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

Re: [alsa-devel] Regression 2.6.35-rc6: ALSA Intel HDA/Realtek: missing Beep
by Takashi Iwai 29 Jul '10

29 Jul '10

At Thu, 29 Jul 2010 10:09:30 +0200, Mario 'BitKoenig' Holbe wrote: > > On Thu, Jul 29, 2010 at 07:44:38AM +0200, Takashi Iwai wrote: > > Mario 'BitKoenig' Holbe wrote: > > > But the BIOS itself beeps through the sound-card at boot :/ > > But BIOS tells that the HD-audio codec shouldn't use, so the driver > > follows it. > > Even grub's beep (play 480 440 1) goes through the sound-card. So it > seems like the BIOS leaves everything set up working as well. Well, my point is that BIOS gives the wrong SSID value. It's bad that the value is compliant with Realtek codecs specification, but it clears the PC-beep bit explicitly. In most cases, SSID value isn't compliant (so BIOS gives a wrong but it's less wrong :), thus the driver takes a fallback to PC-beep enabled. > Please don't get me wrong. I'm not saying the driver does something > wrong, I'm sure it doesn't. I'm just sure that if I could convince it to > behave as if it would have detected the Beep pin everything would work > fine again because it did before... FYI, you can change SSID by writing /sys/class/sound/hwC*D*/subsystem_id file. Then reconfigure the codec via triggering /sys/class/sound/hwC*D*/reconfig. > > Please give alsa-info.sh output instead of codec proc file. It's more > > comprehensive. > > Attached. > > This is from a kernel with both patches applied you sent me. > > > With the patch below, you'll likely have back the system beep sound. > > Nope, no sound. But I guess this wasn't the intention of the patch. Now, > no beep input is registered anymore - which was the intention, I guess. It's the intention. If SSID set by BIOS doesn't indicate the presence of PC-speaker bit *explicitly*, we shouldn't touch beep stuff in codec. > > But it doesn't go through codec, thus no volume control. > > If you mean it should go through the 5V PC Speaker (i.e. pcspkr) - I > don't have such a thing connected. I always appreciated having volume- > and mute control over the beep at night when you can't get away from > work but don't like to wake up anybody just because command completion > beeps, because you pasted something in the wrong window, or whatever. For the driver, the outside connection doesn't matter. The information is given as the spec bit, and just follows it :) So, the fix is likely to override SSID value, or create a special quirk rule to enable PC-beep for known white-list, supposing BIOS won't be fixed in any future... Takashi

2 6

[PATCH 4/4] Rename iMic to Int Mic on Lenovo NB0763
by David Henningsson 29 Jul '10

29 Jul '10

The non-standard name "iMic" makes PulseAudio ignore the microphone. BugLink: https://launchpad.net/bugs/605101 --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1744d4d..b4b7f58 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7087,7 +7087,7 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, - { "iMic", 0x1 }, + { "Int Mic", 0x1 }, { "Line", 0x2 }, { "CD", 0x4 }, }, @@ -8657,8 +8657,8 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; -- 1.7.0.4 --------------030908020603010700040801--

1 0

[PATCH 4/4] Rename iMic to Int Mic on Lenovo NB0763
by David Henningsson 29 Jul '10

29 Jul '10

The non-standard name "iMic" makes PulseAudio ignore the microphone. BugLink: https://launchpad.net/bugs/605101 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1744d4d..b4b7f58 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7087,7 +7087,7 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, - { "iMic", 0x1 }, + { "Int Mic", 0x1 }, { "Line", 0x2 }, { "CD", 0x4 }, }, @@ -8657,8 +8657,8 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; -- 1.7.0.4 --------------000003000303050509090808--

1 0

Re: [alsa-devel] [PATCH 06/10 v2] ALSA: msnd: check request_region() return value
by Takashi Iwai 29 Jul '10

29 Jul '10

At Thu, 29 Jul 2010 14:45:24 +0400, Kulikov Vasiliy wrote: > > request_region() may fail, if so return -EBUSY. > > Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> Thanks, applied both patches now. Takashi > --- > sound/isa/msnd/msnd_pinnacle.c | 5 ++++- > 1 files changed, 4 insertions(+), 1 deletions(-) > > diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c > index 60b6abd..ed42f8e 100644 > --- a/sound/isa/msnd/msnd_pinnacle.c > +++ b/sound/isa/msnd/msnd_pinnacle.c > @@ -549,7 +549,10 @@ static int __devinit snd_msnd_attach(struct snd_card *card) > printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", chip->irq); > return err; > } > - request_region(chip->io, DSP_NUMIO, card->shortname); > + if (request_region(chip->io, DSP_NUMIO, card->shortname) == NULL) { > + free_irq(chip->irq, chip); > + return -EBUSY; > + } > > if (!request_mem_region(chip->base, BUFFSIZE, card->shortname)) { > printk(KERN_ERR LOGNAME > -- > 1.7.0.4 >

1 0

[alsa-devel] test
by Miroslav Šulc 29 Jul '10

29 Jul '10

hi, sorry for this mail but i already sent two emails here and none of them arrived to the mailing list (i did not receive it in my mailbox). miroslav

1 0

Re: [alsa-devel] [PATCH 10/10] ALSA: asihpi: check return value of get_user()
by Takashi Iwai 29 Jul '10

