)amd.com> > --- > sound/pci/hda/hda_intel.c | 14 ++++++++++++++ > 1 files changed, 14 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index d1582dd..ae6b68c 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -487,6 +487,7 @@ enum { > AZX_DRIVER_SCH, > AZX_DRIVER_ATI, > AZX_DRIVER_ATIHDMI, > + AZX_DRIVER_SI_ATIHDMI, > AZX_DRIVER_VIA, > AZX_DRIVER_SIS, > AZX_DRIVER_ULI, > @@ -533,6 +534,7 @@ static char *driver_short_names[] __devinitdata = { > [AZX_DRIVER_SCH] = "HDA Intel MID", > [AZX_DRIVER_ATI] = "HDA ATI SB", > [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", > + [AZX_DRIVER_SI_ATIHDMI] = "HDA ATI SI HDMI", > [AZX_DRIVER_VIA] = "HDA VIA VT82xx", > [AZX_DRIVER_SIS] = "HDA SIS966", > [AZX_DRIVER_ULI] = "HDA ULI M5461", > @@ -2678,6 +2680,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, > > chip->single_cmd = single_cmd; > chip->snoop = hda_snoop; > + /* Default non-snoop mode for ATI SI HDMI AUDIO */ > + if (chip->driver_type == AZX_DRIVER_SI_ATIHDMI) > + chip->snoop = 0; > > if (bdl_pos_adj[dev] < 0) { > switch (chip->driver_type) { > @@ -2776,6 +2781,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, > chip->capture_streams = ULI_NUM_CAPTURE; > break; > case AZX_DRIVER_ATIHDMI: > + case AZX_DRIVER_SI_ATIHDMI: > chip->playback_streams = ATIHDMI_NUM_PLAYBACK; > chip->capture_streams = ATIHDMI_NUM_CAPTURE; > break; > @@ -3037,6 +3043,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > { PCI_DEVICE(0x1002, 0xaa48), > .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0x9902), > + .driver_data = AZX_DRIVER_SI_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0xaaa0), > + .driver_data = AZX_DRIVER_SI_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0xaaa8), > + .driver_data = AZX_DRIVER_SI_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > + { PCI_DEVICE(0x1002, 0xaab0), > + .driver_data = AZX_DRIVER_SI_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, > /* VIA VT8251/VT8237A */ > { PCI_DEVICE(0x1106, 0x3288), > .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, > -- > 1.7.4.1 > >

1 0

[alsa-devel] ASoC updates for 3.2
by Mark Brown 13 Dec '11

13 Dec '11

The following changes since commit cefcc03ffc9527dde56807339edb1719c8dbae5f: ASoC: Provide a more complete DMA driver stub (2011-12-06 10:05:07 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.2 The usual story - nothing major fixed here but all small, useful fixes. Axel Lin (5): ASoC: Include linux/io.h for jz4740 codec ASoC: Fix build dependency for SND_SOC_JZ4740_CODEC ASoC: Fix a typo in jive_wm8750 ASoC: Include linux/module.h for smdk2443_wm9710 ASoC: Fix hx4700 error handling to free gpios if snd_soc_register_card fails Jonathan Neuschäfer (1): ASoC: WM8958: correctly show firmware magic on mismatch Lothar Waßmann (2): ASoC: mxs: Add missing MODULE_LICENSE("GPL") ASoC: mxs: Add appropriate MODULE_ALIAS() Mark Brown (1): ASoC: Fix WM8996 24.576MHz clock operation sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/jz4740.c | 1 + sound/soc/codecs/wm8958-dsp2.c | 2 ++ sound/soc/codecs/wm8996.c | 1 + sound/soc/mxs/mxs-pcm.c | 3 +++ sound/soc/mxs/mxs-sgtl5000.c | 1 + sound/soc/pxa/hx4700.c | 5 +++-- sound/soc/samsung/jive_wm8750.c | 3 +-- sound/soc/samsung/smdk2443_wm9710.c | 1 + 9 files changed, 14 insertions(+), 5 deletions(-)

2 2

Re: [alsa-devel] [PATCH 06/19] mx27vis-aic32x4: only register when running on the right machine
by Mark Brown 13 Dec '11

13 Dec '11

On Tue, Dec 13, 2011 at 07:31:38AM +0100, Eric Bénard wrote: > without this fix, a kernel compiled with imx_v4_v5_defconfig > and running on a cpuimx25sd leads to : I'll apply this but a better fix is to convert the driver to be a normal platform driver and only register the device on the appropraite boards.

1 0

[alsa-devel] [PATCH] ASoC: Rename ALC5632 MICBIAS to common name convention.
by Leon Romanovsky 13 Dec '11

13 Dec '11

Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/codecs/alc5632.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 08613c7..390e437 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -379,8 +379,8 @@ SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0), SND_SOC_DAPM_PGA_E("D Amp", ALC5632_PWR_MANAG_ADD2, 14, 0, NULL, 0, amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), -- 1.7.3.4

2 3

Re: [alsa-devel] [PATCH 3/3] sound/soc/lapis: add platform driver
by Mark Brown 13 Dec '11

13 Dec '11

On Mon, Nov 21, 2011 at 01:08:52PM +0900, Tomoya MORINAGA wrote: > +/* =================== I2S CH0 config =================== */ > +#define I2S_CH0_MCLK (12288000) /* Master Clock Frequency[Hz] */ This looks like it should be board specific? > + #if IOH_I2S_USE_PARAM > + #define I2S_CH0_FS_16000 16000 > + #define I2S_CH0_FS_32000 32000 > + #define I2S_CH0_FS_48000 48000 > + #else > + #define I2S_CH0_FS_8000 8000 > + #define I2S_CH0_FS_11025 11025 > + #define I2S_CH0_FS_22050 22050 > + #define I2S_CH0_FS_44100 44100 > + #endif What are these ifdefs for? This should be configured at runtime. > +/* select master or slave. The value is ioh_mssel_t */ > +#define I2S_CH0_MSSEL (ioh_mssel_master) > +/* select MCLK or MLBCLK into Master Clock. The value is enum ioh_masterclk_t */ > +#define I2S_CH0_MASTERCLKSEL (ioh_masterclksel_mclk) This and most of the following defines also look like compile time configuration for things that should be runtime configured. As with the previous patches in the series the big picture issue here is that you need to update your code to reflect the practices of modern mainline Linux code - here the major issue is that you appear to be doing all your configuration with #defines rather than by using runtime configuration from the machine driver as is normal for ASoC code. I got through a bunch more defines but stopped reviewing at some point as there was so much of this issue.

2 4

[alsa-devel] [PATCH 1/3] ASoC: Staticise rx51_aux_dev
by Axel Lin 13 Dec '11

13 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/omap/rx51.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 4cabb74..ad16db5 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -365,7 +365,7 @@ static struct snd_soc_dai_link rx51_dai[] = { }, }; -struct snd_soc_aux_dev rx51_aux_dev[] = { +static struct snd_soc_aux_dev rx51_aux_dev[] = { { .name = "TLV320AIC34b", .codec_name = "tlv320aic3x-codec.2-0019", -- 1.7.5.4

2 3

[alsa-devel] [PATCH v2 1/6] ASoC: Allow device tree to specify a card's name
by Stephen Warren 13 Dec '11

13 Dec '11

If a card's device was instantiated from device tree, and the device tree has a "user-visible-name" property, use that as the card's name. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: New patch implementing new functionality Re: the binding documentation: * "SoC" here refers to the fact this is a binding oriented at System-on- chip audio complexes, rather than having to do with "ASoC"; both names were derived from the same root. * Do we need a compatible property for this "base class" binding at all? I think it's a good idea, even though the code doesn't actually rely on it. * Should the vendor field in the compatible property be "generic", "linux", or absent? I've tried to make these bindings generic and applicable to other OSs, so "linux," seems wrong. * Should the property "user-visible-name" have a "generic," prefix or similar? .../bindings/sound/soc-audio-complex.txt | 14 ++++++++ sound/soc/soc-core.c | 35 ++++++++++++++++++- 2 files changed, 47 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/soc-audio-complex.txt diff --git a/Documentation/devicetree/bindings/sound/soc-audio-complex.txt b/Documentation/devicetree/bindings/sound/soc-audio-complex.txt new file mode 100644 index 0000000..1cc7059 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/soc-audio-complex.txt @@ -0,0 +1,14 @@ +SoC Audio Complex + +Required properties: +- compatible : "generic,soc-audio-complex". Other compatible values + indicating the specific HW model will typically be present too. +- user-visible-name : The user-visible name of this sound complex. + +Example: + +sound { + compatible = "nvidia,tegra-audio-wm8903", + "generic,soc-audio-complex"; + user-visible-name = "tegra-wm8903-harmony"; +}; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5195f06..56d1bc5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -32,6 +32,7 @@ #include <linux/platform_device.h> #include <linux/ctype.h> #include <linux/slab.h> +#include <linux/of.h> #include <sound/ac97_codec.h> #include <sound/core.h> #include <sound/jack.h> @@ -2809,6 +2810,25 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +/* Retrieve a card's name from device tree */ +static int snd_soc_of_parse_card_name(struct snd_soc_card *card) +{ + struct device_node *np = card->dev->of_node; + int ret; + + ret = of_property_read_string_index(np, "user-visible-name", 0, + &card->name); + /* + * EINVAL means the property does not exist. This is fine providing + * card->name was previously set, which is checked later in + * snd_soc_register_card. + */ + if (ret < 0 && ret != -EINVAL) + return ret; + + return 0; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2817,11 +2837,22 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ int snd_soc_register_card(struct snd_soc_card *card) { - int i; + int i, ret; - if (!card->name || !card->dev) + if (!card->dev) return -EINVAL; + if (card->dev->of_node) { + ret = snd_soc_of_parse_card_name(card); + if (ret < 0) + return ret; + } + + if (!card->name) { + dev_err(card->dev, "Card name is not set\n"); + return -EINVAL; + } + dev_set_drvdata(card->dev, card); snd_soc_initialize_card_lists(card); -- 1.7.0.4

3 18

[alsa-devel] QasTools 0.16.0 released
by Sebastian H. 13 Dec '11

13 Dec '11

QasTools version 0.16.0 is now available. QasTools is a collection of dektop applications for ALSA. Changes: * QasTools replaces separate QasConfig and QasMixer packages * QasHctl: QasHctl forked out of QasMixer into separate application * QasHctl: Wrong pixmap in switch widgets fixed * QasMixer: Command line arguments -D and -c fixed * QasMixer: Command line arguments descriptions added in manpage * All: Info dialog adapted to QasTools Homepage with more information http://xwmw.org/qastools Project page http://sourceforge.net/projects/qastools/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] [PATCH] plug: Move fast_ops copy after slave configuration.
by Reilly Grant 12 Dec '11

12 Dec '11

The plug plugin copies the fast_ops structure and fast_op_arg from its slave PCM on initialization and after inserting a new plugin. This works unless the slave PCM is also an instance of the plug plugin because plug can change its fast_ops structure during _snd_pcm_hw_params. Instead of copying fast_ops in snd_pcm_plug_insert_plugins wait until the end of snd_pcm_plug_hw_params when the slave has been fully initialized. This fixes a crash when two instances of plug are instantiated back to back and the format conversion plugin inserted by the second is skipped by the first because the fast_ops pointer is not properly updated. This is the case for applications requesting plug:default on Intel HDA chips. --- src/pcm/pcm_plug.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index e9d2923..72456d5 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -659,8 +659,6 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, } if (err) { plug->gen.slave = new; - pcm->fast_ops = new->fast_ops; - pcm->fast_op_arg = new->fast_op_arg; } k++; } @@ -1049,6 +1047,9 @@ static int snd_pcm_plug_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) } snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); snd_pcm_unlink_appl_ptr(pcm, plug->req_slave); + + pcm->fast_ops = slave->fast_ops; + pcm->fast_op_arg = slave->fast_op_arg; snd_pcm_link_hw_ptr(pcm, slave); snd_pcm_link_appl_ptr(pcm, slave); return 0; -- 1.7.3.4

2 2

[alsa-devel] Problem calling "snd_pcm_hw_params_set_channels" with a RME 9652 (the same call -with more channels- works OK on a MADI one)
by Natanael Olaiz 12 Dec '11

12 Dec '11

Hi all, I have a problem with an application that worked OK on a RME Madi but now is failing with a 9652. It fails calling to snd_pcm_hw_params_set_channels . With the 9652 i'm getting an error -22 "invalid argument", with any number of channels (I understand it should be 24 ?) Any idea? Alsa ver. : 1.0.24 Kernel: 3.0.4 /proc/asound/cards : 0 [DSP ]: H-DSP - Hammerfall DSP RME Hammerfall HDSP 9652 at 0xfebf0000, irq 23 Thanks in advance, Natanael.

1 0

[alsa-devel] [PATCH] ALSA: snd-usb: added VOX ToneLab ST midi handling
by Daniel Mack 12 Dec '11

12 Dec '11

From: Sergiusz Urbaniak <sergiusz.urbaniak(a)gmail.com> Signed-off-by: Sergiusz Urbaniak <sergiusz.urbaniak(a)googlemail.com> --- Sergiusz writes: Hi, I posted the message below to the mailing list, but it still is in the approval queue since I'm not a mailing list member. In case this gets forgotten or stuck in a spam queue could you please consider my patch for kernel inclusion? This works really great with my Tonelab ST! I can even imagine to write a lib/CLI/GTK based app to control the Tonelab now :-) Thanks, Sergiusz Urbaniak sound/usb/quirks-table.h | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b61945f..57c9b94 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2305,6 +2305,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + USB_DEVICE_VENDOR_SPEC(0x0944, 0x0201), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "KORG, Inc.", + /* .product_name = "ToneLab ST", */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE, + } +}, + /* AKAI devices */ { USB_DEVICE(0x09e8, 0x0062), -- 1.7.7.3

2 2

[alsa-devel] [PATCH] ASoC: Add trivial pm_runtime usage to Samsung DAI drivers
by Mark Brown 12 Dec '11

12 Dec '11

Currently this won't actually do anything but using this will help the core SoC code track when the system is idle. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/i2s.c | 6 +++++- sound/soc/samsung/pcm.c | 5 +++++ 2 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ff5d919..87a874d 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -15,6 +15,7 @@ #include <linux/clk.h> #include <linux/io.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -1095,6 +1096,8 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv); + pm_runtime_enable(&pdev->dev); + return 0; err: release_mem_region(regs_base, resource_size(res)); @@ -1105,6 +1108,7 @@ err: static __devexit int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; + struct resource *res; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; @@ -1113,7 +1117,7 @@ static __devexit int samsung_i2s_remove(struct platform_device *pdev) other->pri_dai = NULL; other->sec_dai = NULL; } else { - struct resource *res; + pm_runtime_disable(&pdev->dev); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res) release_mem_region(res->start, resource_size(res)); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 3a29c26..a1dc141c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -14,6 +14,7 @@ #include <linux/clk.h> #include <linux/io.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -581,6 +582,8 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + pm_runtime_enable(&pdev->dev); + ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); if (ret != 0) { dev_err(&pdev->dev, "failed to get register DAI: %d\n", ret); @@ -610,6 +613,8 @@ static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); + pm_runtime_disable(&pdev->dev); + iounmap(pcm->regs); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ALSA: HDA: Realtek: Take vmaster dac from multiout dac list
by David Henningsson 12 Dec '11

12 Dec '11

With the auto-parser we can choose the dac nid for vmaster from the DACs we already know, instead of hard-coding it. This is more future-proof and was actually wrong on one machine. Reported-by: David Chen <david.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 32 ++++++++++---------------------- 1 files changed, 10 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a74c1e..31dd675 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3153,6 +3153,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) } } + spec->vmaster_nid = spec->multiout.dac_nids[0]; return 0; } @@ -4175,8 +4176,10 @@ static int patch_alc880(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { + spec->vmaster_nid = 0x0c; setup_preset(codec, &alc880_presets[board_config]); + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4196,8 +4199,6 @@ static int patch_alc880(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4304,8 +4305,10 @@ static int patch_alc260(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc260_presets[board_config]); + spec->vmaster_nid = 0x08; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4325,8 +4328,6 @@ static int patch_alc260(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x08; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4698,8 +4699,10 @@ static int patch_alc882(struct hda_codec *codec) goto error; } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc882_presets[board_config]); + spec->vmaster_nid = 0x0c; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4719,8 +4722,6 @@ static int patch_alc882(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4899,8 +4900,6 @@ static int patch_alc262(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5012,8 +5011,6 @@ static int patch_alc268(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5568,8 +5565,6 @@ static int patch_alc269(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; @@ -5674,8 +5669,6 @@ static int patch_alc861(struct hda_codec *codec) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } - spec->vmaster_nid = 0x03; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -5800,8 +5793,6 @@ static int patch_alc861vd(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - spec->vmaster_nid = 0x02; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -6184,7 +6175,6 @@ static int patch_alc662(struct hda_codec *codec) break; } } - spec->vmaster_nid = 0x02; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6240,8 +6230,6 @@ static int patch_alc680(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; -- 1.7.5.4

2 5

[alsa-devel] [PATCH] ASoC: Add missed MODULE_LICENSE("GPL") for imx-pcm-fiq
by Axel Lin 12 Dec '11

12 Dec '11

This driver can be built as module and the file header indicates that the driver is published under the GPL. Thus add MODULE_LICENSE("GPL") for it. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/imx-pcm-fiq.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index d7ea0b3..456b7d7 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -332,3 +332,5 @@ static struct platform_driver imx_pcm_driver = { }; module_platform_driver(imx_pcm_driver); + +MODULE_LICENSE("GPL"); -- 1.7.5.4

4 3

[alsa-devel] [PATCH v2] Export symbols 'mxs_saif_put_mclk()' and 'mxs_saif_get_mclk()'
by Lothar Waßmann 12 Dec '11