29 Jul '10

At Wed, 28 Jul 2010 20:41:56 +0400, Kulikov Vasiliy wrote: > > get_user() may fail, if so return -EFAULT. > > Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> Thanks, applied this one, too. Takashi > --- > sound/pci/asihpi/hpioctl.c | 17 +++++++++++++---- > 1 files changed, 13 insertions(+), 4 deletions(-) > > diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c > index 3114999..62895a7 100644 > --- a/sound/pci/asihpi/hpioctl.c > +++ b/sound/pci/asihpi/hpioctl.c > @@ -121,11 +121,17 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) > phpi_ioctl_data = (struct hpi_ioctl_linux __user *)arg; > > /* Read the message and response pointers from user space. */ > - get_user(puhm, &phpi_ioctl_data->phm); > - get_user(puhr, &phpi_ioctl_data->phr); > + if (get_user(puhm, &phpi_ioctl_data->phm) || > + get_user(puhr, &phpi_ioctl_data->phr)) { > + err = -EFAULT; > + goto out; > + } > > /* Now read the message size and data from user space. */ > - get_user(hm->h.size, (u16 __user *)puhm); > + if (get_user(hm->h.size, (u16 __user *)puhm)) { > + err = -EFAULT; > + goto out; > + } > if (hm->h.size > sizeof(*hm)) > hm->h.size = sizeof(*hm); > > @@ -138,7 +144,10 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) > goto out; > } > > - get_user(res_max_size, (u16 __user *)puhr); > + if (get_user(res_max_size, (u16 __user *)puhr)) { > + err = -EFAULT; > + goto out; > + } > /* printk(KERN_INFO "user response size %d\n", res_max_size); */ > if (res_max_size < sizeof(struct hpi_response_header)) { > HPI_DEBUG_LOG(WARNING, "small res size %d\n", res_max_size); > -- > 1.7.0.4 >

1 0

Re: [alsa-devel] [PATCH 08/10] oss: midi_synth: check get_user() return value
by Takashi Iwai 29 Jul '10

29 Jul '10

At Wed, 28 Jul 2010 20:41:17 +0400, Kulikov Vasiliy wrote: > > get_user() may fail, if so return -EFAULT. > > Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> Applied now. Thanks. Takashi > --- > sound/oss/midi_synth.c | 4 +++- > 1 files changed, 3 insertions(+), 1 deletions(-) > > diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c > index 3bc7104..3c09374 100644 > --- a/sound/oss/midi_synth.c > +++ b/sound/oss/midi_synth.c > @@ -523,7 +523,9 @@ midi_synth_load_patch(int dev, int format, const char __user *addr, > { > unsigned char data; > > - get_user(*(unsigned char *) &data, (unsigned char __user *) &((addr)[hdr_size + i])); > + if (get_user(data, > + (unsigned char __user *)(addr + hdr_size + i))) > + return -EFAULT; > > eox_seen = (i > 0 && data & 0x80); /* End of sysex */ > > -- > 1.7.0.4 >

1 0

Re: [alsa-devel] [PATCH 07/10] ALSA: sb: check get_user() return value
by Takashi Iwai 29 Jul '10

29 Jul '10

At Wed, 28 Jul 2010 20:40:51 +0400, Kulikov Vasiliy wrote: > > get_user() may fail, if so return -EFAULT. > > Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> Thanks, applied this one (with one additional fix for get_user() there). Takashi > --- > sound/isa/sb/emu8000_pcm.c | 6 ++++-- > 1 files changed, 4 insertions(+), 2 deletions(-) > > diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c > index ccedbfe..ae6fa9f 100644 > --- a/sound/isa/sb/emu8000_pcm.c > +++ b/sound/isa/sb/emu8000_pcm.c > @@ -433,7 +433,8 @@ static int emu8k_transfer_block(struct snd_emu8000 *emu, int offset, unsigned sh > while (count > 0) { > unsigned short sval; > CHECK_SCHEDULER(); > - get_user(sval, buf); > + if (get_user(sval, buf)) > + return -EFAULT; > EMU8000_SMLD_WRITE(emu, sval); > buf++; > count--; > @@ -525,7 +526,8 @@ static int emu8k_pcm_copy(struct snd_pcm_substream *subs, > while (count-- > 0) { > unsigned short sval; > CHECK_SCHEDULER(); > - get_user(sval, buf); > + if (get_user(sval, buf)) > + return -EFAULT; > EMU8000_SMLD_WRITE(emu, sval); > buf++; > if (rec->voices > 1) { > -- > 1.7.0.4 >

1 0

[alsa-devel] Fwd: acer aspire 8943g with alc670
by Miroslav Šulc 29 Jul '10

29 Jul '10

hello, i sent this post 6 hours back but it did not reach the mail list so trying again. miroslav -------- Původní zpráva -------- Předmět: acer aspire 8943g with alc670 Datum: Thu, 29 Jul 2010 04:09:37 +0200 Od: Miroslav Šulc <fordfrog(a)gmail.com> Komu: alsa-devel(a)alsa-project.org hello, i'd like to help with making the above soundcard to fully work, but have no clue what to do exactly. currently the soundcard works pretty weird, though at least sound goes out :-) i tried hda analyzer but did not find any documentation for it so have no idea what to do exactly. also, when clicking AUD_IN nodes, i get this error: ** (hda_analyzer.py:10063): WARNING **: column number is a boolean, but will be interpreted as an integer; this is likely not what you intended Traceback (most recent call last): File "/dev/shm/hda-analyzer/hda_analyzer.py", line 308, in selection_changed_cb self.__refresh() File "/dev/shm/hda-analyzer/hda_analyzer.py", line 276, in __refresh self.load() File "/dev/shm/hda-analyzer/hda_analyzer.py", line 336, in load self.node_window.add(NodeGui(node=n)) File "/dev/shm/hda-analyzer/hda_guilib.py", line 86, in __init__ self.__build_node(node, doframe) File "/dev/shm/hda-analyzer/hda_guilib.py", line 682, in __build_node vbox.pack_start(self.__build_amps(node), False, False) File "/dev/shm/hda-analyzer/hda_guilib.py", line 365, in __build_amps node.in_amp and node.amp_vals_in or None) File "/dev/shm/hda-analyzer/hda_guilib.py", line 328, in build_caps if ctl.hdactl.amp_index_match(idx): File "/dev/shm/hda-analyzer/hda_proc.py", line 170, in amp_index_match count = (self.amp_chs & 1) + ((self.amp_chs >> 1) & 1) TypeError: unsupported operand type(s) for &: 'NoneType' and 'int' can anybody pls direct me to some documentation that could help me to master hda analyzer and find the way to make the card work correctly? miroslav

1 0

Re: [alsa-devel] [PATCH 09/10] oss: msnd: check request_region() return value
by Takashi Iwai 29 Jul '10

29 Jul '10

At Wed, 28 Jul 2010 23:20:31 +0200, Dan Carpenter wrote: > > On Wed, Jul 28, 2010 at 10:00:02PM +0200, Dan Carpenter wrote: > > This should be -EBUSY as well. The same for "[PATCH 06/10] ALSA: msnd: > > check request_region() return value" > > > > Another way to write that would be: > > > > Gar. I was thinking of request_resource(). request_region() returns a > pointer of course. > > But still the return code should probably be -EBUSY. Resource > conflicts are more likely than allocation failures. Agreed. Kulikov, could you rewrite the patches with -EBUSY? thanks, Takashi > > regards, > dan carpenter > > > - request_region(dev.io, dev.numio, dev.name); > > + err = request_region(dev.io, dev.numio, dev.name); > > + if (err) { > > + free_irq(dev.irq, &dev); > > + return err; > > + } > > > > regards, > > dan carpenter > > > > > + if (request_region(dev.io, dev.numio, dev.name) == NULL) { > > > + free_irq(dev.irq, &dev); > > > + return -ENOMEM; > > > + } >

1 0

[alsa-devel] [PATCH v2 0/2] ASoC: tlv320dac33: FIFO mode updates
by Peter Ujfalusi 29 Jul '10

29 Jul '10

Hello, Changes since v1: - mode1_i2c_latecy renamed to mode1_latency Intro mail from the first series: The main feature of this series is the automatic FIFO configuration of the FIFO, when one of the FIFO mode is selected. The first patch simplifies the handling of the Mode1 in manual mode, and also removes the hardwired latency definition from the code, so it can be adjusted based on the underlying platform. The second patch adds option to select the automatic FIFO configuration. When this mode is selected the driver will configure the FIFO to match the stream's period size with the burst length (in terms of samples bursted out from the host). In Mode1 this is trivial, since I can configure the nSample to match with the period size. However the Mode7 is a bit more complicated. The formula (UTHR_FROM_PERIOD_SIZE) calculates the UTHR configuration for the given number of samples. The patches are generated against: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git:topic/asoc --- Peter Ujfalusi (2): ASoC: tlv320dac33: Revisit the FIFO Mode1 handling ASoC: tlv320dac33: Add support for automatic FIFO configuration include/sound/tlv320dac33-plat.h | 2 + sound/soc/codecs/tlv320dac33.c | 125 ++++++++++++++++++++++++-------------- 2 files changed, 82 insertions(+), 45 deletions(-) -- 1.7.2

3 4

[alsa-devel] [PATCH] asoc multi-component: mpc8610 hpcd updates
by Timur Tabi 29 Jul '10