12 Dec '11

Fixes the following errors with MXS sound support built as module: |ERROR: "mxs_saif_put_mclk" [sound/soc/mxs/snd-soc-mxs-sgtl5000.ko] undefined! |ERROR: "mxs_saif_get_mclk" [sound/soc/mxs/snd-soc-mxs-sgtl5000.ko] undefined! Signed-off-by: Lothar Waßmann <LW(a)KARO-electronics.de> --- sound/soc/mxs/mxs-saif.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 487d2ff..ec66a29 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -221,6 +221,7 @@ int mxs_saif_put_mclk(unsigned int saif_id) saif->mclk_in_use = 0; return 0; } +EXPORT_SYMBOL_GPL(mxs_saif_put_mclk); /* * Get MCLK and set clock rate, then enable it @@ -274,7 +275,7 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, return 0; } +EXPORT_SYMBOL_GPL(mxs_saif_get_mclk); /* * SAIF DAI format configuration. -- 1.5.6.5

2 1

[alsa-devel] [PATCH 0/5] ASoC: mxs: Several cleanups and bugfixes for the MXS sound driver
by Lothar Waßmann 12 Dec '11

12 Dec '11

Lothar Waßmann (5): General cleanups Add .owner to struct snd_soc_card Export symbols 'mxs_saif_put_mclk()' and 'mxs_saif_get_mclk()' Add missing MODULE_LICENSE("GPL") Add appropriate MODULE_ALIAS() sound/soc/codecs/sgtl5000.c | 13 ++++++------- sound/soc/mxs/mxs-pcm.c | 3 +++ sound/soc/mxs/mxs-saif.c | 4 +++- sound/soc/mxs/mxs-sgtl5000.c | 2 ++ 4 files changed, 14 insertions(+), 8 deletions(-)

3 18

[alsa-devel] [PATCH] ALSA: HDA: Set position fix to LPIB for an Atom/Poulsbo based device
by David Henningsson 12 Dec '11

12 Dec '11

For the Asus 1101HA, reporting position by reading the DMA position buffer map seems unstable and often wrong. The reporter says that position_fix=LPIB works much better (although not 100%, but this is probably due to other issues). The controller chip is an Intel Poulsbo 8086:811b (rev 07) controller, and complete alsa-info is available here: https://launchpadlibrarian.net/86691768/alsa-info.txt.1TNwyE5Ea7 Cc: stable(a)kernel.org (3.0+) BugLink: http://bugs.launchpad.net/bugs/825709 Tested-by: Stefano Lodi Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d1582dd..0746ab4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2506,6 +2506,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS 1101HA", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), -- 1.7.5.4

2 1

[alsa-devel] [linux-next] ASoC: build errors if CONFIG_SND_SOC=m && CONFIG_SND_SOC_ALL_CODECS=m
by Axel Lin 12 Dec '11

12 Dec '11

hi, I got below build errors if CONFIG_SND_SOC=m && CONFIG_SND_SOC_ALL_CODECS=m. ( build test on linux-next 20111212 ). Kernel: arch/arm/boot/zImage is ready Building modules, stage 2. MODPOST 87 modules ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8974.ko] undefined! ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8940.ko] undefined! ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8753.ko] undefined! ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8510.ko] undefined! make[1]: *** [__modpost] Error 1 make: *** [modules] Error 2 Here is my gcc version: $ arm-eabi-gcc -v Using built-in specs. Target: arm-eabi Configured with: /home/jingyu/projects/gcc/android-toolchainsrc/build/../gcc/gcc-4.4.3/configure --prefix=/usr/local --target=arm-eabi --host=x86_64-linux-gnu --build=x86_64-linux-gnu --with-gnu-as --with-gnu-ld --enable-languages=c,c++ --with-gmp=/home/jingyu/projects/gcc/toolchain_build/obj/temp-install --with-mpfr=/home/jingyu/projects/gcc/toolchain_build/obj/temp-install --disable-libssp --enable-threads --disable-nls --disable-libmudflap --disable-libgomp --disable-libstdc__-v3 --disable-sjlj-exceptions --disable-shared --disable-tls --with-float=soft --with-fpu=vfp --with-arch=armv5te --enable-target-optspace --with-abi=aapcs --with-gcc-version=4.4.3 --with-binutils-version=2.19 --with-gmp-version=4.2.4 --with-mpfr-version=2.4.1 --with-gdb-version=7.1.x --with-arch=armv5te --with-multilib-list=mandroid --with-sysroot=/usr/local/google/home/android/cupcake_rel_root --program-transform-name='s&^&arm-eabi-&' Thread model: single gcc version 4.4.3 (GCC) Regards, Axel

1 0

[alsa-devel] [PATCH v3] sound/soc/lapis: add machine driver v3 updated
by Tomoya MORINAGA 12 Dec '11

12 Dec '11

Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- V3 Update Mark's comments - Delete i2c_board_info data structure. - Delete i2c_new_device/i2c_put_adapter. - Delete .startup method. - Add error processing for failure of snd_soc_unregister_card. - Add MODULE_ALIAS - Change platform_driver name - Set to .dai_fmt of snd_soc_dai_link not use snd_soc_dai_set_fmt --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/lapis/Kconfig | 5 + sound/soc/lapis/Makefile | 5 + sound/soc/lapis/ml7213ioh-machine.c | 158 +++++++++++++++++++++++++++++++++++ 5 files changed, 170 insertions(+), 0 deletions(-) create mode 100644 sound/soc/lapis/Kconfig create mode 100644 sound/soc/lapis/Makefile create mode 100644 sound/soc/lapis/ml7213ioh-machine.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 35e662d..63f8582 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -34,6 +34,7 @@ source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/imx/Kconfig" source "sound/soc/jz4740/Kconfig" +source "sound/soc/lapis/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 9ea8ac8..002951f 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -11,6 +11,7 @@ obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += jz4740/ +obj-$(CONFIG_SND_SOC) += lapis/ obj-$(CONFIG_SND_SOC) += mid-x86/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ diff --git a/sound/soc/lapis/Kconfig b/sound/soc/lapis/Kconfig new file mode 100644 index 0000000..f6c2fe5 --- /dev/null +++ b/sound/soc/lapis/Kconfig @@ -0,0 +1,5 @@ +config SND_SOC_ML7213_MACHINE + tristate "ML7213 IOH ASoC machine driver" + select SND_SOC_ML7213_PLATFORM + help + This is ASoC machine driver for ML7213 IOH diff --git a/sound/soc/lapis/Makefile b/sound/soc/lapis/Makefile new file mode 100644 index 0000000..3560bd7 --- /dev/null +++ b/sound/soc/lapis/Makefile @@ -0,0 +1,5 @@ +# Platform +snd-soc-ml7213-machine-objs := ml7213ioh-machine.o + +obj-$(CONFIG_SND_SOC_ML7213_PLATFORM) += snd-soc-ml7213-machine.o + diff --git a/sound/soc/lapis/ml7213ioh-machine.c b/sound/soc/lapis/ml7213ioh-machine.c new file mode 100644 index 0000000..28e7919 --- /dev/null +++ b/sound/soc/lapis/ml7213ioh-machine.c @@ -0,0 +1,158 @@ +/* + * ml7213ioh-machine.c -- SoC Audio for LAPIS Semiconductor ML7213 IOH CRB + * + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <linux/i2c.h> +#include "../codecs/ml26124.h" + +static const struct snd_soc_dapm_widget ml7213_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_LINE("LINEOUT", NULL), + SND_SOC_DAPM_LINE("LINEIN", NULL), + SND_SOC_DAPM_MIC("AnalogMIC", NULL), + SND_SOC_DAPM_MIC("DigitalMIC", NULL), +}; + +static const struct snd_soc_dapm_route ml7213_routes[] = { + {"Speaker", NULL, "SPOUT"}, + {"LINEOUT", NULL, "LOUT"}, + {"MIN", NULL, "AnalogMIC"}, + {"MDIN", NULL, "DigitalMIC"}, + {"MIN", NULL, "LINEIN"}, +}; + +static int ml7213_ioh_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, ml7213_dapm_widgets, + ARRAY_SIZE(ml7213_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, ml7213_routes, ARRAY_SIZE(ml7213_routes)); + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int ml7213_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->codec_dai; + unsigned int clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 16000: + case 32000: + case 48000: + clk = 12288800; + break; + default: + return -EINVAL; + } + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, ML26124_USE_PLL, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops ml7213_ops = { + .hw_params = ml7213_hw_params, +}; + +static struct snd_soc_dai_link ioh_i2s_dai = { + .name = "ML7213", + .stream_name = "I2S HiFi", + .cpu_dai_name = "ml7213-i2s", + .codec_dai_name = "ml26124-hifi", + .platform_name = "ml7213-i2s-pcm-audio", + .codec_name = "ml26124-00GD", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .init = ml7213_ioh_init, + .ops = &ml7213_ops, +}; + +static struct snd_soc_card ioh_i2s_card = { + .name = "ml7213", + .dai_link = &ioh_i2s_dai, + .num_links = 1, + .dapm_widgets = ml7213_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml7213_dapm_widgets), + .dapm_routes = ml7213_routes, + .num_dapm_routes = ARRAY_SIZE(ml7213_routes), +}; + +static int __init ioh_i2s_probe(struct platform_device *pdev) +{ + ioh_i2s_card.dev = &pdev->dev; + snd_soc_register_card(&ioh_i2s_card); + + return 0; +} + +static int __exit ioh_i2s_remove(struct platform_device *pdev) +{ + int ret = snd_soc_unregister_card(&ioh_i2s_card); + + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + ioh_i2s_card.dev = NULL; + return ret; + } + + return 0; +} + +static struct platform_driver ioh_i2s_driver = { + .remove = ioh_i2s_remove, + .driver = { + .name = "ml7213ioh-i2s", + .owner = THIS_MODULE, + }, + .probe = ioh_i2s_probe, + .remove = __devexit_p(ioh_i2s_remove), +}; + +static int __init ioh_i2s_init(void) +{ + return platform_driver_register(&ioh_i2s_driver); +} + +static void __exit ioh_i2s_exit(void) +{ + return platform_driver_unregister(&ioh_i2s_driver); +} + +module_init(ioh_i2s_init); +module_exit(ioh_i2s_exit); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML7213 IOH ALSA SoC machine driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ml7213ioh-i2s"); -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: Fix hx4700 error handling to free gpios if snd_soc_register_card fails
by Axel Lin 12 Dec '11

12 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/hx4700.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index e32afaf..03ef9f3 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -209,9 +209,10 @@ static int __devinit hx4700_audio_probe(struct platform_device *pdev) snd_soc_card_hx4700.dev = &pdev->dev; ret = snd_soc_register_card(&snd_soc_card_hx4700); if (ret) - return ret; + gpio_free_array(hx4700_audio_gpios, + ARRAY_SIZE(hx4700_audio_gpios)); - return 0; + return ret; } static int __devexit hx4700_audio_remove(struct platform_device *pdev) -- 1.7.5.4

2 1

Re: [alsa-devel] [PATCH] ASoC: WM8958: correctly show firmware magic on mismatch
by Mark Brown 12 Dec '11

12 Dec '11

On Sun, Dec 11, 2011 at 07:15:49PM +0100, Jonathan Neuschäfer wrote: > Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 07/10] ASoC: OMAP: ams-delta: Drive modem/codec pins over GPIO API
by Mark Brown 12 Dec '11

12 Dec '11

On Sun, Dec 11, 2011 at 09:12:05PM +0100, Janusz Krzysztofik wrote: > Don't use Amstrad Delta custom I/O functions any longer, replace them > with GPIO. Old pin definitions, no longer used by the modem bits either, > can be dropped. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] [PATCH] ASoC: Raise Speyside audio system clock rate to 512fs
by Mark Brown 11 Dec '11

11 Dec '11

To support advanced system functionality for additional components; the actively used clocks will remain the same for current components. Also factor the rate out to a single #define while we're at it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index e0f55cb5..8cdd4f4 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -19,6 +19,7 @@ #include "../codecs/wm9081.h" #define WM8996_HPSEL_GPIO 214 +#define MCLK_AUDIO_RATE (512 * 48000) static int speyside_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, @@ -67,7 +68,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, 0, WM8996_FLL_MCLK2, - 32768, 48000 * 256); + 32768, MCLK_AUDIO_RATE); if (ret < 0) { pr_err("Failed to start FLL\n"); return ret; @@ -75,7 +76,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, ret = snd_soc_dai_set_sysclk(codec_dai, WM8996_SYSCLK_FLL, - 48000 * 256, + MCLK_AUDIO_RATE, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -197,7 +198,7 @@ static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) { /* At any time the WM9081 is active it will have this clock */ return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, - 48000 * 256, 0); + MCLK_AUDIO_RATE, 0); } static struct snd_soc_aux_dev speyside_aux_dev[] = { -- 1.7.7.3

1 0

[alsa-devel] [PATCH 0/5] ASoC: OMAP4: Device tree support for DMIC, McPDM
by Peter Ujfalusi 11 Dec '11

11 Dec '11

Hello, The following series adds device tree support for omap-dmic, omap-mcpdm ASoC DAI drivers. The OMAP4 platform patches have dependency on the DMIC driver series: http://www.mail-archive.com/linux-omap@vger.kernel.org/msg59161.html Regards, Peter --- Peter Ujfalusi (5): ASoC: omap-dmic: Add device tree bindings ASoC: omap-mcpdm: Add device tree bindings OMAP4: devices: Do not create dmic device if the dtb has been provided OMAP4: devices: Do not create mcpdm device if the dtb has been provided ARM: OMAP4: DTS: Support for dmic, and McPDM in device tree .../bindings/sound/soc/omap/omap-dmic.txt | 13 +++++++++++++ .../bindings/sound/soc/omap/omap-mcpdm.txt | 13 +++++++++++++ arch/arm/boot/dts/omap4-panda.dts | 4 ++++ arch/arm/boot/dts/omap4.dtsi | 9 +++++++++ arch/arm/mach-omap2/devices.c | 8 ++++++++ sound/soc/omap/omap-dmic.c | 8 ++++++++ sound/soc/omap/omap-mcpdm.c | 8 ++++++++ 7 files changed, 63 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/soc/omap/omap-dmic.txt create mode 100644 Documentation/devicetree/bindings/sound/soc/omap/omap-mcpdm.txt -- 1.7.8.rc4

2 13

Re: [alsa-devel] [PATCH] ASoC: Fix an obvious copy paste error in an error message
by Mark Brown 11 Dec '11

11 Dec '11

On Fri, Dec 09, 2011 at 02:16:29PM +0100, Lothar Waßmann wrote: > The message was obviously copied from soc_init_codec_debugfs() Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: Fix WM8996 24.576MHz clock operation
by Mark Brown 10 Dec '11

10 Dec '11

Record the clock after the divider as that is what all SYSCLK users see. Without this the other clock configuration in the device comes out at half rate. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 41cc9d2..8f88f5a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2043,6 +2043,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, break; case 24576000: ratediv = WM8996_SYSCLK_DIV; + wm8996->sysclk /= 2; case 12288000: snd_soc_update_bits(codec, WM8996_AIF_RATE, WM8996_SYSCLK_RATE, WM8996_SYSCLK_RATE); -- 1.7.7.3

1 0

Re: [alsa-devel] [PATCH 0/5] RFC v2 for snd-usb endpoint logic rework
by Aurélien Leblond 09 Dec '11

09 Dec '11

Hi Daniel, Compiled and running! So far so good! I haven't pushed the jack parameters more than I would usually though... I'm still at 16ms latency (Frame/period: 256 ; Sample Rate: 48000 ; Periods Buffer: 3). That was the minimum I could go with the previous driver, so I'll try to push it a bit tomorrow and see how it goes. Is there any particular testing you would like to see done? Another thing, in your previous post, you asked to try the driver with other sound card. I have an Edirol UA-25Ex that never worked on this machine (I have the dredded EHCI USB port, it's the reason why I bought the FTU in the first place). The error I used to get while starting Jack in Full Duplex with this sound card was "Cannot submit datapipe for urb 0, error -28: not enough bandwidth" But, "apparently".......now it works! I tried it not later than last week with Kernel 3.0 and it still wasn't working. But with 3.2 and your patch, (and after running the FTU on the very same USB port), I got the UA-25Ex to work in full duplex for the first time ever. I don't know if it is due to the new version of the kernel, or your patch, or something else, but this is huge because a lot of people used to have this problem with EHCI and USB 1 sound cards. (or may be it was simply fixed between 3.0 and 3.2 and I simply missed it!). So the second question is: what do you think? Any info you would like to see posted here? (dmesg trace....anything else?). Anyway, thanks a lot for everything :) Aurélien On Thu, Dec 8, 2011 at 7:38 PM, Daniel Mack <zonque(a)gmail.com> wrote: > Hi Aurélien, > > On 12/08/2011 08:33 PM, Aurélien Leblond wrote: >> I finally got some time to test this driver! >> I'm not sure I understand where to get Takashi's sound tree. >> >> I tried to apply your patch against a clone of >> git://git.alsa-project.org/alsa-kmirror.git alsa-kmirror, but then >> GitCompile failed. >> >> I then tried to use >> git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git , but >> I'm not even sure what I'm pulling to be honest... >> >> Would you mind just guiding me on how to prepare the source code? > > No problem :) > > Clone Linus' tree: > > git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git > > Make a local branch and pull in Takashi's tree: > > cd linux > git checkout -b alsa > git pull git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git > master > > Make another branch on top of that > > git checkout -b usbaudio > > Then apply my patch(es). For the next round, just kill the local > usbaudio branch again and start over: > > git reset --hard > git checkout alsa > git branch -D usbaudio > git checkout -b usbaudio > > Then apply the next round :) > >> Thanks in advance, and thanks for all the work you have already done! > > No problem. I'm planning to continue work on this these days. It's just > that I'm constantly distracted by other things :-/ > > Daniel

1 1

[alsa-devel] [PATCH v4] ASoC: Allow DAI links to be specified using device tree nodes
by Stephen Warren 09 Dec '11