29 Jul '10

Additional updates to the multi-component branch. Liam, please squash these in. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 68 +++++++++++++++-------- sound/soc/fsl/fsl_dma.h | 3 - sound/soc/fsl/fsl_ssi.c | 37 +++++++----- sound/soc/fsl/mpc8610_hpcd.c | 125 ++++++++++++++++++++++++++--------------- 4 files changed, 144 insertions(+), 89 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 7f87297..d09e194 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -32,6 +32,7 @@ #include <asm/io.h> #include "fsl_dma.h" +#include "fsl_ssi.h" /* For the offset of stx0 and srx0 */ /* * The formats that the DMA controller supports, which is anything @@ -798,36 +799,35 @@ static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) static LIST_HEAD(dma_list); /** - * fsl_soc_dma_to_dai - returning the DAI struct for a given DMA node - * - * When a machine driver registers itself with ASoC, it must provide the - * address of the DAI structures that it wants to connect. Most drivers simply - * reference global data structures in the various drivers, but we PowerPC - * developers are better than that. This driver keeps a list of every DMA - * node that it probes, and when the machine driver wants to reference a - * DMA DAI, this function provides that service. - * - * Hopefully one day, ASoC will just want the name of the DMA device, not - * some obscure pointer, and this function will go away. + * find_ssi_node -- returns the SSI node that points to his DMA channel node + * + * Although this DMA driver attempts to operate independently of the other + * devices, it still needs to determine some information about the SSI device + * that it's working with. Unfortunately, the device tree does not contain + * a pointer from the DMA channel node to the SSI node -- the pointer goes the + * other way. So we need to scan the device tree for SSI nodes until we find + * the one that points to the given DMA channel node. It's ugly, but at least + * it's contained in this one function. */ -struct snd_soc_platform_driver *fsl_soc_dma_to_dai(const char *path, - dma_addr_t ssi_stx_phys, dma_addr_t ssi_srx_phys) +static struct device_node *find_ssi_node(struct device_node *dma_channel_np) { - struct list_head *ptr; - struct dma_object *dma; + struct device_node *ssi_np, *np; - list_for_each(ptr, &dma_list) { - dma = list_entry(ptr, struct dma_object, list); - if (strcmp(path, dma->path) == 0) { - dma->ssi_stx_phys = ssi_stx_phys; - dma->ssi_srx_phys = ssi_srx_phys; - return &dma->dai; - } + for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { + /* Check each DMA phandle to see if it points to us. We + * assume that device_node pointers are a valid comparison. + */ + np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0); + if (np == dma_channel_np) + return ssi_np; + + np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0); + if (np == dma_channel_np) + return ssi_np; } return NULL; } -EXPORT_SYMBOL(fsl_soc_dma_to_dai); static struct snd_pcm_ops fsl_dma_ops = { .open = fsl_dma_open, @@ -843,8 +843,24 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, { struct dma_object *dma; struct device_node *np = of_dev->dev.of_node; + struct device_node *ssi_np; + struct resource res; int ret; + /* Find the SSI node that points to us. */ + ssi_np = find_ssi_node(np); + if (!ssi_np) { + dev_err(&of_dev->dev, "cannot find parent SSI node\n"); + return -ENODEV; + } + + ret = of_address_to_resource(ssi_np, 0, &res); + of_node_put(ssi_np); + if (ret) { + dev_err(&of_dev->dev, "could not determine device resources\n"); + return ret; + } + dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); if (!dma) { dev_err(&of_dev->dev, "could not allocate dma object\n"); @@ -856,6 +872,10 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, dma->dai.pcm_new = fsl_dma_new; dma->dai.pcm_free = fsl_dma_free_dma_buffers; + /* Store the SSI-specific information that we need */ + dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); + dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); + ret = snd_soc_register_platform(&of_dev->dev, &dma->dai); if (ret) { dev_err(&of_dev->dev, "could not register platform\n"); @@ -921,7 +941,7 @@ static void __exit fsl_soc_dma_exit(void) * will already have been probed. The easiest way to do that is to make the * __init function called via arch_initcall(). */ -arch_initcall(fsl_soc_dma_init); +module_init(fsl_soc_dma_init); module_exit(fsl_soc_dma_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index e152bc6..78fee97 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -126,7 +126,4 @@ struct fsl_dma_link_descriptor { u8 res[4]; /* Reserved */ } __attribute__ ((aligned(32), packed)); -struct snd_soc_platform_driver *fsl_soc_dma_to_dai(const char *path, - dma_addr_t ssi_stx_phys, dma_addr_t ssi_srx_phys); - #endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2388484..51b089f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -292,8 +292,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, /* The 'name' should not have any slashes in it. */ ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, - strrchr(ssi_private->name, '/') + 1, - ssi_private); + ssi_private->name, ssi_private); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not claim irq %u\n", ssi_private->irq); @@ -622,11 +621,9 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, int ret = 0; struct device_attribute *dev_attr; struct device_node *np = of_dev->dev.of_node; - const char *sprop; + const char *p, *sprop; struct resource res; char name[64]; - struct platform_device *pdev; - void *platform_data; /* We are only interested in SSIs with a codec phandle in them, so let's * make sure this SSI has one. @@ -641,21 +638,22 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, return -ENODEV; } - ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + - strlen(np->full_name), GFP_KERNEL); + /* The DAI name is the last part of the full name of the node. */ + p = strrchr(np->full_name, '/') + 1; + ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p), + GFP_KERNEL); if (!ssi_private) { dev_err(&of_dev->dev, "could not allocate DAI object\n"); return -ENOMEM; } + strcpy(ssi_private->name, p); + /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, sizeof(fsl_ssi_dai_template)); ssi_private->cpu_dai_drv.name = ssi_private->name; - /* The name we register with ASoC is the full node name */ - strlcpy(ssi_private->name, np->full_name, sizeof(ssi_private->name)); - /* Get the addresses and IRQ */ ret = of_address_to_resource(np, 0, &res); if (ret) { @@ -698,19 +696,26 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, } /* Trigger the machine driver's probe function. The platform driver - * name of the machine driver is taken from the /model node of the + * name of the machine driver is taken from the /model property of the * device tree. We also pass the address of the CPU DAI driver * structure. */ sprop = of_get_property(of_find_node_by_path("/"), "model", NULL); + /* Sometimes the model name has a "fsl," prefix, so we strip that. */ + p = strrchr(sprop, ','); + if (p) + sprop = p + 1; snprintf(name, sizeof(name), "snd-soc-%s", sprop); make_lowercase(name); - platform_data = &ssi_private->cpu_dai_drv; - pdev = platform_device_register_data(&of_dev->dev, name, - 0, &platform_data, sizeof(void *)); - - ssi_private->pdev = pdev; + ssi_private->pdev = + platform_device_register_data(&of_dev->dev, name, 0, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&of_dev->dev, "failed to register platform: %d\n", ret); + kfree(ssi_private); + return ret; + } return 0; } diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index e87cd1e..38339c1 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -22,6 +22,8 @@ /* There's only one global utilities register */ static phys_addr_t guts_phys; +#define DAI_NAME_SIZE 32 + /** * mpc8610_hpcd_data: machine-specific ASoC device data * @@ -38,7 +40,9 @@ struct mpc8610_hpcd_data { unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ - char dai_name[32]; + char codec_dai_name[DAI_NAME_SIZE]; + char codec_name[DAI_NAME_SIZE]; + char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ }; /** @@ -233,25 +237,68 @@ static int get_parent_cell_index(struct device_node *np) return *iprop; } +/** + * codec_node_dev_name - determine the dev_name for a codec node + * + * This function determines the dev_name for an I2C node. This is the name + * that would be returned by dev_name() if this device_node were part of a + * 'struct device' It's ugly and hackish, but it works. + * + * The dev_name for such devices include the bus number and I2C address. For + * example, "cs4270-codec.0-004f". + */ +static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) +{ + const u32 *iprop; + int bus, addr; + char temp[DAI_NAME_SIZE]; + + of_modalias_node(np, temp, DAI_NAME_SIZE); + + iprop = of_get_property(np, "reg", NULL); + if (!iprop) + return -EINVAL; + + addr = *iprop; + + bus = get_parent_cell_index(np); + if (bus < 0) + return bus; + + snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr); + + return 0; +} + static int get_dma_channel(struct device_node *ssi_np, const char *compatible, - dma_addr_t ssi_stx_phys, - dma_addr_t ssi_srx_phys, struct snd_soc_dai_link *dai, unsigned int *dma_channel_id, unsigned int *dma_id) { + struct resource res; struct device_node *dma_channel_np; const u32 *iprop; + int ret; dma_channel_np = get_node_by_phandle_name(ssi_np, compatible, "fsl,ssi-dma-channel"); if (!dma_channel_np) return -EINVAL; - // FIXME: change this to get the OF device name - dai->platform_name = fsl_soc_dma_to_dai(dma_channel_np->full_name, - ssi_stx_phys, ssi_srx_phys); + /* Determine the dev_name for the device_node. This code mimics the + * behavior of of_device_make_bus_id(). We need this because ASoC uses + * the dev_name() of the device to match the platform (DMA) device with + * the CPU (SSI) device. It's all ugly and hackish, but it works (for + * now). + * + * dai->platform name should already point to an allocated buffer. + */ + ret = of_address_to_resource(dma_channel_np, 0, &res); + if (ret) + return ret; + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", + (unsigned long long) res.start, dma_channel_np->name); iprop = of_get_property(dma_channel_np, "cell-index", NULL); if (!iprop) { @@ -265,6 +312,7 @@ static int get_dma_channel(struct device_node *ssi_np, return 0; } + /** * mpc8610_hpcd_probe: platform probe function for the machine driver * @@ -274,22 +322,16 @@ static int get_dma_channel(struct device_node *ssi_np, */ static int mpc8610_hpcd_probe(struct platform_device *pdev) { - struct snd_soc_dai_driver **platform_data = pdev->dev.platform_data; struct device *dev = pdev->dev.parent; + /* of_dev is the OF device for the SSI node that probed us */ struct of_device *of_dev = container_of(dev, struct of_device, dev); struct device_node *np = of_dev->dev.of_node; struct device_node *codec_np = NULL; - struct device_node *dma_np = NULL; - struct device_node *dma_channel_np = NULL; struct platform_device *sound_device = NULL; struct mpc8610_hpcd_data *machine_data; - int id; - dma_addr_t ssi_stx_phys; - dma_addr_t ssi_srx_phys; int ret = -ENODEV; const char *sprop; const u32 *iprop; - struct resource res; /* We are only interested in SSIs with a codec phandle in them, * so let's make sure this SSI has one. The MPC8610 HPCD only @@ -306,20 +348,27 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) if (!machine_data) return -ENOMEM; - machine_data->dai[0].name = machine_data->dai_name; + machine_data->dai[0].cpu_dai_name = dev_name(&of_dev->dev); + machine_data->dai[0].ops = &mpc8610_hpcd_ops; - /* FIXME: Timur, do we need to append .0 or .1 suffix here ? */ - machine_data->dai[0].cpu_dai_name = "fsl-ssi-dai"; - machine_data->dai[0].platform_name = "fsl-pcm-audio"; + /* Determine the codec name, it will be used as the codec DAI name */ + ret = codec_node_dev_name(codec_np, machine_data->codec_name, + DAI_NAME_SIZE); + if (ret) { + dev_err(&pdev->dev, "invalid codec node %s\n", + codec_np->full_name); + ret = -EINVAL; + goto error; + } + machine_data->dai[0].codec_name = machine_data->codec_name; + /* The DAI name from the codec (snd_soc_dai_driver.name) */ machine_data->dai[0].codec_dai_name = "cs4270-hifi"; - machine_data->dai[0].codec_name = "cs4270-codec"; - machine_data->dai[0].ops = &mpc8610_hpcd_ops; - - /* Store the codec name, it will be used as the codec DAI name */ - of_modalias_node(codec_np, machine_data->dai_name, - sizeof(machine_data->dai_name)); + /* We register two DAIs per SSI, one for playback and the other for + * capture. Currently, we only support codecs that have one DAI for + * both playback and capture. + */ memcpy(&machine_data->dai[1], &machine_data->dai[0], sizeof(struct snd_soc_dai_link)); @@ -331,7 +380,6 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) goto error; } machine_data->ssi_id = *iprop; - id = *iprop; /* Get the serial format and clock direction. */ sprop = of_get_property(np, "fsl,mode", NULL); @@ -399,18 +447,9 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) goto error; } - /* Read the SSI information from the device tree */ - ret = of_address_to_resource(np, 0, &res); - if (ret || !res.start) { - dev_err(&pdev->dev, "could not obtain SSI address\n"); - goto error; - } - ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); - ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); - /* Find the playback DMA channel to use. */ - ret = get_dma_channel(np, "fsl,playback-dma", ssi_stx_phys, - ssi_srx_phys, &machine_data->dai[0], + machine_data->dai[0].platform_name = machine_data->platform_name[0]; + ret = get_dma_channel(np, "fsl,playback-dma", &machine_data->dai[0], &machine_data->dma_channel_id[0], &machine_data->dma_id[0]); if (ret) { @@ -419,8 +458,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } /* Find the capture DMA channel to use. */ - ret = get_dma_channel(np, "fsl,capture-dma", ssi_stx_phys, - ssi_srx_phys, &machine_data->dai[1], + machine_data->dai[1].platform_name = machine_data->platform_name[1]; + ret = get_dma_channel(np, "fsl,capture-dma", &machine_data->dai[1], &machine_data->dma_channel_id[1], &machine_data->dma_id[1]); if (ret) { @@ -429,18 +468,14 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } /* Initialize our DAI data structure. */ - iprop = of_get_property(codec_np, "reg", NULL); - if (!iprop) { - dev_err(&pdev->dev, "codec node is missing 'reg' property\n"); - goto error; - } - machine_data->dai[0].stream_name = "playback"; machine_data->dai[1].stream_name = "capture"; + machine_data->dai[0].name = machine_data->dai[0].stream_name; + machine_data->dai[1].name = machine_data->dai[1].stream_name; machine_data->card.probe = mpc8610_hpcd_machine_probe; machine_data->card.remove = mpc8610_hpcd_machine_remove; - machine_data->card.name = "MPC8610 HPCD"; + machine_data->card.name = pdev->name; /* The platform driver name */ machine_data->card.num_links = 2; machine_data->card.dai_link = machine_data->dai; @@ -468,8 +503,6 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) error: of_node_put(codec_np); - of_node_put(dma_np); - of_node_put(dma_channel_np); if (sound_device) platform_device_unregister(sound_device); -- 1.7.0.1