09 Dec '11

DAI link endpoints and platform (DMA) devices are currently specified by name. When instantiating sound cards from device tree, it may be more convenient to refer to these devices by phandle in the device tree, and for code to describe DAI links using the "struct device_node *" ("of_node") those phandles map to. This change adds new fields to snd_soc_dai_link which can "name" devices using of_node, enhances soc_bind_dai_link() to allow binding based on of_node, and enhances snd_soc_register_card() to ensure that illegal combinations of name and of_node are not used. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-by: Liam Girdwood <lrg(a)ti.com> COMMIT NOTES: v2: New patch implementing a new feature v3: Add Liam's ack v4: Rebased to re-order this patch directly onto ASoC for-next --- include/sound/soc.h | 4 +++ sound/soc/soc-core.c | 64 ++++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 61 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d9aa66b..40c256e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -231,6 +231,7 @@ enum snd_soc_bias_level { SND_SOC_BIAS_ON = 3, }; +struct device_node; struct snd_jack; struct snd_soc_card; struct snd_soc_pcm_stream; @@ -704,8 +705,11 @@ struct snd_soc_dai_link { const char *name; /* Codec name */ const char *stream_name; /* Stream name */ const char *codec_name; /* for multi-codec */ + const struct device_node *codec_of_node; const char *platform_name; /* for multi-platform */ + const struct device_node *platform_of_node; const char *cpu_dai_name; + const struct device_node *cpu_dai_of_node; const char *codec_dai_name; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ebb1048..56792d4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -763,8 +763,13 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { - if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; + if (dai_link->cpu_dai_of_node) { + if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) + continue; + } else { + if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; + } rtd->cpu_dai = cpu_dai; goto find_codec; @@ -780,8 +785,13 @@ find_codec: /* no, then find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { - if (strcmp(codec->name, dai_link->codec_name)) - continue; + if (dai_link->codec_of_node) { + if (codec->dev->of_node != dai_link->codec_of_node) + continue; + } else { + if (strcmp(codec->name, dai_link->codec_name)) + continue; + } rtd->codec = codec; @@ -813,13 +823,19 @@ find_platform: /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; - if (!platform_name) + if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; /* no, then find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { - if (strcmp(platform->name, platform_name)) - continue; + if (dai_link->platform_of_node) { + if (platform->dev->of_node != + dai_link->platform_of_node) + continue; + } else { + if (strcmp(platform->name, platform_name)) + continue; + } rtd->platform = platform; goto out; @@ -2830,6 +2846,40 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai_link *link = &card->dai_link[i]; + + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codec_name == !!link->codec_of_node) { + dev_err(card->dev, + "Neither/both codec name/of_node are set\n"); + return -EINVAL; + } + + /* + * Platform may be specified by either name or OF node, but + * can be left unspecified, and a dummy platform will be used. + */ + if (link->platform_name && link->platform_of_node) { + dev_err(card->dev, + "Both platform name/of_node are set\n"); + return -EINVAL; + } + + /* + * CPU DAI must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { + dev_err(card->dev, + "Neither/both cpu_dai name/of_node are set\n"); + return -EINVAL; + } + } + dev_set_drvdata(card->dev, card); snd_soc_initialize_card_lists(card); -- 1.7.0.4

1 0

[alsa-devel] [PATCH] ASoC: Fix comments for disabling amplifier and PGA
by Axel Lin 09 Dec '11

09 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8961.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index dc087c1..58fbf0a 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -422,11 +422,11 @@ static int wm8961_spk_event(struct snd_soc_dapm_widget *w, } if (event & SND_SOC_DAPM_PRE_PMD) { - /* Enable the amplifier */ + /* Disable the amplifier */ spk_reg &= ~(WM8961_SPKL_ENA | WM8961_SPKR_ENA); snd_soc_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg); - /* Enable the PGA */ + /* Disable the PGA */ pwr_reg &= ~(WM8961_SPKL_PGA | WM8961_SPKR_PGA); snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg); } -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: Convert WM8776 to devm_kzalloc()
by Mark Brown 09 Dec '11

09 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8776.c | 14 ++++++-------- 1 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index fbf80c5..38b4556 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -452,7 +452,8 @@ static int __devinit wm8776_spi_probe(struct spi_device *spi) struct wm8776_priv *wm8776; int ret; - wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + wm8776 = devm_kzalloc(&spi->dev, sizeof(struct wm8776_priv), + GFP_KERNEL); if (wm8776 == NULL) return -ENOMEM; @@ -461,15 +462,13 @@ static int __devinit wm8776_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8776, wm8776_dai, ARRAY_SIZE(wm8776_dai)); - if (ret < 0) - kfree(wm8776); + return ret; } static int __devexit wm8776_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -491,7 +490,8 @@ static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, struct wm8776_priv *wm8776; int ret; - wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + wm8776 = devm_kzalloc(&i2c->dev, sizeof(struct wm8776_priv), + GFP_KERNEL); if (wm8776 == NULL) return -ENOMEM; @@ -500,15 +500,13 @@ static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8776, wm8776_dai, ARRAY_SIZE(wm8776_dai)); - if (ret < 0) - kfree(wm8776); + return ret; } static __devexit int wm8776_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.7.3

1 3

[PATCH] ALSA: HDA: Realtek: Take vmaster dac from multiout dac list
by David Henningsson 09 Dec '11

09 Dec '11

With the auto-parser we can choose the dac nid for vmaster from the DACs we already know, instead of hard-coding it. This is more future-proof and was actually wrong on one machine. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 36 ++++++++++++++---------------------- 1 files changed, 14 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a74c1e..f394865 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3039,6 +3039,8 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, static int alc_auto_fill_multi_ios(struct hda_codec *codec, unsigned int location, int offset); +static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec, + hda_nid_t pin, hda_nid_t dac); /* fill in the dac_nids table from the parsed pin configuration */ static int alc_auto_fill_dac_nids(struct hda_codec *codec) @@ -3153,6 +3155,9 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) } } + i = cfg->line_outs ? cfg->line_out_pins[0] : + (cfg->speaker_outs ? cfg->speaker_pins[0] : cfg->hp_pins[0]); + spec->vmaster_nid = alc_look_for_out_vol_nid(codec, spec->multiout.dac_nids[0], i); return 0; } @@ -4175,8 +4180,10 @@ static int patch_alc880(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { + spec->vmaster_nid = 0x0c; setup_preset(codec, &alc880_presets[board_config]); + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4196,8 +4203,6 @@ static int patch_alc880(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4304,8 +4309,10 @@ static int patch_alc260(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc260_presets[board_config]); + spec->vmaster_nid = 0x08; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4325,8 +4332,6 @@ static int patch_alc260(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x08; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4698,8 +4703,10 @@ static int patch_alc882(struct hda_codec *codec) goto error; } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc882_presets[board_config]); + spec->vmaster_nid = 0x0c; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4719,8 +4726,6 @@ static int patch_alc882(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4899,8 +4904,6 @@ static int patch_alc262(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5012,8 +5015,6 @@ static int patch_alc268(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5568,8 +5569,6 @@ static int patch_alc269(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; @@ -5674,8 +5673,6 @@ static int patch_alc861(struct hda_codec *codec) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } - spec->vmaster_nid = 0x03; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -5800,8 +5797,6 @@ static int patch_alc861vd(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - spec->vmaster_nid = 0x02; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -6184,7 +6179,6 @@ static int patch_alc662(struct hda_codec *codec) break; } } - spec->vmaster_nid = 0x02; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6240,8 +6234,6 @@ static int patch_alc680(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; -- 1.7.5.4 --------------000303040509040305040806--

1 0

[PATCH] ALSA: HDA: Realtek: Take vmaster dac from multiout dac list
by David Henningsson 09 Dec '11

09 Dec '11

With the auto-parser we can choose the dac nid for vmaster from the DACs we already know, instead of hard-coding it. This is more future-proof and was actually wrong on one machine. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 36 ++++++++++++++---------------------- 1 files changed, 14 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a74c1e..2bf08fe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3039,6 +3039,8 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, static int alc_auto_fill_multi_ios(struct hda_codec *codec, unsigned int location, int offset); +static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec, + hda_nid_t pin, hda_nid_t dac); /* fill in the dac_nids table from the parsed pin configuration */ static int alc_auto_fill_dac_nids(struct hda_codec *codec) @@ -3153,6 +3155,9 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) } } + if (cfg->line_out_pins[0]) + spec->vmaster_nid = alc_look_for_out_vol_nid(codec, spec->multiout.dac_nids[0], + cfg->line_out_pins[0]); return 0; } @@ -4175,8 +4180,10 @@ static int patch_alc880(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { + spec->vmaster_nid = 0x0c; setup_preset(codec, &alc880_presets[board_config]); + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4196,8 +4203,6 @@ static int patch_alc880(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4304,8 +4309,10 @@ static int patch_alc260(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc260_presets[board_config]); + spec->vmaster_nid = 0x08; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4325,8 +4332,6 @@ static int patch_alc260(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x08; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4698,8 +4703,10 @@ static int patch_alc882(struct hda_codec *codec) goto error; } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc882_presets[board_config]); + spec->vmaster_nid = 0x0c; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4719,8 +4726,6 @@ static int patch_alc882(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4899,8 +4904,6 @@ static int patch_alc262(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5012,8 +5015,6 @@ static int patch_alc268(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5568,8 +5569,6 @@ static int patch_alc269(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; @@ -5674,8 +5673,6 @@ static int patch_alc861(struct hda_codec *codec) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } - spec->vmaster_nid = 0x03; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -5800,8 +5797,6 @@ static int patch_alc861vd(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - spec->vmaster_nid = 0x02; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -6184,7 +6179,6 @@ static int patch_alc662(struct hda_codec *codec) break; } } - spec->vmaster_nid = 0x02; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6240,8 +6234,6 @@ static int patch_alc680(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; -- 1.7.5.4 --------------070204090502030907040600--

1 0

Re: [alsa-devel] [PATCH v2] soc/lapis: add machine driver
by Mark Brown 09 Dec '11

09 Dec '11

On Fri, Dec 02, 2011 at 06:45:15PM +0900, Tomoya MORINAGA wrote: > +#define CODEC_DEV_ADDR 0x1A > +static struct i2c_board_info ioh_hwmon_info[] = { > + {I2C_BOARD_INFO("ioh_i2c-0", CODEC_DEV_ADDR + 1)}, > + {I2C_BOARD_INFO("ioh_i2c-1", CODEC_DEV_ADDR + 2)}, > + {I2C_BOARD_INFO("ioh_i2c-2", CODEC_DEV_ADDR + 3)}, > + {I2C_BOARD_INFO("ioh_i2c-3", CODEC_DEV_ADDR + 4)}, > + {I2C_BOARD_INFO("ioh_i2c-4", CODEC_DEV_ADDR + 5)}, > + {I2C_BOARD_INFO("ioh_i2c-5", CODEC_DEV_ADDR + 0)}, This looks completely wrong. I'd not expect to see any I2C_BOARD_INFO usage at all in a machine driver (that should be done by whatever enumerates the system as a whole) and I'm not clear what the intended effect of the above is - the names look like the names of I2C controllers... > +static int ml7213_ioh_init(struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_codec *codec = rtd->codec; > + struct snd_soc_dapm_context *dapm = &codec->dapm; > + > + snd_soc_dapm_new_controls(dapm, ml7213_dapm_widgets, > + ARRAY_SIZE(ml7213_dapm_widgets)); > + > + snd_soc_dapm_add_routes(dapm, ml7213_routes, ARRAY_SIZE(ml7213_routes)); Use table based init for these. > +static int ml7213_startup(struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_codec *codec = rtd->codec; > + struct i2c_client *i2c; > + struct i2c_adapter *adapter; > + > + mutex_lock(&codec->mutex); > + > + adapter = i2c_get_adapter(1); > + if (!adapter) > + return -ENODEV; > + > + i2c = i2c_new_device(adapter, &ioh_hwmon_info[substream->number]); > + if (!i2c) { > + mutex_unlock(&codec->mutex); > + return -1; > + } > + i2c_put_adapter(adapter); > + mutex_unlock(&codec->mutex); No, this is definitely wrong - we certainly shouldn't be registering new devices whenever someone starts an audio stream. I'm struggling to understand what the intended effect is though. > + switch (params_rate(params)) { > + case 16000: > + case 32000: > + case 48000: > + clk = 12288800; > + break; You may as well just set the CODEC clock rate once on init() and let the CODEC driver worry about what it can do with the clock it's got, if the values aren't going to change there's no need to set them every time. > + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); > + if (ret < 0) > + return ret; Use the dai_fmt field in the dai_link to set this. > +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); > +MODULE_DESCRIPTION("LAPIS Semiconductor ML7213 IOH ALSA SoC machine driver"); > +MODULE_LICENSE("GPL"); Should have MODULE_ALIAS too.

3 13

[alsa-devel] [PATCH] ASoC: wm8960: Use snd_soc_update_bits for read-modify-write
by Axel Lin 08 Dec '11

08 Dec '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8960.c | 67 ++++++++++++++------------------------------- 1 files changed, 21 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 6a9c41d..2315b86 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -543,30 +543,24 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, static int wm8960_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, WM8960_DACCTL1) & 0xfff7; if (mute) - snd_soc_write(codec, WM8960_DACCTL1, mute_reg | 0x8); + snd_soc_update_bits(codec, WM8960_DACCTL1, 0x8, 0x8); else - snd_soc_write(codec, WM8960_DACCTL1, mute_reg); + snd_soc_update_bits(codec, WM8960_DACCTL1, 0x8, 0); return 0; } static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* Set VMID to 2x50k */ - reg = snd_soc_read(codec, WM8960_POWER1); - reg &= ~0x180; - reg |= 0x80; - snd_soc_write(codec, WM8960_POWER1, reg); + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); break; case SND_SOC_BIAS_STANDBY: @@ -579,23 +573,19 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, WM8960_BUFDCOPEN | WM8960_BUFIOEN); /* Enable & ramp VMID at 2x50k */ - reg = snd_soc_read(codec, WM8960_POWER1); - reg |= 0x80; - snd_soc_write(codec, WM8960_POWER1, reg); + snd_soc_update_bits(codec, WM8960_POWER1, 0x80, 0x80); msleep(100); /* Enable VREF */ - snd_soc_write(codec, WM8960_POWER1, reg | WM8960_VREF); + snd_soc_update_bits(codec, WM8960_POWER1, WM8960_VREF, + WM8960_VREF); /* Disable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, WM8960_BUFIOEN); } /* Set VMID to 2x250k */ - reg = snd_soc_read(codec, WM8960_POWER1); - reg &= ~0x180; - reg |= 0x100; - snd_soc_write(codec, WM8960_POWER1, reg); + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x100); break; case SND_SOC_BIAS_OFF: @@ -787,10 +777,8 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Disable the PLL: even if we are changing the frequency the * PLL needs to be disabled while we do so. */ - snd_soc_write(codec, WM8960_CLOCK1, - snd_soc_read(codec, WM8960_CLOCK1) & ~1); - snd_soc_write(codec, WM8960_POWER2, - snd_soc_read(codec, WM8960_POWER2) & ~1); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, 0); + snd_soc_update_bits(codec, WM8960_POWER2, 0x1, 0); if (!freq_in || !freq_out) return 0; @@ -809,11 +797,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8960_PLL1, reg); /* Turn it on */ - snd_soc_write(codec, WM8960_POWER2, - snd_soc_read(codec, WM8960_POWER2) | 1); + snd_soc_update_bits(codec, WM8960_POWER2, 0x1, 0x1); msleep(250); - snd_soc_write(codec, WM8960_CLOCK1, - snd_soc_read(codec, WM8960_CLOCK1) | 1); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, 0x1); return 0; } @@ -913,7 +899,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); struct wm8960_data *pdata = dev_get_platdata(codec->dev); int ret; - u16 reg; wm8960->set_bias_level = wm8960_set_bias_level_out3; @@ -944,26 +929,16 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = snd_soc_read(codec, WM8960_LINVOL); - snd_soc_write(codec, WM8960_LINVOL, reg | 0x100); - reg = snd_soc_read(codec, WM8960_RINVOL); - snd_soc_write(codec, WM8960_RINVOL, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LADC); - snd_soc_write(codec, WM8960_LADC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_RADC); - snd_soc_write(codec, WM8960_RADC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LDAC); - snd_soc_write(codec, WM8960_LDAC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_RDAC); - snd_soc_write(codec, WM8960_RDAC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LOUT1); - snd_soc_write(codec, WM8960_LOUT1, reg | 0x100); - reg = snd_soc_read(codec, WM8960_ROUT1); - snd_soc_write(codec, WM8960_ROUT1, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LOUT2); - snd_soc_write(codec, WM8960_LOUT2, reg | 0x100); - reg = snd_soc_read(codec, WM8960_ROUT2); - snd_soc_write(codec, WM8960_ROUT2, reg | 0x100); + snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LADC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_RADC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LDAC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_RDAC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LOUT1, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_ROUT1, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LOUT2, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_ROUT2, 0x100, 0x100); snd_soc_add_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: WM8903: DT bindings & related
by Stephen Warren 08 Dec '11

08 Dec '11

v2: * Fixed default pdata logic so that 0 in pdata fields really means "do nothing". * Added a patch to fix pdata gpio_cfg value confusion. * Moved all the pdata default & DT parsing logic to i2c_probe from codec probe. * Split parsing logic out into separate functions so probe is easier to read. * Rebased on top of the WM8903 cleanup patches Mark posted today. Stephen Warren (5): ASoC: WM8903: Fix platform data gpio_cfg confusion ASoC: WM8903: Create default platform data structure ASoC: WM8903: Remove conditionals checking pdata != NULL ASoC: WM8903: Get default irq_active_low from IRQ controller ASoC: WM8903: Add device tree binding Documentation/devicetree/bindings/sound/wm8903.txt | 50 ++++++ arch/arm/mach-tegra/board-harmony.c | 8 +- arch/arm/mach-tegra/board-seaboard.c | 8 +- include/sound/wm8903.h | 7 +- sound/soc/codecs/wm8903.c | 181 +++++++++++++++----- 5 files changed, 205 insertions(+), 49 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8903.txt

3 13

Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Nallasellan, Singaravelan 07 Dec '11

07 Dec '11

> > Even the thread which actually writes the data to ring buffer may not > > have the idea of frames So the start threshold does not require to be > > on frame boundaries. It is just to avoid START when the buffer is > > written with the data. If the write gives more data than the ring > > buffer, write will not unblock and the user space application may not > > be able to call START. > > This will be an issue if you go for non-blocking implementation. > > Otherwise, START should be called in another thread than the thread > > which writes the data. > > I did not understand a single line of this comment. > Why is this start threshold needed again and how would its value be defined for > compressed data? I will try to explain it with more details. Consider the case of blocking "write" call: If the user space app writes the buffer with the content of size greater than the ring buffer size, write call will block. How do you expect the user space to invoke START ioctl? I assume the write call will block until it writes the complete content into the buffer. To avoid this behavior, the app has to query only the empty space and write only that much data. Then it should call START.