2 1

[alsa-devel] Regression 2.6.35-rc6: ALSA Intel HDA/Realtek: missing Beep
by Mario 'BitKoenig' Holbe 29 Jul '10

29 Jul '10

Hello, My PC-Speaker Beep control worked in 2.6.34, but is gone in 2.6.35-rc6. The sound device is an Asus P5E-V HDMI (Intel G35) onboard Intel HDA w/ Realtek ALC883 codec. The driver does still register a PCBeep input, but the Controls as well as the sound (:)) are gone. HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: irq 47 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input5 $ diff card0-codec#0-2.6.34 card0-codec#0-2.6.35-rc6 111,114d110 < Control: name="Beep Playback Volume", index=0, device=0 < ControlAmp: chs=3, dir=In, idx=5, ofs=0 < Control: name="Beep Playback Switch", index=0, device=0 < ControlAmp: chs=3, dir=In, idx=5, ofs=0 Attached /proc/asound/card0/codec#0 from 2.6.35-rc6. Thanks for your work & regards Mario -- Good, Fast, Cheap: Pick any two (you can't have all three). -- RFC 1925, 7a

2 4

[alsa-devel] asoc multi-component: almost there!
by Timur Tabi 28 Jul '10

28 Jul '10

Liam, I almost got it working. I can generate names that match, so it finds it: soc_bind_dai_link:1136 cpu_dai->name=e0016000.ssi, dai_link->cpu_dai_name=e0016000.ssi soc_bind_dai_link:1159 codec->name=cs4270-codec.0-004f, dai_link->codec_name=cs4270-codec.0-004f soc_bind_dai_link:1169 codec_dai->name=e0016000.ssi, dai_link->codec_dai_name=cs4270-hifi soc_bind_dai_link:1169 codec_dai->name=cs4270-hifi, dai_link->codec_dai_name=cs4270-hifi soc_bind_dai_link:1194 platform->name=e0021180.dma-channel, dai_link->platform_name=e0021100.dma-channel soc_bind_dai_link:1194 platform->name=e0021100.dma-channel, dai_link->platform_name=e0021100.dma-channel soc_bind_dai_link:1136 cpu_dai->name=e0016000.ssi, dai_link->cpu_dai_name=e0016000.ssi soc_bind_dai_link:1159 codec->name=cs4270-codec.0-004f, dai_link->codec_name=cs4270-codec.0-004f soc_bind_dai_link:1169 codec_dai->name=e0016000.ssi, dai_link->codec_dai_name=cs4270-hifi soc_bind_dai_link:1169 codec_dai->name=cs4270-hifi, dai_link->codec_dai_name=cs4270-hifi soc_bind_dai_link:1194 platform->name=e0021180.dma-channel, dai_link->platform_name=e0021180.dma-channel asoc: cs4270-hifi <-> e0016000.ssi mapping ok but then I get this: sysfs: cannot create duplicate filename '/devices/platform/soc-audio/cs4270' I'm registering two DAIs -- one for playback, one for capture, but I'm using the same name for the codec DAI. How do I resolve this? -- Timur Tabi Linux kernel developer at Freescale

4 10

Re: [alsa-devel] [PATCH 2/2] ARM: S5PV210: Add the GPIO lib
by Chanwoo Choi 28 Jul '10

28 Jul '10

Marek Szyprowski wrote: > Hello, > > On Wednesday, July 28, 2010 5:05 AM Chanwoo Choi wrote: > >> This patch add the definition of GPIO to support WM8994 audio codec on >> I2C bus. > > I've already submitted similar patch that does this job correctly. See: > http://git.kernel.org/?p=linux/kernel/git/kgene/linux-samsung.git;a=commit;… > d303c159aca6db11fb9a03068f52cbf7c145d > Ah~, I haven't checked your patch. I will only send the patch which support audio on Goni/Aquila board.

1 0

[alsa-devel] [PATCH] ARM: S5PV210: Add audio support to Goni and Aquila board
by Chanwoo Choi 28 Jul '10