1 0

[alsa-devel] [PATCH 1/3] ASoC: Fix error handling in e800_init to free gpios
by Axel Lin 07 Dec '11

07 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/e800_wm9712.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 6a8f38b..26e0232 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -136,8 +136,10 @@ static int __init e800_init(void) goto free_spk_amp_gpio; e800_snd_device = platform_device_alloc("soc-audio", -1); - if (!e800_snd_device) - return -ENOMEM; + if (!e800_snd_device) { + ret = -ENOMEM; + goto free_spk_amp_gpio; + } platform_set_drvdata(e800_snd_device, &e800); ret = platform_device_add(e800_snd_device); -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ASoC: Fix a typo in s3c24xx_simtec_tlv320aic23 driver
by Axel Lin 07 Dec '11

07 Dec '11

Fix a typo introduced by commit e00c3f55 "ASoC: Convert Samsung directory to module_platform_driver". This fixes the build error: CC sound/soc/samsung/s3c24xx_simtec_tlv320aic23.o sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c: In function 'simtec_audio_tlv320aic32_driver_init': sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: 'simtec_audio_tlv320aic32_driver' undeclared (first use in this function) sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: (Each undeclared identifier is reported only once sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: for each function it appears in.) sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c: In function 'simtec_audio_tlv320aic32_driver_exit': sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: 'simtec_audio_tlv320aic32_driver' undeclared (first use in this function) make[3]: *** [sound/soc/samsung/s3c24xx_simtec_tlv320aic23.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 I think we had better naming it with *driver, thus I change it to simtec_audio_tlv320aic23_driver. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 7324609..89b57b5 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -92,7 +92,7 @@ static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) return simtec_audio_core_probe(pd, &snd_soc_machine_simtec_aic23); } -static struct platform_driver simtec_audio_tlv320aic23_platdrv = { +static struct platform_driver simtec_audio_tlv320aic23_driver = { .driver = { .owner = THIS_MODULE, .name = "s3c24xx-simtec-tlv320aic23", @@ -102,7 +102,7 @@ static struct platform_driver simtec_audio_tlv320aic23_platdrv = { .remove = __devexit_p(simtec_audio_remove), }; -module_platform_driver(simtec_audio_tlv320aic32_driver); +module_platform_driver(simtec_audio_tlv320aic23_driver); MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Include linux/module.h for smdk2443_wm9710
by Axel Lin 07 Dec '11

07 Dec '11

Include linux/module.h to fix below build error: CC sound/soc/samsung/smdk2443_wm9710.o sound/soc/samsung/smdk2443_wm9710.c:64: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/smdk2443_wm9710.c:64: warning: data definition has no type or storage class sound/soc/samsung/smdk2443_wm9710.c:64: warning: type defaults to 'int' in declaration of 'MODULE_AUTHOR' sound/soc/samsung/smdk2443_wm9710.c:64: warning: function declaration isn't a prototype sound/soc/samsung/smdk2443_wm9710.c:65: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/smdk2443_wm9710.c:65: warning: data definition has no type or storage class sound/soc/samsung/smdk2443_wm9710.c:65: warning: type defaults to 'int' in declaration of 'MODULE_DESCRIPTION' sound/soc/samsung/smdk2443_wm9710.c:65: warning: function declaration isn't a prototype sound/soc/samsung/smdk2443_wm9710.c:66: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/smdk2443_wm9710.c:66: warning: data definition has no type or storage class sound/soc/samsung/smdk2443_wm9710.c:66: warning: type defaults to 'int' in declaration of 'MODULE_LICENSE' sound/soc/samsung/smdk2443_wm9710.c:66: warning: function declaration isn't a prototype make[3]: *** [sound/soc/samsung/smdk2443_wm9710.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/smdk2443_wm9710.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c index 3a0dbfc7..8bd1dc5 100644 --- a/sound/soc/samsung/smdk2443_wm9710.c +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -12,6 +12,7 @@ * */ +#include <linux/module.h> #include <sound/soc.h> static struct snd_soc_card smdk2443; -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Fix a typo in jive_wm8750
by Axel Lin 07 Dec '11

07 Dec '11

Fix a typo in jive_wm8750 that introduces below build error. Also removes an unused err variable. CC sound/soc/samsung/jive_wm8750.o sound/soc/samsung/jive_wm8750.c: In function 'jive_wm8750_init': sound/soc/samsung/jive_wm8750.c:104: warning: unused variable 'err' sound/soc/samsung/jive_wm8750.c: At top level: sound/soc/samsung/jive_wm8750.c:134: error: unknown field 'dapm_widgtets' specified in initializer sound/soc/samsung/jive_wm8750.c:134: warning: initialization from incompatible pointer type make[3]: *** [sound/soc/samsung/jive_wm8750.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/jive_wm8750.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 1826acf..8e523fd 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -101,7 +101,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; /* These endpoints are not being used. */ snd_soc_dapm_nc_pin(dapm, "LINPUT2"); @@ -131,7 +130,7 @@ static struct snd_soc_card snd_soc_machine_jive = { .dai_link = &jive_dai, .num_links = 1, - .dapm_widgtets = wm8750_dapm_widgets, + .dapm_widgets = wm8750_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: WM8903: Add of_match_table
by Stephen Warren 07 Dec '11

07 Dec '11

This allows the device to be matched against the device tree using the compatible flag directly, as is standard, rather than falling back to matching .id_table against the non-vendor portion of the first compatible property value. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/wm8903.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index adfbefa..7a14210 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2205,6 +2205,12 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id wm8903_of_match[] = { + { .compatible = "wlf,wm8903", }, + {}, +}; +MODULE_DEVICE_TABLE(of, wm8903_of_match); + static const struct i2c_device_id wm8903_i2c_id[] = { { "wm8903", 0 }, { } @@ -2215,6 +2221,7 @@ static struct i2c_driver wm8903_i2c_driver = { .driver = { .name = "wm8903", .owner = THIS_MODULE, + .of_match_table = wm8903_of_match, }, .probe = wm8903_i2c_probe, .remove = __devexit_p(wm8903_i2c_remove), -- 1.7.0.4

2 1

[alsa-devel] Support for GT440 HDMI audio
by Andrey Tataranovich 07 Dec '11

07 Dec '11

Hi, Please include following patch to add support for NVidia GT440 HDMI audio. This patch working for me on Debian Squeeze with patched alsa-source 1.0.23-dfsg+4. -- WBR, Andrey Tataranovich

2 2

[alsa-devel] [PATCH 1/5] ASoC: Hold runtime PM references to components of active DAIs
by Mark Brown 07 Dec '11

07 Dec '11

Every device that implements runtime power management for DAIs is doing it in pretty much the same way: in the startup callback they take a runtime PM reference and then in the shutdown callback they release that reference, keeping the device active while the DAI is active. Given the frequency with which this is done and the obviousness of the need to keep the device active in this period factor the code out into the core, taking references on the device for each CPU DAI, CODEC DAI and DMA device in the core. As runtime PM is reference counted this shouldn't interfere with any other reference holding by the drivers, and since (in common with the existing implementations) we don't check for errors on enabling it shouldn't matter if the device actually has runtime PM enabled or not. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 15 +++++++++++++++ 1 files changed, 15 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 49aa71e..8aa7cec 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -19,6 +19,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/pm_runtime.h> #include <linux/slab.h> #include <linux/workqueue.h> #include <sound/core.h> @@ -77,6 +78,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); + pm_runtime_get_sync(codec_dai->dev); + pm_runtime_get_sync(platform->dev); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* startup the audio subsystem */ @@ -233,6 +238,11 @@ platform_err: cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: mutex_unlock(&rtd->pcm_mutex); + + pm_runtime_put(platform->dev); + pm_runtime_put(codec_dai->dev); + pm_runtime_put(cpu_dai->dev); + return ret; } @@ -339,6 +349,11 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } mutex_unlock(&rtd->pcm_mutex); + + pm_runtime_put(platform->dev); + pm_runtime_put(codec_dai->dev); + pm_runtime_put(cpu_dai->dev); + return 0; } -- 1.7.7.3

2 14

[alsa-devel] [PATCH] ASoC: Convert Samsung I2S driver to devm_kzalloc()
by Mark Brown 07 Dec '11

07 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/i2s.c | 20 ++++++-------------- 1 files changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 5de500c..ff5d919 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -945,7 +945,7 @@ struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) { struct i2s_dai *i2s; - i2s = kzalloc(sizeof(struct i2s_dai), GFP_KERNEL); + i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL); if (i2s == NULL) return NULL; @@ -972,10 +972,8 @@ struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->pdev = platform_device_register_resndata(NULL, pdev->name, pdev->id + SAMSUNG_I2S_SECOFF, NULL, 0, NULL, 0); - if (IS_ERR(i2s->pdev)) { - kfree(i2s); + if (IS_ERR(i2s->pdev)) return NULL; - } } /* Pre-assign snd_soc_dai_set_drvdata */ @@ -1048,7 +1046,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) if (!pri_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_pri\n"); ret = -ENOMEM; - goto err1; + goto err; } pri_dai->dma_playback.dma_addr = regs_base + I2STXD; @@ -1073,7 +1071,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); ret = -ENOMEM; - goto err2; + goto err; } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = @@ -1092,17 +1090,13 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); ret = -EINVAL; - goto err3; + goto err; } snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv); return 0; -err3: - kfree(sec_dai); -err2: - kfree(pri_dai); -err1: +err: release_mem_region(regs_base, resource_size(res)); return ret; @@ -1128,8 +1122,6 @@ static __devexit int samsung_i2s_remove(struct platform_device *pdev) i2s->pri_dai = NULL; i2s->sec_dai = NULL; - kfree(i2s); - snd_soc_unregister_dai(&pdev->dev); return 0; -- 1.7.7.3

2 1

Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Nallasellan, Singaravelan 06 Dec '11

06 Dec '11

> > The notion of a START threshold isn't self-explanatory for compressed data. The > decoder may or may not have enough data available for specific values of the > threshold, and the behavior would be decoder/implementation specific. > Besides, this was never implemented in the proprietary Intel API we've been using for > 2+ years and no one complained... Even the thread which actually writes the data to ring buffer may not have the idea of frames So the start threshold does not require to be on frame boundaries. It is just to avoid START when the buffer is written with the data. If the write gives more data than the ring buffer, write will not unblock and the user space application may not be able to call START. This will be an issue if you go for non-blocking implementation. Otherwise, START should be called in another thread than the thread which writes the data.

3 3

Re: [alsa-devel] [PATCH v2 5/5] ASoC: WM8903: Add device tree binding
by Stephen Warren 06 Dec '11

06 Dec '11

Rob Herring wrote at Tuesday, December 06, 2011 12:07 PM: > On 12/06/2011 12:22 PM, Stephen Warren wrote: > > Stephen Warren wrote at Friday, December 02, 2011 3:09 PM: > >> Document the device tree binding for the WM8903 codec, and modify the > >> driver to extract platform data from the device tree, if present. > > > > Mark, > > > > I just realized that when I was re-organizing all the WM8903 patches, I > > dropped the part that added the of_match table to the driver: > > > > +static const struct of_device_id wm8903_of_match[] __devinitconst = { > > + { .compatible = "wlf,wm8903", }, > > + {}, > > +}; > > +MODULE_DEVICE_TABLE(of, wm8903_of_match); > > > > Now, everything still works without this. Looking at the Linux OF code, > > it works by retrieving the compatible property, taking everything after > > the comma if present, and then creating an i2c_board_info with that > > type, which in this case is "wm8903" and matches wm8903.c's i2c_device_id > > table. See drivers/of/of_i2c.c:of_i2c_register_devices() and the call to > > base.c:of_modalias_node(). > > > > So, the question is: Should I go back and add the of_match table, or > > is I2C intended to work without it perpetually? I notice that you added > > an of_match table for all the other WM codecs. > > It definitely tries to match first with the OF match table, so it should > probably be added back. OK, I have no issues putting it back. But, I certainly can't find any code in drivers/i2c or drivers/of/of_i2c.c that matches the compatible property against the of_match_table. Can you point it out please. As best I can tell, I2C bus/controller drivers call of_i2c_register_devices() which calls of_modalias_node() which only operates in the manner I described. I don't think any generic instantiation code can be operating instead, since there's nothing in of_i2c_register_devices() that skips devices that have already been instantiated via other means. Nothing in drivers/i2c seems to touch of_node or the compatible property. -- nvpublic

1 0

[alsa-devel] [PATCH] ASoC: Don't fail if we can't read the IRQ type in WM8903
by Mark Brown 06 Dec '11

06 Dec '11

If we fail to read the IRQ type from the interrupt controller don't fail, just assume a value and solider on - we may fail later when we try to request the IRQ but it's possible we'll succeed. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 6 +----- 1 files changed, 1 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index adfbefa..21b9fdc 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2049,6 +2049,7 @@ static int wm8903_set_pdata_irq_trigger(struct i2c_client *i2c, switch (irqd_get_trigger_type(irq_data)) { case IRQ_TYPE_NONE: + default: /* * We assume the controller imposes no restrictions, * so we are able to select active-high @@ -2060,11 +2061,6 @@ static int wm8903_set_pdata_irq_trigger(struct i2c_client *i2c, case IRQ_TYPE_LEVEL_LOW: pdata->irq_active_low = true; break; - default: - dev_err(&i2c->dev, - "Unsupported IRQ_TYPE %x\n", - irqd_get_trigger_type(irq_data)); - return -EINVAL; } return 0; -- 1.7.7.3

2 1

[alsa-devel] ASoC: Build error if CONFIG_I2C=m && CONFIG_SND_SOC=y
by Axel Lin 06 Dec '11

06 Dec '11

Hi, I got below build error if CONFIG_I2C=m && CONFIG_SND_SOC=y. LD vmlinux sound/built-in.o: In function `snd_soc_codec_set_cache_io': last.c:(.text+0x2a0a8): undefined reference to `regmap_init_i2c' make: *** [vmlinux] Error 1 ( I test the build on linux-next 20111206 ) Regards, Axel

2 1

[alsa-devel] [PATCH] ASoC: Provide a more complete DMA driver stub
by Mark Brown 06 Dec '11

06 Dec '11

Allow userspace applications to do more parameter setting by providing a more complete stub DMA driver specifying a wildcard set of formats and channels and essentially random values for the DMA parameters. This is required for useful runtime operation of the dummy DMA driver until we are able to figure out how to power up links and do hw_params() from DAPM. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-utils.c | 31 ++++++++++++++++++++++++++++++- 1 files changed, 30 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 0c12b98..4220bb0 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -58,7 +58,36 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) } EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); -static struct snd_soc_platform_driver dummy_platform; +static const struct snd_pcm_hardware dummy_dma_hardware = { + .formats = 0xffffffff, + .channels_min = 1, + .channels_max = UINT_MAX, + + /* Random values to keep userspace happy when checking constraints */ + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .buffer_bytes_max = 128*1024, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = PAGE_SIZE*2, + .periods_min = 2, + .periods_max = 128, +}; + +static int dummy_dma_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware); + + return 0; +} + +static struct snd_pcm_ops dummy_dma_ops = { + .open = dummy_dma_open, + .ioctl = snd_pcm_lib_ioctl, +}; + +static struct snd_soc_platform_driver dummy_platform = { + .ops = &dummy_dma_ops, +}; static __devinit int snd_soc_dummy_probe(struct platform_device *pdev) { -- 1.7.7.3

3 3

Re: [alsa-devel] [PATCH] ALSA: ctxf: Use kcalloc instead of kzalloc to allocate array
by Takashi Iwai 06 Dec '11

06 Dec '11

At Tue, 29 Nov 2011 22:08:00 +0100, Thomas Meyer wrote: > > The advantage of kcalloc is, that will prevent integer overflows which could > result from the multiplication of number of elements and size and it is also > a bit nicer to read. > > The semantic patch that makes this change is available > in https://lkml.org/lkml/2011/11/25/107 > > Signed-off-by: Thomas Meyer <thomas(a)m3y3r.de> Thanks, applied both this and asihpi patches. Takashi > --- > > diff -u -p a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c > --- a/sound/pci/ctxfi/ctsrc.c 2011-11-13 11:08:15.090921741 +0100 > +++ b/sound/pci/ctxfi/ctsrc.c 2011-11-28 20:14:46.651112893 +0100 > @@ -437,7 +437,7 @@ get_src_rsc(struct src_mgr *mgr, const s > > /* Allocate mem for master src resource */ > if (MEMRD == desc->mode) > - src = kzalloc(sizeof(*src)*desc->multi, GFP_KERNEL); > + src = kcalloc(desc->multi, sizeof(*src), GFP_KERNEL); > else > src = kzalloc(sizeof(*src), GFP_KERNEL); > >