28 Jul '10

This patch the I2C board information for the WM8994 used in the Goni/Aquila as audio codec and adds the I2C/I2S platform drivers. Additionlly, to control power consumption have registerd the voltage consumer of WM8994 to the regulator framework. I initialize gpio settting relevant to operation of audio codec. I explain following comment concering gpio setting: - CODEC_XTAL_EN : This gpio enables that the main clock is provided to operate audio codec. (Only used on Aquila board) - ADC_EN : This gpio enable the ADC device which is used to detect the kind of jack. (SND_JACK_HEADPHONE/HEADSET/MECHANICAL/AVOUT) According to the kind of jack, an electric current is changed. (Only used on Aquila board) - MICBIAS_EN : This gpio enable the microphone to support capture. Signed-off-by : Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by : Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by : Kyungmin Park <kyungmin.park(a)samsung.com> --- arch/arm/mach-s5pv210/mach-aquila.c | 177 +++++++++++++++++++++++++++++++++++ arch/arm/mach-s5pv210/mach-goni.c | 165 ++++++++++++++++++++++++++++++++ 2 files changed, 342 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s5pv210/mach-aquila.c b/arch/arm/mach-s5pv210/mach-aquila.c index f0f960f..312ff76 100644 --- a/arch/arm/mach-s5pv210/mach-aquila.c +++ b/arch/arm/mach-s5pv210/mach-aquila.c @@ -17,9 +17,11 @@ #include <linux/i2c-gpio.h> #include <linux/i2c/mcs.h> #include <linux/mfd/max8998.h> +#include <linux/mfd/wm8994/pdata.h> #include <linux/gpio_keys.h> #include <linux/input.h> #include <linux/gpio.h> +#include <linux/regulator/fixed.h> #include <asm/mach/arch.h> #include <asm/mach/map.h> @@ -397,6 +399,120 @@ static struct max8998_platform_data aquila_max8998_pdata = { }; #endif +static struct regulator_consumer_supply wm8994_fixed_voltage0_supplies[] = { + { + .dev_name = "5-001a", + .supply = "DBVDD", + }, { + .dev_name = "5-001a", + .supply = "AVDD2", + }, { + .dev_name = "5-001a", + .supply = "CPVDD", + }, + +}; + +static struct regulator_consumer_supply wm8994_fixed_voltage1_supplies[] = { + { + .dev_name = "5-001a", + .supply = "SPKVDD1", + }, { + .dev_name = "5-001a", + .supply = "SPKVDD2", + }, +}; + +static struct regulator_init_data wm8994_fixed_voltage0_init_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(wm8994_fixed_voltage0_supplies), + .consumer_supplies = wm8994_fixed_voltage0_supplies, +}; + +static struct regulator_init_data wm8994_fixed_voltage1_init_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(wm8994_fixed_voltage1_supplies), + .consumer_supplies = wm8994_fixed_voltage1_supplies, +}; + +static struct fixed_voltage_config wm8994_fixed_voltage0_config = { + .supply_name = "VCC_1.8V_PDA", + .microvolts = 1800000, + .gpio = -EINVAL, + .init_data = &wm8994_fixed_voltage0_init_data, +}; + +static struct fixed_voltage_config wm8994_fixed_voltage1_config = { + .supply_name = "V_BAT", + .microvolts = 3700000, + .gpio = -EINVAL, + .init_data = &wm8994_fixed_voltage1_init_data, +}; + +static struct platform_device wm8994_fixed_voltage0 = { + .name = "reg-fixed-voltage", + .id = 0, + .dev = { + .platform_data = &wm8994_fixed_voltage0_config, + }, +}; + +static struct platform_device wm8994_fixed_voltage1 = { + .name = "reg-fixed-voltage", + .id = 1, + .dev = { + .platform_data = &wm8994_fixed_voltage1_config, + }, +}; + +static struct regulator_consumer_supply wm8994_avdd1_supply = { + .dev_name = "5-001a", + .supply = "AVDD1", +}; + +static struct regulator_consumer_supply wm8994_dcvdd_supply = { + .dev_name = "5-001a", + .supply = "DCVDD", +}; + +static struct regulator_init_data wm8994_ldo1_data = { + .constraints = { + .name = "AVDD1_3.0V", + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, + .num_consumer_supplies = 1, + .consumer_supplies = &wm8994_avdd1_supply, +}; + +static struct regulator_init_data wm8994_ldo2_data = { + .constraints = { + .name = "DCVDD_1.0V", + }, + .num_consumer_supplies = 1, + .consumer_supplies = &wm8994_dcvdd_supply, +}; + +static struct wm8994_pdata wm8994_platform_data = { + /* configure gpio1 function: 0x0001(Logic level input/output) */ + .gpio_defaults[0] = 0x0001, + /* configure gpio3/4/5/7 function for AIF2 voice */ + .gpio_defaults[2] = 0x8100, + .gpio_defaults[3] = 0x8100, + .gpio_defaults[4] = 0x8100, + .gpio_defaults[6] = 0x0100, + /* configure gpio8/9/10/11 function for AIF3 BT */ + .gpio_defaults[7] = 0x8100, + .gpio_defaults[8] = 0x0100, + .gpio_defaults[9] = 0x0100, + .gpio_defaults[10] = 0x0100, + .ldo[0] = { S5PV210_MP03(6), NULL, &wm8994_ldo1_data }, /* XM0FRNB_2 */ + .ldo[1] = { 0, NULL, &wm8994_ldo2_data }, +}; + /* GPIO I2C PMIC */ #define AP_I2C_GPIO_PMIC_BUS_4 4 static struct i2c_gpio_platform_data aquila_i2c_gpio_pmic_data = { @@ -422,6 +538,58 @@ static struct i2c_board_info i2c_gpio_pmic_devs[] __initdata = { #endif }; +/* GPIO I2C AP 1.8V */ +#define AP_I2C_GPIO_BUS_5 5 +static struct i2c_gpio_platform_data i2c_gpio5_data = { + .sda_pin = S5PV210_MP05(3), /* XM0ADDR_11 */ + .scl_pin = S5PV210_MP05(2), /* XM0ADDR_10 */ +}; + +static struct platform_device i2c_gpio5 = { + .name = "i2c-gpio", + .id = AP_I2C_GPIO_BUS_5, + .dev = { + .platform_data = &i2c_gpio5_data, + }, +}; + +static struct i2c_board_info i2c_gpio5_devs[] __initdata = { + { + /* CS/ADDR = low 0x34 (FYI: high = 0x36) */ + I2C_BOARD_INFO("wm8994", 