1 0

[alsa-devel] ASoC updates for 3.2
by Mark Brown 06 Dec '11

06 Dec '11

The following changes since commit bda63586bc5929e97288cdb371bb6456504867ed: firmware: Sigma: Fix endianess issues (2011-11-29 11:59:50 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.2 Mostly the usual small driver fixes - the DMA driver stub is really the same thing, it's just for a stub driver that is provided by the core rather than for an actual driver. Axel Lin (4): ASoC: kirkwood: Make SND_KIRKWOOD_SOC_OPENRD and SND_KIRKWOOD_SOC_T5325 depend on I2C ASoC: uda1380: Return proper error in uda1380_modinit failure path ASoC: Fix dependency for SND_SOC_RAUMFELD and SND_PXA2XX_SOC_HX4700 ASoC: Make SND_SOC_MX27VIS_AIC32X4 depend on I2C Mark Brown (3): ASoC: Mark WM8994 ADC muxes as virtual ASoC: Remove references to corgi and spitz from machine driver document ASoC: Provide a more complete DMA driver stub Documentation/sound/alsa/soc/machine.txt | 6 +--- sound/soc/codecs/uda1380.c | 4 +- sound/soc/codecs/wm8994.c | 12 +++++----- sound/soc/imx/Kconfig | 2 +- sound/soc/kirkwood/Kconfig | 3 +- sound/soc/pxa/Kconfig | 3 +- sound/soc/soc-utils.c | 31 +++++++++++++++++++++++++++++- 7 files changed, 45 insertions(+), 16 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: Include linux/io.h for jz4740 codec
by Axel Lin 06 Dec '11

06 Dec '11

Include linux/io.h to fix below build errors: CC sound/soc/codecs/jz4740.o sound/soc/codecs/jz4740.c: In function 'jz4740_codec_read': sound/soc/codecs/jz4740.c:82: error: implicit declaration of function 'readl' sound/soc/codecs/jz4740.c: In function 'jz4740_codec_write': sound/soc/codecs/jz4740.c:92: error: implicit declaration of function 'writel' sound/soc/codecs/jz4740.c: In function 'jz4740_codec_probe': sound/soc/codecs/jz4740.c:373: error: implicit declaration of function 'ioremap' sound/soc/codecs/jz4740.c:373: warning: assignment makes pointer from integer without a cast sound/soc/codecs/jz4740.c:393: error: implicit declaration of function 'iounmap' make[3]: *** [sound/soc/codecs/jz4740.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/jz4740.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 517e2a5..d73d283 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> +#include <linux/io.h> #include <linux/delay.h> -- 1.7.5.4

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[alsa-devel] [RFC][PATCH] ASoC: Fix build dependency for SND_SOC_JZ4740_CODEC
by Axel Lin 06 Dec '11

06 Dec '11

Currently SND_SOC_JZ4740_CODEC depends on SOC_JZ4740 but SOC_JZ4740 is not defined in any Kconfig. Thus the codec driver will not be built when select "Build all ASoC CODEC drivers". (Unless it is selected by SND_JZ4740_SOC_QI_LB60). Remove the dependency with SOC_JZ4740, then this code driver can be built when select "Build all ASoC CODEC drivers". Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I'm a little bit doubt if I should just remove the dependency as this patch does. Or alternatively make it depends on MACH_JZ4740. Comments? Axel sound/soc/codecs/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bc2364ac..7c205e7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,7 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 - select SND_SOC_JZ4740_CODEC if SOC_JZ4740 + select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98095 if I2C -- 1.7.5.4

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[alsa-devel] ASoC updates for 3.3
by Mark Brown 06 Dec '11

06 Dec '11

The following changes since commit fd8f2e496f38264a64c0cc7948217659019a9e8a: Merge branch 'for-3.2' into for-3.3 (2011-11-28 23:18:39 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 Lots of mechanical updates here so a large diffstat. Axel Lin (10): ASoC: Convert smdk_wm8994pcm to use module_platform_driver() ASoC: cs42l73: Fix clear wrong bits in cs42l73_set_dai_fmt ASoC: kirkwood: Make SND_KIRKWOOD_SOC_OPENRD and SND_KIRKWOOD_SOC_T5325 depend on I2C ASoC: Make SND_SOC_LITTLEMILL select MFD_WM8994 ASoC: uda1380: Return proper error in uda1380_modinit failure path ASoC: Fix dependency for SND_SOC_RAUMFELD and SND_PXA2XX_SOC_HX4700 ASoC: Make SND_SOC_MX27VIS_AIC32X4 depend on I2C ASoC: Convert e740_wm9705 to use gpio_request_one() ASoC: uda1380: Convert to gpio_request_one() ASoC: Fix reg_cache_size for stac9766 Lars-Peter Clausen (8): firmware: Sigma: Prevent out of bounds memory access firmware: Sigma: Skip header during CRC generation firmware: Sigma: Fix endianess issues ASoC: Move SigmaDSP firmware loader to ASoC ASoC: SigmaDSP: Provide diagnostic error messages ASoC: SigmaDSP: Move private structs and functions to C file ASoC: SigmaDSP: Add regmap support ASoC: Drop unused state parameter from CODEC suspend callback Mark Brown (58): Merge branch 'for-3.2' into for-3.3 ASoC: Ensure SYSCLK is enabled for WM8958 accessory detection ASoC: Fix __iomem annotation for IDMA registers ASoC: Put WM8958 and WM1811 MICBIAS into bypass mode when no audio ASoC: Enhance default WM8958 microphone detection ASoC: Rename Speyside WM8962 to Tobermory ASoC: Provide debug log of accessory status on WM8958 ASoC: Don't use control_data to get struct wm8994 ASoC: Tune down active mode detection rate for WM8958 mic detection ASoC: Allow more WM8958/WM1811 button levels with default handler ASoC: Rename WM8994 detecting flag to mic_detecting ASoC: Implement support for WM1811A jack detection ASoC: Ensure we reconfigure WM8958 microphone detection on rate changes ASoC: Add platform data for WM8958/WM1811 microphone detection rates ASoC: Convert WM8994 to devm_kzalloc() ASoC: Add signal generator widget type ASoC: Make WM5100 tone generator widgets signal generators ASoC: Make WM8962 beep a signal generator ASoC: Add missing err label ASoC: Add WM8958 based headset detection on Littlemill ASoC: Convert WM8994 MICBIASes to supply widgets ASoC: Map microphones on Littlemill ASoC: Mark WM8994 ADC muxes as virtual Merge branch 'for-3.2' into for-3.3 ASoC: Convert WM8903 to devm_kzalloc() ASoC: Use table based control init for WM8903 ASoC: WM8903 only supports I2C so don't ifdef it ASoC: Use a normal cache sync for WM8903 ASoC: Don't resync WM8903 register cache on reset ASoC: Convert WM8903 to direct regmap API usage ASoC: Move initial WM8903 identification and reset to I2C probe Merge branch 'for-3.2' into for-3.3 ASoC: Remove unused -codec from Wolfson device driver names ASoC: Remove I2C ifdefs from WM8960 ASoC: Remove unused AUDIO_NAME define from WM8960 ASoC: Convert WM2000 to devm_kzalloc() ASoC: Convert WM8350 to devm_kzalloc() ASoC: Convert WM8400 to devm_kzalloc() ASoC: Convert WM8741 to devm_kzalloc() ASoC: Convert WM8750 to devm_kzalloc() ASoC: Convert WM9090 to devm_kzalloc() ASoC: Convert WM8350 to table based DAPM and control init ASoC: Convert WM8400 to table based DAPM and control init ASoC: Convert WM8510 to table based DAPM and control init ASoC: Convert WM8580 to table based DAPM and control init ASoC: Convert WM8741 to table based DAPM and control init ASoC: Convert WM8750 to table based DAPM and control init ASoC: Convert WM8711 to table based control init ASoC: Convert WM8728 to table based control init ASoC: Remove unused struct wm2000_setup_data ASoC: Convert wm2000 to use regmap API ASoC: Convert WM2000 into a standard CODEC driver ASoC: Sort WM9090 in with the CODEC drivers Merge branch 'for-3.2' into for-3.3 ASoC: Remove references to corgi and spitz from machine driver document Merge branch 'for-3.2' into for-3.3 ASoC: Provide a more complete DMA driver stub Merge branch 'for-3.2' into for-3.3 Stephen Warren (3): ASoC: Tegra I2S: Add device tree binding ASoC: WM8903: Disallow all invalid gpio_cfg pdata values ASoC: WM8903: Create default platform data structure Timur Tabi (2): ASoC: p1022ds: add support for fsl,P1022 and fsl,P1022DS model names ASoC: fsl/powerpc: don't rely on the cell-index property Vinod Koul (1): ASoC: sst_platform: fix the dsp driver interface .../devicetree/bindings/sound/tegra20-i2s.txt | 17 + Documentation/sound/alsa/soc/machine.txt | 6 +- MAINTAINERS | 1 + arch/arm/mach-s3c64xx/mach-crag6410.c | 6 +- drivers/firmware/Kconfig | 12 - drivers/firmware/Makefile | 1 - drivers/firmware/sigma.c | 118 ----- include/linux/mfd/wm8994/pdata.h | 23 + include/linux/mfd/wm8994/registers.h | 16 + include/linux/sigma.h | 60 --- include/sound/soc-dapm.h | 4 + include/sound/soc.h | 3 +- sound/soc/codecs/Kconfig | 18 +- sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/cs42l73.c | 5 +- sound/soc/codecs/jz4740.c | 2 +- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sigmadsp.c | 246 ++++++++++ sound/soc/codecs/sigmadsp.h | 21 + sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/stac9766.c | 5 +- sound/soc/codecs/tlv320aic23.c | 3 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/uda134x.c | 3 +- sound/soc/codecs/uda1380.c | 34 +- sound/soc/codecs/wm2000.c | 226 ++++----- sound/soc/codecs/wm2000.h | 7 - sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8350.c | 49 +-- sound/soc/codecs/wm8400.c | 40 +- sound/soc/codecs/wm8510.c | 27 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 24 +- sound/soc/codecs/wm8711.c | 7 +- sound/soc/codecs/wm8727.c | 2 +- sound/soc/codecs/wm8728.c | 7 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 40 +- sound/soc/codecs/wm8750.c | 37 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 6 +- sound/soc/codecs/wm8903.c | 478 +++++++++--------- sound/soc/codecs/wm8904.c | 4 +- sound/soc/codecs/wm8940.c | 4 +- sound/soc/codecs/wm8955.c | 4 +- sound/soc/codecs/wm8960.c | 12 +- sound/soc/codecs/wm8961.c | 4 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 4 +- sound/soc/codecs/wm8974.c | 4 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 4 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 4 +- sound/soc/codecs/wm8994.c | 517 +++++++++++++++++--- sound/soc/codecs/wm8994.h | 11 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 9 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 3 +- sound/soc/codecs/wm9713.c | 3 +- sound/soc/codecs/wm_hubs.c | 4 +- sound/soc/fsl/mpc8610_hpcd.c | 13 +- sound/soc/fsl/p1022_ds.c | 49 ++- sound/soc/imx/Kconfig | 2 +- sound/soc/kirkwood/Kconfig | 3 +- sound/soc/mid-x86/Kconfig | 1 - sound/soc/mid-x86/sst_platform.c | 130 ++++-- sound/soc/mid-x86/sst_platform.h | 82 +++- sound/soc/pxa/Kconfig | 3 +- sound/soc/pxa/e740_wm9705.c | 20 +- sound/soc/samsung/Kconfig | 5 +- sound/soc/samsung/Makefile | 4 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/idma.c | 2 +- sound/soc/samsung/idma.h | 2 +- sound/soc/samsung/littlemill.c | 25 + sound/soc/samsung/smdk_wm8994pcm.c | 14 +- .../soc/samsung/{speyside_wm8962.c => tobermory.c} | 66 ++-- sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 7 + sound/soc/soc-utils.c | 31 ++- sound/soc/tegra/tegra_i2s.c | 27 +- 112 files changed, 1664 insertions(+), 1065 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra20-i2s.txt delete mode 100644 drivers/firmware/sigma.c delete mode 100644 include/linux/sigma.h create mode 100644 sound/soc/codecs/sigmadsp.c create mode 100644 sound/soc/codecs/sigmadsp.h rename sound/soc/samsung/{speyside_wm8962.c => tobermory.c} (72%)

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[alsa-devel] [PATCH] sound: soc: samsung: neo1973_wm8753: remove references to machine_is_neo1973_gta01
by Denis 'GNUtoo' Carikli 06 Dec '11

06 Dec '11

The Openmoko GTA01 machine has been removed from the machine ID database, so we need to remove references to it as well. Without that fix we have: sound/soc/samsung/neo1973_wm8753.c: In function ‘neo1973_wm8753_init’: sound/soc/samsung/neo1973_wm8753.c:325:2: error: implicit declaration of function ‘machine_is_neo1973_gta01’ Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo(a)no-log.org> --- sound/soc/samsung/neo1973_wm8753.c | 6 +----- 1 files changed, 1 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7207189..4d35297 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -322,10 +322,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) int ret; /* set up NC codec pins */ - if (machine_is_neo1973_gta01()) { - snd_soc_dapm_nc_pin(dapm, "LOUT2"); - snd_soc_dapm_nc_pin(dapm, "ROUT2"); - } snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "OUT4"); snd_soc_dapm_nc_pin(dapm, "LINE1"); @@ -479,7 +475,7 @@ static int __init neo1973_init(void) { int ret; - if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02()) + if (!machine_is_neo1973_gta02()) return -ENODEV; if (machine_is_neo1973_gta02()) { -- 1.7.4.1

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[alsa-devel] Mp3 files by ALSA
by `/ET!-!iS!-! 06 Dec '11

06 Dec '11

Hi I want to know how mp3 files are handled by ALSA . how audio data is handled for 4.1 channel and 5.1 channel . Regards yethish

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Re: [alsa-devel] [PATCH 4/6] compress: add API header and driver header files
by Mark Brown 06 Dec '11

06 Dec '11

On Mon, Dec 05, 2011 at 08:14:13AM -0600, Pierre-Louis Bossart wrote: > The answer has already been provided. I agree with Mark there's no reason > why this API would need to provide resource management capabilities, it'd > need to be handled with a separate interface addressing PCM and effects as > well. In the mean time user-space can put product-specific restrictions on > concurrent decoding. Or implement graceful fallback to the CPU if it can't set up an offloaded stream.

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[alsa-devel] [PATCH] plug: Move fast_ops copy after slave configuration.
by Reilly Grant 06 Dec '11

06 Dec '11

The plug plugin copies the fast_ops structure and fast_op_arg from its slave PCM on initialization and after inserting a new plugin. This works unless the slave PCM is also an instance of the plug plugin because plug can change its fast_ops structure during _snd_pcm_hw_params. Instead of copying fast_ops in snd_pcm_plug_insert_plugins wait until the end of snd_pcm_plug_hw_params when the slave has been fully initialized. This fixes a crash when two instances of plug are instantiated back to back and the format conversion plugin inserted by the second is skipped by the first because the fast_ops pointer is not properly updated. --- src/pcm/pcm_plug.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index e9d2923..72456d5 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -659,8 +659,6 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, } if (err) { plug->gen.slave = new; - pcm->fast_ops = new->fast_ops; - pcm->fast_op_arg = new->fast_op_arg; } k++; } @@ -1049,6 +1047,9 @@ static int snd_pcm_plug_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) } snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); snd_pcm_unlink_appl_ptr(pcm, plug->req_slave); + + pcm->fast_ops = slave->fast_ops; + pcm->fast_op_arg = slave->fast_op_arg; snd_pcm_link_hw_ptr(pcm, slave); snd_pcm_link_appl_ptr(pcm, slave); return 0; -- 1.7.3.4

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[alsa-devel] [PATCH] ASoC: Remove references to corgi and spitz from machine driver document
by Mark Brown 05 Dec '11

05 Dec '11

They're not currently actively worked on, the hardware being rather obsolete by now. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/sound/alsa/soc/machine.txt | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 3e2ec9c..d50c14d 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -50,8 +50,7 @@ Machine DAI Configuration The machine DAI configuration glues all the codec and CPU DAIs together. It can also be used to set up the DAI system clock and for any machine related DAI initialisation e.g. the machine audio map can be connected to the codec audio -map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c -for examples. +map, unconnected codec pins can be set as such. struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. @@ -83,8 +82,7 @@ Machine Power Map The machine driver can optionally extend the codec power map and to become an audio power map of the audio subsystem. This allows for automatic power up/down of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack -sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for -details. +sockets in the machine init function. Machine Controls -- 1.7.7.3

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[alsa-devel] [PATCH 1/2] ASoC: Take a pm_runtime reference on DAPM devices that are enabled
by Mark Brown 05 Dec '11

05 Dec '11

As for PCMs take a runtime power management reference to devices that are in a non-off bias, avoiding the need to do this in individual drivers. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f47e869..1a9fa47 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -39,6 +39,7 @@ #include <linux/platform_device.h> #include <linux/jiffies.h> #include <linux/debugfs.h> +#include <linux/pm_runtime.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1213,6 +1214,9 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) /* If we're off and we're not supposed to be go into STANDBY */ if (d->bias_level == SND_SOC_BIAS_OFF && d->target_bias_level != SND_SOC_BIAS_OFF) { + if (d->dev) + pm_runtime_get_sync(d->dev); + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, @@ -1252,6 +1256,9 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF); if (ret != 0) dev_err(d->dev, "Failed to turn off bias: %d\n", ret); + + if (d->dev) + pm_runtime_put_sync(d->dev); } /* If we just powered up then move to active bias */ -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Fix reg_cache_size for stac9766
by Axel Lin 05 Dec '11

05 Dec '11

reg_cache_size is supposed to be the number of elements in the register cache, not the size in bytes. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/stac9766.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 5581953..da71615 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -380,7 +380,7 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { .remove = stac9766_codec_remove, .suspend = stac9766_codec_suspend, .resume = stac9766_codec_resume, - .reg_cache_size = sizeof(stac9766_reg), + .reg_cache_size = ARRAY_SIZE(stac9766_reg), .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = stac9766_reg, -- 1.7.5.4

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Re: [alsa-devel] [PATCH 4/6] compress: add API header and driver header files
by Nallasellan, Singaravelan 05 Dec '11

05 Dec '11

> > > + * @decoded: Frames decoded by DSP > > > + * @rendered: Frames rendered by DSP into a mixer or an audio > > > + output > > > + * @sampling_rate: sampling rate of audio */ struct > > snd_compr_tstamp { > > > + size_t copied_bytes; > > > + size_t copied_total; > > > + size_t decoded; > > > + size_t rendered; > > > + __u32 sampling_rate; > > > > Is it possible to track the rendering sampling rate in the driver > > itself? > > What do you mean? This is a streaming API, the actual rendering rate will depend on > what endpoint you play on. I guess the driver should know the sampling rate of the endpoint. If only the DSP knows about the endpoint sampling rate, this will make sense. > > > > +struct snd_compr_caps { > > > + __u32 num_codecs; > > > + __u32 min_fragment_size; > > > + __u32 max_fragment_size; > > > + __u32 min_fragments; > > > + __u32 max_fragments; > > > + __u32 codecs[MAX_NUM_CODECS]; > > > + __u32 reserved[11]; > > > > Can we add number of compress streams supported in DSP? > > You are confusing number of formats supported with number of streams supported. > The latter is exposed through the actual devices supported. No, DSP may be able to support only so many simultaneous streams at any point of time. Yet, DSP can support any number of formats. Though the device entry may be created for each compressed stream, how will the user space programs get the number of compressed streams supported?

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[alsa-devel] [PATCH] asoc: sst_platform: fix the dsp driver interface
by Vinod Koul 05 Dec '11

05 Dec '11

lower level drivers typically register with upper layers. So fix by exporting symbols from sst_platform driver for dsp driver to register to sst platform driver Now this driver doesnt depend on sst driver, so remove the dependency and the header files Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/soc/mid-x86/Kconfig | 1 - sound/soc/mid-x86/sst_platform.c | 130 ++++++++++++++++++++++++-------------- sound/soc/mid-x86/sst_platform.h | 82 ++++++++++++++++++++++-- 3 files changed, 158 insertions(+), 55 deletions(-) diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig index 2935042..61c10bf 100644 --- a/sound/soc/mid-x86/Kconfig +++ b/sound/soc/mid-x86/Kconfig @@ -1,7 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC - depends on SND_INTEL_SST select SND_SOC_SN95031 select SND_SST_PLATFORM help diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 3e78260..0012f67 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -31,10 +31,51 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include "../../../drivers/staging/intel_sst/intel_sst_ioctl.h" -#include "../../../drivers/staging/intel_sst/intel_sst.h" #include "sst_platform.h" +static struct sst_device *sst; +static DEFINE_MUTEX(sst_lock); + +int sst_register_dsp(struct sst_device *dev) +{ + BUG_ON(!dev); + if (!try_module_get(dev->dev->driver->owner)) + return -ENODEV; + mutex_lock(&sst_lock); + if (sst) { + pr_err("we already have a device %s\n", sst->name); + module_put(dev->dev->driver->owner); + mutex_unlock(&sst_lock); + return -EEXIST; + } + pr_debug("registering device %s\n", dev->name); + sst = dev; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_register_dsp); + +int sst_unregister_dsp(struct sst_device *dev) +{ + BUG_ON(!dev); + if (dev != sst) + return -EINVAL; + + mutex_lock(&sst_lock); + + if (!sst) { + mutex_unlock(&sst_lock); + return -EIO; + } + + module_put(sst->dev->driver->owner); + pr_debug("unreg %s\n", sst->name); + sst = NULL; + mutex_unlock(&sst_lock); + return 0; +} +EXPORT_SYMBOL_GPL(sst_unregister_dsp); + static struct snd_pcm_hardware sst_platform_pcm_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_DOUBLE | @@ -134,37 +175,34 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream) } static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct snd_sst_stream_params *param) + struct sst_pcm_params *param) { - param->uc.pcm_params.codec = SST_CODEC_TYPE_PCM; - param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; - param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; - param->uc.pcm_params.reserved = 0; - param->uc.pcm_params.sfreq = substream->runtime->rate; - param->uc.pcm_params.ring_buffer_size = - snd_pcm_lib_buffer_bytes(substream); - param->uc.pcm_params.period_count = substream->runtime->period_size; - param->uc.pcm_params.ring_buffer_addr = - virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->uc.pcm_params.period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", - param->uc.pcm_params.sfreq, param->uc.pcm_params.pcm_wd_sz); + param->codec = SST_CODEC_TYPE_PCM; + param->num_chan = (u8) substream->runtime->channels; + param->pcm_wd_sz = substream->runtime->sample_bits; + param->reserved = 0; + param->sfreq = substream->runtime->rate; + param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); + param->period_count = substream->runtime->period_size; + param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); + pr_debug("period_cnt = %d\n", param->period_count); + pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); } static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream = substream->runtime->private_data; - struct snd_sst_stream_params param = {{{0,},},}; - struct snd_sst_params str_params = {0}; + struct sst_pcm_params param = {0}; + struct sst_stream_params str_params = {0}; int ret_val; /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, &param); substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; - str_params.codec = param.uc.pcm_params.codec; + str_params.codec = param.codec; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { str_params.ops = STREAM_OPS_PLAYBACK; str_params.device_type = substream->pcm->device + 1; @@ -176,7 +214,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) pr_debug("Capture stream,Device %d\n", substream->pcm->device); } - ret_val = stream->sstdrv_ops->pcm_control->open(&str_params); + ret_val = stream->ops->open(&str_params); pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); if (ret_val < 0) return ret_val; @@ -215,7 +253,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.mad_substream = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->sstdrv_ops->pcm_control->device_control( + ret_val = stream->ops->device_control( SST_SND_STREAM_INIT, &stream->stream_info); if (ret_val) pr_err("control_set ret error %d\n", ret_val); @@ -228,7 +266,6 @@ static int sst_platform_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; struct sst_runtime_stream *stream; - int ret_val = 0; pr_debug("sst_platform_open called\n"); runtime = substream->runtime; @@ -237,27 +274,27 @@ static int sst_platform_open(struct snd_pcm_substream *substream) if (!stream) return -ENOMEM; spin_lock_init(&stream->status_lock); - stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.mad_substream = substream; - /* allocate memory for SST API set */ - stream->sstdrv_ops = kzalloc(sizeof(*stream->sstdrv_ops), - GFP_KERNEL); - if (!stream->sstdrv_ops) { - pr_err("sst: mem allocation for ops fail\n"); + + /* get the sst ops */ + mutex_lock(&sst_lock); + if (!sst) { + pr_err("no device available to run\n"); + mutex_unlock(&sst_lock); kfree(stream); - return -ENOMEM; + return -ENODEV; } - stream->sstdrv_ops->vendor_id = MSIC_VENDOR_ID; - stream->sstdrv_ops->module_name = SST_CARD_NAMES; - /* registering with SST driver to get access to SST APIs to use */ - ret_val = register_sst_card(stream->sstdrv_ops); - if (ret_val) { - pr_err("sst: sst card registration failed\n"); - kfree(stream->sstdrv_ops); + if (!try_module_get(sst->dev->driver->owner)) { + mutex_unlock(&sst_lock); kfree(stream); - return ret_val; + return -ENODEV; } + stream->ops = sst->ops; + mutex_unlock(&sst_lock); + + stream->stream_info.str_id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + stream->stream_info.mad_substream = substream; + /* allocate memory for SST API set */ runtime->private_data = stream; return snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -272,9 +309,8 @@ static int sst_platform_close(struct snd_pcm_substream *substream) stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) - ret_val = stream->sstdrv_ops->pcm_control->close(str_id); - unregister_sst_card(stream->sstdrv_ops); - kfree(stream->sstdrv_ops); + ret_val = stream->ops->close(str_id); + module_put(sst->dev->driver->owner); kfree(stream); return ret_val; } @@ -288,8 +324,8 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { - ret_val = stream->sstdrv_ops->pcm_control->device_control( - SST_SND_DROP, &str_id); + ret_val = stream->ops->device_control( + SST_SND_DROP, &str_id); return ret_val; } @@ -341,8 +377,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, default: return -EINVAL; } - ret_val = stream->sstdrv_ops->pcm_control->device_control(str_cmd, - &str_id); + ret_val = stream->ops->device_control(str_cmd, &str_id); if (!ret_val) sst_set_stream_status(stream, status); @@ -362,7 +397,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->sstdrv_ops->pcm_control->device_control( + ret_val = stream->ops->device_control( SST_SND_BUFFER_POINTER, str_info); if (ret_val) { pr_err("sst: error code = %d\n", ret_val); @@ -469,6 +504,7 @@ static struct platform_driver sst_platform_driver = { static int __init sst_soc_platform_init(void) { pr_debug("sst_soc_platform_init called\n"); + sst = NULL; return platform_driver_register(&sst_platform_driver); } module_init(sst_soc_platform_init); diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h index df37028..f04f4f7 100644 --- a/sound/soc/mid-x86/sst_platform.h +++ b/sound/soc/mid-x86/sst_platform.h @@ -42,14 +42,14 @@ #define SST_MIN_PERIODS 2 #define SST_MAX_PERIODS (1024*2) #define SST_FIFO_SIZE 0 -#define SST_CARD_NAMES "intel_mid_card" -#define MSIC_VENDOR_ID 3 +#define SST_CODEC_TYPE_PCM 1 -struct sst_runtime_stream { - int stream_status; - struct pcm_stream_info stream_info; - struct intel_sst_card_ops *sstdrv_ops; - spinlock_t status_lock; +struct pcm_stream_info { + int str_id; + void *mad_substream; + void (*period_elapsed) (void *mad_substream); + unsigned long long buffer_ptr; + int sfreq; }; enum sst_drv_status { @@ -60,4 +60,72 @@ enum sst_drv_status { SST_PLATFORM_DROPPED, }; +enum sst_controls { + SST_SND_ALLOC = 0x00, + SST_SND_PAUSE = 0x01, + SST_SND_RESUME = 0x02, + SST_SND_DROP = 0x03, + SST_SND_FREE = 0x04, + SST_SND_BUFFER_POINTER = 0x05, + SST_SND_STREAM_INIT = 0x06, + SST_SND_START = 0x07, + SST_MAX_CONTROLS = 0x07, +}; + +enum sst_stream_ops { + STREAM_OPS_PLAYBACK = 0, + STREAM_OPS_CAPTURE, +}; + +enum sst_audio_device_type { + SND_SST_DEVICE_HEADSET = 1, + SND_SST_DEVICE_IHF, + SND_SST_DEVICE_VIBRA, + SND_SST_DEVICE_HAPTIC, + SND_SST_DEVICE_CAPTURE, +}; + +/* PCM Parameters */ +struct sst_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u32 ring_buffer_size; + u32 period_count; /* period elapsed in samples*/ + u32 ring_buffer_addr; +}; + +struct sst_stream_params { + u32 result; + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct sst_pcm_params sparams; +}; + +struct sst_ops { + int (*open) (struct sst_stream_params *str_param); + int (*device_control) (int cmd, void *arg); + int (*close) (unsigned int str_id); +}; + +struct sst_runtime_stream { + int stream_status; + struct pcm_stream_info stream_info; + struct sst_ops *ops; + spinlock_t status_lock; +}; + +struct sst_device { + char *name; + struct device *dev; + struct sst_ops *ops; +}; + +int sst_register_dsp(struct sst_device *sst); +int sst_unregister_dsp(struct sst_device *sst); #endif -- 1.7.0.4

2 4

[alsa-devel] [PATCH] ASoC: Convert e740_wm9705 to use gpio_request_one()
by Axel Lin 05 Dec '11

05 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/e740_wm9705.c | 20 ++++++-------------- 1 files changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 35ed7eb8c..818dc57 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -146,29 +146,21 @@ static int __init e740_init(void) if (!machine_is_e740()) return -ENODEV; - ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp"); + /* Disable audio */ + ret = gpio_request_one(GPIO_E740_MIC_ON, GPIOF_OUT_INIT_LOW, "Mic amp"); if (ret) return ret; - ret = gpio_request(GPIO_E740_AMP_ON, "Output amp"); + ret = gpio_request_one(GPIO_E740_AMP_ON, GPIOF_OUT_INIT_LOW, + "Output amp"); if (ret) goto free_mic_amp_gpio; - ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power"); + ret = gpio_request_one(GPIO_E740_WM9705_nAVDD2, GPIOF_OUT_INIT_HIGH, + "Audio power"); if (ret) goto free_op_amp_gpio; - /* Disable audio */ - ret = gpio_direction_output(GPIO_E740_MIC_ON, 0); - if (ret) - goto free_apwr_gpio; - ret = gpio_direction_output(GPIO_E740_AMP_ON, 0); - if (ret) - goto free_apwr_gpio; - ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1); - if (ret) - goto free_apwr_gpio; - e740_snd_device = platform_device_alloc("soc-audio", -1); if (!e740_snd_device) { ret = -ENOMEM; -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2] ASoC: uda1380: Convert to gpio_request_one()
by Axel Lin 05 Dec '11

05 Dec '11

Using gpio_request_one can make the error handling simpler. Also remove a redundant "Failed to issue reset" error message. We already show the error message in uda1380_reset() error path. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/uda1380.c | 28 ++++++++-------------------- 1 files changed, 8 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 3a238cb..c1b05b7 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -732,27 +732,21 @@ static int uda1380_probe(struct snd_soc_codec *codec) return -EINVAL; if (gpio_is_valid(pdata->gpio_reset)) { - ret = gpio_request(pdata->gpio_reset, "uda1380 reset"); + ret = gpio_request_one(pdata->gpio_reset, GPIOF_OUT_INIT_LOW, + "uda1380 reset"); if (ret) goto err_out; - ret = gpio_direction_output(pdata->gpio_reset, 0); - if (ret) - goto err_gpio_reset_conf; } if (gpio_is_valid(pdata->gpio_power)) { - ret = gpio_request(pdata->gpio_power, "uda1380 power"); - if (ret) - goto err_gpio; - ret = gpio_direction_output(pdata->gpio_power, 0); + ret = gpio_request_one(pdata->gpio_power, GPIOF_OUT_INIT_LOW, + "uda1380 power"); if (ret) - goto err_gpio_power_conf; + goto err_free_gpio; } else { ret = uda1380_reset(codec); - if (ret) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_reset; - } + if (ret) + goto err_free_gpio; } INIT_WORK(&uda1380->work, uda1380_flush_work); @@ -776,13 +770,7 @@ static int uda1380_probe(struct snd_soc_codec *codec) return 0; -err_reset: -err_gpio_power_conf: - if (gpio_is_valid(pdata->gpio_power)) - gpio_free(pdata->gpio_power); - -err_gpio_reset_conf: -err_gpio: +err_free_gpio: if (gpio_is_valid(pdata->gpio_reset)) gpio_free(pdata->gpio_reset); err_out: -- 1.7.5.4

2 1

[alsa-devel] [PATCH 01/21] ASoC: Remove unused -codec from Wolfson device driver names
by Mark Brown 05 Dec '11

05 Dec '11

Devices that aren't MFDs don't need to distinguish this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8727.c | 2 +- sound/soc/codecs/wm8900.c | 4 ++-- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- 14 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 3a65571..9e92bbb 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -666,7 +666,7 @@ MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id); static struct i2c_driver wm8510_i2c_driver = { .driver = { - .name = "wm8510-codec", + .name = "wm8510", .owner = THIS_MODULE, .of_match_table = wm8510_of_match, }, diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index fad90a3..e817056 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -59,7 +59,7 @@ static int __devexit wm8727_remove(struct platform_device *pdev) static struct platform_driver wm8727_codec_driver = { .driver = { - .name = "wm8727-codec", + .name = "wm8727", .owner = THIS_MODULE, }, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6ac80cf..aa0302b 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1258,7 +1258,7 @@ static int __devexit wm8900_spi_remove(struct spi_device *spi) static struct spi_driver wm8900_spi_driver = { .driver = { - .name = "wm8900-codec", + .name = "wm8900", .owner = THIS_MODULE, }, .probe = wm8900_spi_probe, @@ -1302,7 +1302,7 @@ MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); static struct i2c_driver wm8900_i2c_driver = { .driver = { - .name = "wm8900-codec", + .name = "wm8900", .owner = THIS_MODULE, }, .probe = wm8900_i2c_probe, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index babca49..1149a93 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2564,7 +2564,7 @@ MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); static struct i2c_driver wm8904_i2c_driver = { .driver = { - .name = "wm8904-codec", + .name = "wm8904", .owner = THIS_MODULE, }, .probe = wm8904_i2c_probe, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 9f1cce8..4d2854b 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -779,7 +779,7 @@ MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); static struct i2c_driver wm8940_i2c_driver = { .driver = { - .name = "wm8940-codec", + .name = "wm8940", .owner = THIS_MODULE, }, .probe = wm8940_i2c_probe, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index ca38722..f02256e 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -1037,7 +1037,7 @@ MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id); static struct i2c_driver wm8955_i2c_driver = { .driver = { - .name = "wm8955-codec", + .name = "wm8955", .owner = THIS_MODULE, }, .probe = wm8955_i2c_probe, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ed2773f..b04f808 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1030,7 +1030,7 @@ MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); static struct i2c_driver wm8960_i2c_driver = { .driver = { - .name = "wm8960-codec", + .name = "wm8960", .owner = THIS_MODULE, }, .probe = wm8960_i2c_probe, diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index c058701..434413b 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1116,7 +1116,7 @@ MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id); static struct i2c_driver wm8961_i2c_driver = { .driver = { - .name = "wm8961-codec", + .name = "wm8961", .owner = THIS_MODULE, }, .probe = wm8961_i2c_probe, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index b01df56..0606484 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -724,7 +724,7 @@ MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id); static struct i2c_driver wm8971_i2c_driver = { .driver = { - .name = "wm8971-codec", + .name = "wm8971", .owner = THIS_MODULE, }, .probe = wm8971_i2c_probe, diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index e41f999..b6136bb 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -671,7 +671,7 @@ MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); static struct i2c_driver wm8974_i2c_driver = { .driver = { - .name = "wm8974-codec", + .name = "wm8974", .owner = THIS_MODULE, }, .probe = wm8974_i2c_probe, diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 608c672..fbff293 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -822,7 +822,7 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi) static struct spi_driver wm8988_spi_driver = { .driver = { - .name = "wm8988-codec", + .name = "wm8988", .owner = THIS_MODULE, }, .probe = wm8988_spi_probe, diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 58d7f0b..9a53dc4 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1417,7 +1417,7 @@ MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id); static struct i2c_driver wm8990_i2c_driver = { .driver = { - .name = "wm8990-codec", + .name = "wm8990", .owner = THIS_MODULE, }, .probe = wm8990_i2c_probe, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 780c24c..e386127 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1641,7 +1641,7 @@ MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id); static struct i2c_driver wm8993_i2c_driver = { .driver = { - .name = "wm8993-codec", + .name = "wm8993", .owner = THIS_MODULE, }, .probe = wm8993_i2c_probe, diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index f94c060..4d90c54 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -685,7 +685,7 @@ MODULE_DEVICE_TABLE(i2c, wm9090_id); static struct i2c_driver wm9090_i2c_driver = { .driver = { - .name = "wm9090-codec", + .name = "wm9090", .owner = THIS_MODULE, }, .probe = wm9090_i2c_probe, -- 1.7.7.3

2 22

[alsa-devel] ALSA Underrun / Silence playback
by Irfan shaikh 05 Dec '11

05 Dec '11

Hello devel, I am trying to implement s/w params for ALSA playback device such that there is SILENCE playback for UNDERRUN (-32). To achieve above i used below s/w params API's. But still i observe UNDERRUN in my AUDIO Playback. Please correct me if i am mising somenting. {When i do get boundry, i get some weird value 1572864000. Thus i tried to use alsa buffer size for threshold} /*************************************************************************************/ if (snd_pcm_sw_params_get_boundary (alsaParams->p_swparams, &boundary) < 0) { printf ("Could not gete snd_pcm_sw_params_get_boundary"); return NULL; } // threshold = boundary; threshold = alsaParams->periodsize * alsaParams->no_of_periods; if (snd_pcm_sw_params_set_stop_threshold (alsaParams->pcm_handle, alsaParams->p_swparams, threshold) < 0) { printf("Could not gete snd_pcm_sw_params_get_boundary"); return NULL; } if (snd_pcm_sw_params_set_silence_threshold (alsaParams->pcm_handle, alsaParams->p_swparams, 0) < 0 || snd_pcm_sw_params_set_silence_size (alsaParams->pcm_handle, alsaParams->p_swparams, boundary) < 0) /* play silence on XRUN */ { printf("Could not set silence params"); return NULL; } /*************************************************************************************/ Thanks in advance !! Regards,

2 4

[alsa-devel] ASoC: Dynamic CODEC DAI
by Patrick Lai 05 Dec '11

05 Dec '11

Problem overview: The CODEC which I am working on transports digital audio through SLIMBUS instead of I2S. The concept of CODEC DAI does not apply too well with SLIMBUS architecture as CODEC DAIs are typically defined base on a set of I2S digital audio interface(bit clock, world select, sd lines). On the CODEC, there are 10 digital capture ports. They can be independently configured as mono channels or can be run-time grouped together to function like multi-channel CODEC DAIs. On top of that, only some ports out of 10 ports can accept all analog mic inputs or digital mic inputs. Hence, even though the CODEC satisfies the concurrent use cases it was designed for, software would have to be articulate on grouping the ports in order to utilize all ports. So, I cannot simply code up CODEC DAI definitions in the CODEC driver for a particular machine. For now, I am looking for compile time grouping of these ports as use cases are known at the time machine is designed. Proposal: I want to minimize amount of change to soc framework at the same time meeting my requirement. Here is my proposed design In CODEC driver, I will declare codec dai for each port. Then, I will enhance ASoC framework to take a list of CODEC DAIs and pass list of CODEC DAIs back to callback function such as startup, hw_param, prepare, trigger. I believe this approach will not require massive amount of change to framework. Your feedback is appreciated. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

4 8

[alsa-devel] [PATCH 1/3] ASoC: uda1380: Return proper error in uda1380_modinit failure path
by Axel Lin 05 Dec '11

05 Dec '11

Return proper error for uda1380_modinit if i2c_add_driver() fails. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/uda1380.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6b933ef..3a238cb 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -863,13 +863,13 @@ static struct i2c_driver uda1380_i2c_driver = { static int __init uda1380_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&uda1380_i2c_driver); if (ret != 0) pr_err("Failed to register UDA1380 I2C driver: %d\n", ret); #endif - return 0; + return ret; } module_init(uda1380_modinit); -- 1.7.5.4

3 8

[alsa-devel] [PATCH] ASoC: Fix dependency for SND_SOC_RAUMFELD and SND_PXA2XX_SOC_HX4700
by Axel Lin 05 Dec '11

05 Dec '11

SND_SOC_RAUMFELD selects SND_SOC_CS4270 which needs CONFIG_I2C, and also selects SND_SOC_AK4104 which needs SPI_MASTER. Thus make SND_SOC_RAUMFELD depend on I2C && SPI_MASTER. Add depend on SPI_MASTER to fix below build error if CONFIG_SPI_MASTER is not selected. LD .tmp_vmlinux1 sound/built-in.o: In function `ak4104_spi_write': last.c:(.text+0x290cc): undefined reference to `spi_sync' sound/built-in.o: In function `ak4104_probe': last.c:(.text+0x292a0): undefined reference to `spi_write_then_read' sound/built-in.o: In function `ak4104_spi_probe': last.c:(.text+0x29398): undefined reference to `spi_setup' sound/built-in.o: In function `ak4104_init': last.c:(.init.text+0x4ec): undefined reference to `spi_register_driver' make: *** [.tmp_vmlinux1] Error 1 Add depend on I2C to fix below build error if CONFIG_I2C is not selected: CC sound/soc/codecs/cs4270.o sound/soc/codecs/cs4270.c: In function 'cs4270_i2c_probe': sound/soc/codecs/cs4270.c:657: error: implicit declaration of function 'i2c_smbus_read_byte_data' sound/soc/codecs/cs4270.c: In function 'cs4270_init': sound/soc/codecs/cs4270.c:730: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/cs4270.c: In function 'cs4270_exit': sound/soc/codecs/cs4270.c:736: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/cs4270.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 SND_PXA2XX_SOC_HX4700 selects SND_SOC_AK4641 which needs CONFIG_I2C. Thus make SND_PXA2XX_SOC_HX4700 depend on I2C. Add depend on I2C to fix below build error if CONFIG_I2C is not selected: CC sound/soc/codecs/ak4641.o sound/soc/codecs/ak4641.c: In function 'ak4641_modinit': sound/soc/codecs/ak4641.c:646: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/ak4641.c: In function 'ak4641_exit': sound/soc/codecs/ak4641.c:656: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/ak4641.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/Kconfig | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ffd2242..a0f7d3c 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -151,6 +151,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + depends on I2C && SPI_MASTER select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 @@ -159,7 +160,7 @@ config SND_SOC_RAUMFELD config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" - depends on SND_PXA2XX_SOC && MACH_H4700 + depends on SND_PXA2XX_SOC && MACH_H4700 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_AK4641 help -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Make SND_SOC_MX27VIS_AIC32X4 depend on I2C
by Axel Lin 05 Dec '11

05 Dec '11

SND_SOC_MX27VIS_AIC32X4 selects SND_SOC_TLV320AIC32X4, but SND_SOC_TLV320AIC32X4 needs CONFIG_I2C. So we need to make SND_SOC_MX27VIS_AIC32X4 depend on I2C. otherwise I got below build error if CONFIG_I2C is not selected. CC sound/soc/codecs/tlv320aic32x4.o sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_read': sound/soc/codecs/tlv320aic32x4.c:323: error: implicit declaration of function 'i2c_smbus_read_byte_data' sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_probe': sound/soc/codecs/tlv320aic32x4.c:641: error: 'i2c_master_send' undeclared (first use in this function) sound/soc/codecs/tlv320aic32x4.c:641: error: (Each undeclared identifier is reported only once sound/soc/codecs/tlv320aic32x4.c:641: error: for each function it appears in.) sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_modinit': sound/soc/codecs/tlv320aic32x4.c:763: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_exit': sound/soc/codecs/tlv320aic32x4.c:774: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/tlv320aic32x4.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index b133bfc..7383917 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -28,7 +28,7 @@ config SND_MXC_SOC_WM1133_EV1 config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" - depends on MACH_IMX27_VISSTRIM_M10 + depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 select SND_MXC_SOC_MX2 help -- 1.7.5.4

2 1

[alsa-devel] Clevo P170HM / Sager NP8170 audio
by Andrew Mahone 04 Dec '11

04 Dec '11

This laptop is detected as ALC892 audio (and by the codec ID this seems to be valid), but none of the configurations for it quite match this device. I've been working on documenting the connections with HDA Analyzer. alsa-info output: http://www.alsa-project.org/db/?f=b86e4c6447867fbd129a36d8a63e2f5caa836c52 The laptop has built-in 5.1, and 5.1 or 7.1 output via jacks. DAC NIDs: 0x02 front 0x03 rear 0x04 clfe 0x05 side internal mixers: 0x0c front 0x0d rear 0x0e clfe 0x0f side internal speaker pins: 0x14 front 0x15 "rear" (front of laptop) 0x16 clfe external ports: fixed route: 0x17 side can select any output: 0x18 mic / output 0x1a line-in / output 0x1b headphone I've been experimenting with patch_realtek.c to add the extra DAC NID, and have had some success with a 6-channel mode for the built-in speakers and an 8-channel mode that mutes all internal speakers and sets the jacks up for output. I still need to manually connect pins 0x18, 0x1a, 0x1b to the correct internal mixers for 7.1 to work, and headphone plugin does not mute internal speakers in 5.1 mode. I have not added a plain stereo mode for internal speakers, although it appears that i can probably configure the codec to mix the front channels to center and LFE. Are there any suggestions for what to do next? My current state as a patch against the linux-3.0.y repository is at http://pastie.org/2400649 -- Andrew Mahone andrew DOT mahone AT gmail DOT com Q: How many voters does it take to change a lightbulb? A: None - if voting could change anything they'd make it illegal

3 29

Re: [alsa-devel] snd-usb-audio - NI Traktor 10 - phono settings - partly solved
by Peter Dunker 04 Dec '11

04 Dec '11

Hi Daniel, I tried amixer contents, which looks like more info, but it was not obvious: numid=1,iface=MIXER,name='Direct Thru Channel A' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='Direct Thru Channel B' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='Direct Thru Channel C' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=4,iface=MIXER,name='Direct Thru Channel D' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=5,iface=MIXER,name='Phono Input Channel A' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=6,iface=MIXER,name='Phono Input Channel B' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='Phono Input Channel C' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=8,iface=MIXER,name='Phono Input Channel D' ; type=BOOLEAN,access=rw------,values=1 : values=on I used now amixer -c 1 numid=<1..8> off and it worked! The channels are on line level now: numid=1,iface=MIXER,name='Direct Thru Channel A' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=2,iface=MIXER,name='Direct Thru Channel B' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=3,iface=MIXER,name='Direct Thru Channel C' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=4,iface=MIXER,name='Direct Thru Channel D' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=5,iface=MIXER,name='Phono Input Channel A' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=6,iface=MIXER,name='Phono Input Channel B' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=7,iface=MIXER,name='Phono Input Channel C' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=8,iface=MIXER,name='Phono Input Channel D' ; type=BOOLEAN,access=rw------,values=1 : values=off I have 2 problems left: 1) The amixer shows only 8, but the card has 10 channels. It might be no problem, as 2 channels were always line, but I have to double check, as soon as I have the card in hand (currently remote). Any idea, in case the 9. and 10. channel are still phono, where and how to look to set these channels too. 2) When I connect 2 cards, how can I for sure discriminate which is which and assign certain plugins in asoundrc to a specific one. I plan to use more then one T10. I read people use the name of the card to avoid confusion, but they have no wall the same name. The problem is that the order of card numbers is given by the order the cards are connected and its not clear what happens when reboot, maybe every time something different. Any idea, how I could define for sure, which one is which one? Maybe with a serial number, but I haven't seen one. Thanks, Peter Message: 5 Date: Sat, 03 Dec 2011 17:37:05 +0100 From: Daniel Mack <zonque(a)gmail.com> Subject: Re: [alsa-devel] snd-usb-audio - NI Traktor 10 - phono settings To: alsa-devel(a)alsa-project.org Message-ID: <4EDA5031.1050502(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On 12/03/2011 05:02 AM, Peter Dunker wrote: > Hi, I am new to this list, so please excuse if my questions are not > at the right target group. No, this is the perfectly right place for asking such questions :) For the record, please make sure you send your follow-ups to the list as well. > I am using since a couple days the NI Traktor 10 USB audio sound card > that has 10 channels. I use Ubuntu 11.10. and I was able to capture > with arecord from the device out of the box all 10 channels and I was > able to make an asoundrc to make 10 devices each 1 channel of it > too. > > My problem is now that this device enables by default 4 of the 10 > channels as phono level input, which leads to clipping and corrupt > audio on these channels when I feed line level. NI provides an config > app for Mac and Windows to set phono or non phono to the channels > that you like. Unfortunately, these settings do not stay as you set > under win or mac. There is no such tool, but controls for all 4 analog stereo pairs to configure the "direct thru" modes and the phono pre-amps should be exported as generic amixer controls. > My question, is there anything in this snd-usb-audio driver module > that could help me set this phono on/off values or can you guys think > of any other Linux tool that I could use to set these fields? I also > noticed that some channels actually changed the phono setting while > connecting to win and back to linux. So if you guys have an idea how > to set these values so that they might stay even when connect to > Linux, this would be helpful too. Post the output of "amixer controls", please. If there is more than one card connected and the TA10 is not your default, you might need to pass the "-D" option to amixer as well. See the output of "cat /proc/asound/cards" for the right value to pass. With this output, it should be easy to find the correct ALSA control to use with "amixer cset". Daniel

2 1

[alsa-devel] [PATCH] ASoC: Make SND_SOC_LITTLEMILL select MFD_WM8994
by Axel Lin 04 Dec '11

04 Dec '11

SND_SOC_LITTLEMILL selects SND_SOC_WM8994, but SND_SOC_WM8994 needs MFD_WM8994. Thus we need to select MFD_WM8994 to fix below build error: LD .tmp_vmlinux1 sound/built-in.o: In function `wm8994_write': sound/soc/codecs/wm8994.c:201: undefined reference to `wm8994_reg_write' sound/built-in.o: In function `wm8994_read': sound/soc/codecs/wm8994.c:222: undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8994_resume': sound/soc/codecs/wm8994.c:2847: undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8994_codec_probe': sound/soc/codecs/wm8994.c:3501: undefined reference to `wm8994_reg_read' sound/soc/codecs/wm8994.c:3660: undefined reference to `wm8994_reg_read' sound/soc/codecs/wm8994.c:3672: undefined reference to `wm8994_reg_read' sound/built-in.o: In function `wm8958_dsp2_fw': sound/soc/codecs/wm8958-dsp2.c:154: undefined reference to `wm8994_bulk_write' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 09d636c..f3417f2 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -210,4 +210,5 @@ config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S + select MFD_WM8994 select SND_SOC_WM8994 -- 1.7.5.4

2 1

Re: [alsa-devel] [patch][saa7134] do not change mute state for capturing audio
by Mauro Carvalho Chehab 03 Dec '11

03 Dec '11

Em 24-07-2011 14:45, Stas Sergeev escreveu: > 23.07.2011 19:09, Mauro Carvalho Chehab wrote: >> > In this case, it will not be autounmuted after tuning. >> Hard to tell about your solution without seeing a patch. > OK, it turns out the automute code is already there, > but it doesn't work. The driver for some reasons > starts the scan on initialization, finds the carrier: > --- > saa7134[0]/audio: found PAL main sound carrier @ 6.000 MHz [3969/324] > --- > and, because of that, disables the automute. If the > real mute is not enabled at that point, you get the > white noise right away. The automute code works fine. Maybe you have a strong interference at the default tuning frequency, leading into saa7134 miss-detection. For saa7134 driver: 1) There is an audio carrier; 2) an application wants to listen audio; 3) the device is at automute mode. So, when there's an audio carrier, it will unmute the audio at streaming start. The driver is doing the _right_ thing by letting the audio to flow to PA, when it starts the capture. Unfortunately, on your specific device, starting audio capture also enables audio at the audio output pin. So, you're noticing a problem. People with a saa7134 device without alsa stream won't notice it (old devices). People with alsa stream without anything connected to the LINE OUT people aren't noticing it, if PA is not copying the saa7134 PCM IN stream to the sound card PCM out device (the default, for PA). So, only people that has saa7134 with alsa stream that opted to wire the saa7134 device to the sound card, and with a strong interference at the default tunning frequency (400 MHz) would notice a problem. > Since I have no idea why it finds some carrier, I can't > fix that in any way. Or, maybe, not to call the scan > on driver init? What will that break? Analog tuners need to be tuned at the device init on a high frequency according with their datasheets, otherwise the PLL may fail. > Anyway, as long as the automute code is broken, > we should either start fixing it, or fix PA, or fix mplayer... As I always said, the fix should be at PA, as it makes no sense to start capturing at saa7134 without first configuring it. So, or PA should be converted into a V4L2-aware application, in order to properly init the device (with seems to be the wrong approach) or it _SHOULD_NOT_ automatically enable capture on those devices, as this may cause undesired side effects, on the devices that have the capture pin directly wired to the output pin, witch seems to be the case of your device. PS.: it seems that you've removed Lennart/alsa people from the c/c. I'm re-adding them, as I'm expecting a fix from their side. > Dunno. I wonder how come so many bugs left unfixed > for so long, resulting in a white noise to people... You're the first one that reported it, and the code is there for _years_. So, this is not a commonly noticed problem at all. Mauro.

2 15

[alsa-devel] snd-usb-audio - NI Traktor 10 - phono settings
by Peter Dunker 03 Dec '11

03 Dec '11

Hi, I am new to this list, so please excuse if my questions are not at the right target group. First of all, awesome work of the alsa developer to provide these system and modules! I am using since a couple days the NI Traktor 10 USB audio sound card that has 10 channels. I use Ubuntu 11.10. and I was able to capture with arecord from the device out of the box all 10 channels and I was able to make an asoundrc to make 10 devices each 1 channel of it too. My problem is now that this device enables by default 4 of the 10 channels as phono level input, which leads to clipping and corrupt audio on these channels when I feed line level. NI provides an config app for Mac and Windows to set phono or non phono to the channels that you like. Unfortunately, these settings do not stay as you set under win or mac. My question, is there anything in this snd-usb-audio driver module that could help me set this phono on/off values or can you guys think of any other Linux tool that I could use to set these fields? I also noticed that some channels actually changed the phono setting while connecting to win and back to linux. So if you guys have an idea how to set these values so that they might stay even when connect to Linux, this would be helpful too. Thanks in advance for reading, Peter

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[alsa-devel] [PATCH] ASoC: kirkwood: Make SND_KIRKWOOD_SOC_OPENRD and SND_KIRKWOOD_SOC_T5325 depend on I2C
by Axel Lin 03 Dec '11

03 Dec '11

SND_KIRKWOOD_SOC_T5325 selects SND_SOC_ALC5623, but SND_SOC_ALC5623 needs CONFIG_I2C. So we need to make SND_KIRKWOOD_SOC_T5325 depend on I2C, otherwise I got below build error if CONFIG_I2C is not selected. CC sound/soc/codecs/alc5623.o sound/soc/codecs/alc5623.c: In function 'alc5623_i2c_probe': sound/soc/codecs/alc5623.c:1002: error: implicit declaration of function 'i2c_smbus_read_word_data' sound/soc/codecs/alc5623.c:1009: error: implicit declaration of function 'i2c_smbus_read_byte_data' sound/soc/codecs/alc5623.c: In function 'alc5623_modinit': sound/soc/codecs/alc5623.c:1096: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/alc5623.c: In function 'alc5623_modexit': sound/soc/codecs/alc5623.c:1108: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/alc5623.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Also fix the same issue for SND_KIRKWOOD_SOC_OPENRD. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/kirkwood/Kconfig | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 8f49e16..c62d715 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -12,6 +12,7 @@ config SND_KIRKWOOD_SOC_I2S config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) + depends on I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help @@ -20,7 +21,7 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && MACH_T5325 + depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help -- 1.7.5.4

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[alsa-devel] [PATCH 0/6] Random WM8903 updates
by Mark Brown 02 Dec '11

02 Dec '11

This is a random collection of WM8903 updates that I started coding because I was looking at the driver and spotted a few style issues but haven't actually tested yet. I'm pushing them out now mostly because Stephen is also working on the driver and I want to avoid collisions. There's also a patch to convert to devm_kzalloc() that I posted earlier but that's trivial enough. It's pretty nice that even though I added a readable register function for regmap listing every register in the chip the overall diffstat is neutral, mostly as a result of the more efficient cache specification format that regmap has. Mark Brown (6): ASoC: Use table based control init for WM8903 ASoC: WM8903 only supports I2C so don't ifdef it ASoC: Use a normal cache sync for WM8903 ASoC: Don't resync WM8903 register cache on reset ASoC: Convert WM8903 to direct regmap API usage ASoC: Move initial WM8903 identification and reset to I2C probe sound/soc/codecs/wm8903.c | 449 +++++++++++++++++++++++---------------------- 1 files changed, 225 insertions(+), 224 deletions(-)

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[alsa-devel] [PATCH] ASoC: Convert WM8903 to devm_kzalloc()
by Mark Brown 02 Dec '11

02 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 7 +++---- sound/soc/codecs/wm8960.c | 23 ++++++++++++++++------- 2 files changed, 19 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 12dae43..fa44c9a 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2281,7 +2281,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, struct wm8903_priv *wm8903; int ret; - wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL); + wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), + GFP_KERNEL); if (wm8903 == NULL) return -ENOMEM; @@ -2290,15 +2291,13 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); - if (ret < 0) - kfree(wm8903); + return ret; } static __devexit int wm8903_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ed2773f..4393394 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -14,6 +14,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -71,6 +72,7 @@ static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { struct wm8960_priv { enum snd_soc_control_type control_type; + void *control_data; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); struct snd_soc_dapm_widget *lout1; @@ -264,7 +266,7 @@ SND_SOC_DAPM_INPUT("RINPUT2"), SND_SOC_DAPM_INPUT("LINPUT3"), SND_SOC_DAPM_INPUT("RINPUT3"), -SND_SOC_DAPM_SUPPLY("MICB", WM8960_POWER1, 1, 0, NULL, 0), +SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0), SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0, wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)), @@ -573,8 +575,6 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); - /* Enable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -677,9 +677,6 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF | WM8960_VMID_MASK, 0); break; - case SND_SOC_BIAS_OFF: - snd_soc_cache_sync(codec); - break; default: break; } @@ -868,7 +865,7 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static const struct snd_soc_dai_ops wm8960_dai_ops = { +static struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, @@ -905,6 +902,16 @@ static int wm8960_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8960_resume(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -918,6 +925,7 @@ static int wm8960_probe(struct snd_soc_codec *codec) u16 reg; wm8960->set_bias_level = wm8960_set_bias_level_out3; + codec->control_data = wm8960->control_data; if (!pdata) { dev_warn(codec->dev, "No platform data supplied\n"); @@ -1007,6 +1015,7 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); wm8960->control_type = SND_SOC_I2C; + wm8960->control_data = i2c; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Map microphones on Littlemill
by Mark Brown 02 Dec '11

02 Dec '11

Littlemill has one analogue microphone on the board (connected to IN1LN) and an array of four DMICs connected to both DMICDAT lines. The biases can be selected by jumpers but pick the default jumper fit. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/littlemill.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 5d7680f..5cea59b 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -147,11 +147,21 @@ static struct snd_soc_dai_link littlemill_dai[] = { static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), + + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), }; static struct snd_soc_dapm_route audio_paths[] = { { "Headphone", NULL, "HPOUT1L" }, { "Headphone", NULL, "HPOUT1R" }, + + { "AMIC", NULL, "MICBIAS1" }, /* Default for AMICBIAS jumper */ + { "IN1LN", NULL, "AMIC" }, + + { "DMIC", NULL, "MICBIAS2" }, /* Default for DMICBIAS jumper */ + { "DMIC1DAT", NULL, "DMIC" }, + { "DMIC2DAT", NULL, "DMIC" }, }; static struct snd_soc_jack littlemill_headset; -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Convert WM8994 MICBIASes to supply widgets
by Mark Brown 02 Dec '11

02 Dec '11

There are some in tree systems using the driver but none use the MICBIAS widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index d1debfb..2a61094 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -610,8 +610,8 @@ SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), SND_SOC_DAPM_INPUT("IN2RP:VXRP"), -SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0), -SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Add WM8958 based headset detection on Littlemill
by Mark Brown 02 Dec '11

02 Dec '11

The board supports CODECs that won't work with this but the CODEC driver will check to see if it's running on the right chip for us. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/littlemill.c | 15 +++++++++++++++ 1 files changed, 15 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index d2a44ab..5d7680f 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -154,8 +154,11 @@ static struct snd_soc_dapm_route audio_paths[] = { { "Headphone", NULL, "HPOUT1R" }, }; +static struct snd_soc_jack littlemill_headset; + static int littlemill_late_probe(struct snd_soc_card *card) { + struct snd_soc_codec *codec = card->rtd[0].codec; struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; @@ -164,6 +167,18 @@ static int littlemill_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4 | SND_JACK_BTN_5, + &littlemill_headset); + if (ret) + return ret; + + /* This will check device compatibility itself */ + wm8958_mic_detect(codec, &littlemill_headset, NULL, NULL); + return 0; } -- 1.7.7.3

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Re: [alsa-devel] [PATCH 1/5] ASoC: omap-dmic: Add device tree bindings
by Mark Brown 02 Dec '11

02 Dec '11

On Fri, Dec 02, 2011 at 03:59:01PM +0100, Cousson, Benoit wrote: > On 12/2/2011 3:00 PM, Mark Brown wrote: > >At least the DMA bindings seem fairly well sorted - we just merged the > >Tegra audio bindings which define a Tegra property for the DMA request > >signal. There's a reasonable amount of variation in how these things > >get plumbed together. > Mmm, I missed that series, but I'm not sure it is doing the right thing. > - First the name is missing :-( > - Secondly a dma-channel is not a dma-request... So I'm not sure > what that series is supposed to add... The idea is that the thing going into the block is the request line, this is sufficient to identify to the DMA controller code which hardware is being talked about. > >This is also OK for clocks since they'll go through the clock API so > >nothing needs doing there either. > Yep, my point was about the DT churn that this will generate. Adding new things isn't churn in quite the same way - extending bindings is a more natural process than dropping old bits of binding. > At the end, the effort should be only at OMAP core level, so the > driver should not be affected if we decide to move hwmod stuff into > DT or not. Well, the binding will at least need to be updated. Though of course if we manage to ship the DTS for the OMAP with the kernel then it won't really matter if we churn the binding and we may as well not document it at all, treating it as an internal part of the kernel that happens to go through device tree.

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Re: [alsa-devel] [PATCH 1/5] ASoC: omap-dmic: Add device tree bindings
by Mark Brown 02 Dec '11

02 Dec '11

On Fri, Dec 02, 2011 at 02:31:13PM +0100, Cousson, Benoit wrote: > Even if the reg-names and interrupts-names are accepted, which is > still not obvious due to a little bit of resistance, we still do not Yeah, it seems like there's very little traction on any of the problems with legacy bindings that device tree has. > have dma request bindings, nor clock bindings, nor pin mux. > All these information are provided today by hwmod, so it will still > be there for a couple of more merge windows. At least the DMA bindings seem fairly well sorted - we just merged the Tegra audio bindings which define a Tegra property for the DMA request signal. There's a reasonable amount of variation in how these things get plumbed together. > The important point for me, it to avoid having to change the driver > whenever these bindings will be there. This will be OK today for reg > and interrupts since they both go through Linux resources. This is also OK for clocks since they'll go through the clock API so nothing needs doing there either. > >Could we define and implement the "real" version now (it should be reasonable to > >code blind except possibly for the DMA channel as it's such a direct > >mapping) with hwmod left in there as something that's legacy for old > >kernels? > I think DT churn is unavoidable due to the ongoing effort to add > clock, regulator, dma, pin mux into DT core. > I don't think we should wait to have all that ready to start > migrating to DT. Well, it does make the whole process a bit silly. If we're planning to discard hwmod it seems like we shouldn't be transitioning it into device tree in the first place and instead just add the new properties to the bindings as we go. If hwmod must go into the device tree somehow is it not possible to keep the hwmod data in its own binding (referencing the nodes it affects rather than being in their bindings).

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[alsa-devel] hda-jack-retask - a GUI for easy retasking of your HDA Intel codecs!
by David Henningsson 02 Dec '11

02 Dec '11

Something fun for friday afternoon? :-) Over the past few weeks I’ve been working (from time to time) with HDA-Jack-Retask, a small application that makes the kernel interface for jack retasking really easy to use. Although primarily meant for power users, it focuses on simplicity: Just select your codec, then select which pins you want to override and what you want them to override to: Headphones, Line Out, Mic, Line In, and so on. There are buttons for trying it out right away, and for making your override the boot-time default. And of course, a button for removing all overrides in case things did not go as planned. Consider it beta quality for now, and it’s one of those "won’t work for everyone" programs, but that’s mostly due to hardware and driver limitations. If you're running Ubuntu 11.10: You can install it by adding ppa:diwic/hda [1] and then installing the hda-jack-retask package. Start it by running “hda-jack-retask” in a terminal. Or from the command line: sudo add-ppa-repository ppa:diwic/hda sudo apt-get update sudo apt-get install hda-jack-retask hda-jack-retask & If you're running something else: 1) download the source tarball [2] 2) Ensure you have a build environment (gcc, make, sed, etc), gtk3 (and glib) development headers, and the "gksu" application all are installed. On debian platforms, these packages are known as "build-essential", "libgtk-3-dev" and "gksu". 3) make 4) ./hda-jack-retask & Enjoy! End user blog post is here [3]. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] https://launchpad.net/~diwic/+archive/hda [2] https://launchpad.net/~diwic/+archive/hda/+files/hda-jack-retask_0.20111129… [3] http://voices.canonical.com/david.henningsson/2011/11/29/turn-your-mic-jack…

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Re: [alsa-devel] [PATCH 2/3] sound/soc/lapis: add machine driver
by Mark Brown 02 Dec '11

02 Dec '11

On Fri, Dec 02, 2011 at 09:09:34PM +0900, Tomoya MORINAGA wrote: > 2011/12/2 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > The CPU side audio drivers are not there - there seemed to be some quite > > serious review issues with those. > Let me clarify your saying. > Does the above "audio drivers" mean i2s (of south bridge) driver or HD > audio (of north bridge) or another? I2S and audio DMA.

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[alsa-devel] [PATCH 1/2] ASoC: p1022ds: add support for fsl, P1022 and fsl, P1022DS model names
by Timur Tabi 02 Dec '11

02 Dec '11

Commit ab827d97 ("powerpc/85xx: Rework P1022DS device tree") renamed the the /model property of the P1022DS device tree from "fsl,P1022" to "fsl,P1022DS". To support both old and new device trees, the ASoC machine driver for the P1022DS needs to query the /model property and update the platform driver object dynamically. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/p1022_ds.c | 36 ++++++++++++++++++++++++++++-------- 1 files changed, 28 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 2c064a9..3091626 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -540,12 +540,6 @@ static struct platform_driver p1022_ds_driver = { .probe = p1022_ds_probe, .remove = __devexit_p(p1022_ds_remove), .driver = { - /* The name must match the 'model' property in the device tree, - * in lowercase letters, but only the part after that last - * comma. This is because some model properties have a "fsl," - * prefix. - */ - .name = "snd-soc-p1022", .owner = THIS_MODULE, }, }; @@ -559,13 +553,39 @@ static int __init p1022_ds_init(void) { struct device_node *guts_np; struct resource res; + const char *sprop; + + /* + * Check if we're actually running on a P1022DS. Older device trees + * have a model of "fsl,P1022" and newer ones use "fsl,P1022DS", so we + * need to support both. The SSI driver uses that property to link to + * the machine driver, so have to match it. + */ + sprop = of_get_property(of_find_node_by_path("/"), "model", NULL); + if (!sprop) { + pr_err("snd-soc-p1022ds: missing /model node"); + return -ENODEV; + } + + pr_debug("snd-soc-p1022ds: board model name is %s\n", sprop); - pr_info("Freescale P1022 DS ALSA SoC machine driver\n"); + /* + * The name of this board, taken from the device tree. Normally, this is a* + * fixed string, but some P1022DS device trees have a /model property of + * "fsl,P1022", and others have "fsl,P1022DS". + */ + if (strcasecmp(sprop, "fsl,p1022ds") == 0) + p1022_ds_driver.driver.name = "snd-soc-p1022ds"; + else if (strcasecmp(sprop, "fsl,p1022") == 0) + p1022_ds_driver.driver.name = "snd-soc-p1022"; + else + return -ENODEV; /* Get the physical address of the global utilities registers */ guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts"); if (of_address_to_resource(guts_np, 0, &res)) { - pr_err("p1022-ds: missing/invalid global utilities node\n"); + pr_err("snd-soc-p1022ds: missing/invalid global utils node\n"); + of_node_put(guts_np); return -EINVAL; } guts_phys = res.start; -- 1.7.4.4

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Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Nallasellan, Singaravelan 02 Dec '11

02 Dec '11

> > > A general note: it's not necessarily to split playback and capture > > devices. For example, rawmidi has a single device for full-duplex. > > In the case of PCM, we had to split them because of mmap. The mmap > > for the capture was supposed not to be read-only, some apps may want > > to add mark on the buffer. > > Beats me why an app would want to write anything into the capture ring buffer. > Do you have an example of this use case? Maybe there's a similar need for > compressed data. Thanks! > -Pierre > There was a requirement to add the timestamp at the beginning of each block or frame if the encoded data is in RAW format. If the encoder does add the timestamp or marker, then we don't need the write option.

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[alsa-devel] data write in alsa
by Vinod Koul 02 Dec '11

02 Dec '11

Hi Takashi, For alsa pcm we have two ways to write data, either we use write() system call or use ioctl SNDRV_PCM_IOCTL_WRITEI_FRAMES. Is there any specific reason why we have two methods and which one is preferred and why? This question is basically for compress API, should we add similar write method thru ioctls, if so why? -- ~Vinod

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Re: [alsa-devel] [PATCH 2/3] sound/soc/lapis: add machine driver
by Mark Brown 02 Dec '11

02 Dec '11

On Mon, Nov 21, 2011 at 01:08:51PM +0900, Tomoya MORINAGA wrote: > Add machine driver for LAPIS Semiconductor ML7213 IOH. This needs to come after you add the base platform support in the series - clearly this won't work without the CPU side drivers it depends on. As with the CODEC driver you really should be looking at the standards for modern drivers and following them. > @@ -60,6 +60,7 @@ source "sound/soc/s6000/Kconfig" > source "sound/soc/sh/Kconfig" > source "sound/soc/tegra/Kconfig" > source "sound/soc/txx9/Kconfig" > +source "sound/soc/lapis/Kconfig" Keep the Makefile and Kconfig entries sorted. > +static struct snd_soc_dai_link ioh_i2s_dai = { > + .name = "I2S", > + .stream_name = "I2S HiFi", > + .cpu_dai_name = "ml7213ioh-i2s", > + .codec_dai_name = "ml7213ioh-hifi", I'm very surprised this works... If nothing else I'd expec the CODEC DAI to have a different name to the CPU. > +static int __init ioh_i2s_probe(struct platform_device *pdev) > +{ > + int ret; > + > + soc_pdev = platform_device_alloc("soc-audio", -1); > + if (!soc_pdev) > + return -ENOMEM; You should register the card as a standard platform device for new drivers and use snd_soc_register_card(). > +static struct platform_driver ioh_i2s_driver = { > + .remove = ioh_i2s_remove, > + .driver = { > + .name = "ml7213ioh-machine", > + .owner = THIS_MODULE, > + }, > +}; You're already doing the platform device bit so it should be be a case of changing the body of probe() and remove(). > +static int __init ioh_i2s_init(void) > +{ > + return platform_driver_probe(&ioh_i2s_driver, > + ioh_i2s_probe); > +} Just make this a standard platform_driver_register().

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