0x34 >> 1), + .platform_data = &wm8994_platform_data, + }, +}; + +static void __init aquila_sound_init(void) +{ + unsigned int gpio; + + /* CODEC_XTAL_EN */ + gpio = S5PV210_GPH3(2); /* XEINT_26 */ + gpio_request(gpio, "CODEC_XTAL_EN"); + s3c_gpio_cfgpin(gpio, S3C_GPIO_OUTPUT); + s3c_gpio_setpull(gpio, S3C_GPIO_PULL_NONE); + gpio_direction_output(gpio, 1); + + /* CLKOUT[9:8] set to 0x3(XUSBXTI) of 0xE010E000(OTHERS) + * for 24MHZ + */ + writel(readl(S5P_OTHERS) | (0x3 << 8), S5P_OTHERS); + + /* MICBIAS_EN */ + gpio = S5PV210_GPJ4(2); /* XMSMRN */ + gpio_request(gpio, "MICBIAS_EN"); + s3c_gpio_cfgpin(gpio, S3C_GPIO_OUTPUT); + gpio_direction_output(gpio, 1); + + /* ADC_EN */ + gpio = S5PV210_GPJ3(2); + gpio_request(gpio, "ADC_EN"); + s3c_gpio_cfgpin(gpio, S3C_GPIO_OUTPUT); + gpio_direction_output(gpio, 1); +} + /* PMIC Power button */ static struct gpio_keys_button aquila_gpio_keys_table[] = { { @@ -523,6 +691,10 @@ static struct platform_device *aquila_devices[] __initdata = { &s5pc110_device_onenand, &samsung_device_keypad, &i2c_gpio10, + &i2c_gpio5, + &wm8994_fixed_voltage0, + &wm8994_fixed_voltage1, + &s5pv210_device_iis0, }; static void __init aquila_map_io(void) @@ -539,6 +711,11 @@ static void __init aquila_machine_init(void) i2c_register_board_info(AP_I2C_GPIO_PMIC_BUS_4, i2c_gpio_pmic_devs, ARRAY_SIZE(i2c_gpio_pmic_devs)); + /* SOUND */ + aquila_sound_init(); + i2c_register_board_info(AP_I2C_GPIO_BUS_5, i2c_gpio5_devs, + ARRAY_SIZE(i2c_gpio5_devs)); + /* FB */ s3c_fb_set_platdata(&aquila_lcd_pdata); diff --git a/arch/arm/mach-s5pv210/mach-goni.c b/arch/arm/mach-s5pv210/mach-goni.c index 5fec4a0..944e600 100644 --- a/arch/arm/mach-s5pv210/mach-goni.c +++ b/arch/arm/mach-s5pv210/mach-goni.c @@ -17,9 +17,11 @@ #include <linux/i2c-gpio.h> #include <linux/i2c/mcs.h> #include <linux/mfd/max8998.h> +#include <linux/mfd/wm8994/pdata.h> #include <linux/gpio_keys.h> #include <linux/input.h> #include <linux/gpio.h> +#include <linux/regulator/fixed.h> #include <asm/mach/arch.h> #include <asm/mach/map.h> @@ -373,6 +375,120 @@ static struct max8998_regulator_data goni_regulators[] = { { MAX8998_BUCK4, &goni_buck4_data }, }; +static struct regulator_consumer_supply wm8994_fixed_voltage0_supplies[] = { + { + .dev_name = "5-001a", + .supply = "DBVDD", + }, { + .dev_name = "5-001a", + .supply = "AVDD2", + }, { + .dev_name = "5-001a", + .supply = "CPVDD", + }, + +}; + +static struct regulator_consumer_supply wm8994_fixed_voltage1_supplies[] = { + { + .dev_name = "5-001a", + .supply = "SPKVDD1", + }, { + .dev_name = "5-001a", + .supply = "SPKVDD2", + }, +}; + +static struct regulator_init_data wm8994_fixed_voltage0_init_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(wm8994_fixed_voltage0_supplies), + .consumer_supplies = wm8994_fixed_voltage0_supplies, +}; + +static struct regulator_init_data wm8994_fixed_voltage1_init_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(wm8994_fixed_voltage1_supplies), + .consumer_supplies = wm8994_fixed_voltage1_supplies, +}; + +static struct fixed_voltage_config wm8994_fixed_voltage0_config = { + .supply_name = "VCC_1.8V_PDA", + .microvolts = 1800000, + .gpio = -EINVAL, + .init_data = &wm8994_fixed_voltage0_init_data, +}; + +static struct fixed_voltage_config wm8994_fixed_voltage1_config = { + .supply_name = "V_BAT", + .microvolts = 3700000, + .gpio = -EINVAL, + .init_data = &wm8994_fixed_voltage1_init_data, +}; + +static struct platform_device wm8994_fixed_voltage0 = { + .name = "reg-fixed-voltage", + .id = 0, + .dev = { + .platform_data = &wm8994_fixed_voltage0_config, + }, +}; + +static struct platform_device wm8994_fixed_voltage1 = { + .name = "reg-fixed-voltage", + .id = 1, + .dev = { + .platform_data = &wm8994_fixed_voltage1_config, + }, +}; + +static struct regulator_consumer_supply wm8994_avdd1_supply = { + .dev_name = "5-001a", + .supply = "AVDD1", +}; + +static struct regulator_consumer_supply wm8994_dcvdd_supply = { + .dev_name = "5-001a", + .supply = "DCVDD", +}; + +static struct regulator_init_data wm8994_ldo1_data = { + .constraints = { + .name = "AVDD1_3.0V", + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, + .num_consumer_supplies = 1, + .consumer_supplies = &wm8994_avdd1_supply, +}; + +static struct regulator_init_data wm8994_ldo2_data = { + .constraints = { + .name = "DCVDD_1.0V", + }, + .num_consumer_supplies = 1, + .consumer_supplies = &wm8994_dcvdd_supply, +}; + +static struct wm8994_pdata wm8994_platform_data = { + /* configure gpio1 function: 0x0001(Logic level input/output) */ + .gpio_defaults[0] = 0x0001, + /* configure gpio3/4/5/7 function for AIF2 voice */ + .gpio_defaults[2] = 0x8100, + .gpio_defaults[3] = 0x8100, + .gpio_defaults[4] = 0x8100, + .gpio_defaults[6] = 0x0100, + /* configure gpio8/9/10/11 function for AIF3 BT */ + .gpio_defaults[7] = 0x8100, + .gpio_defaults[8] = 0x0100, + .gpio_defaults[9] = 0x0100, + .gpio_defaults[10] = 0x0100, + .ldo[0] = { S5PV210_GPF3(4), NULL, &wm8994_ldo1_data }, /* XVVSYNC_LDI */ + .ldo[1] = { 0, NULL, &wm8994_ldo2_data }, +}; + static struct max8998_platform_data goni_max8998_pdata = { .num_regulators = ARRAY_SIZE(goni_regulators), .regulators = goni_regulators, @@ -404,6 +520,45 @@ static struct i2c_board_info i2c_gpio_pmic_devs[] __initdata = { #endif }; +/* GPIO I2C AP 1.8V */ +#define AP_I2C_GPIO_BUS_5 5 +static struct i2c_gpio_platform_data i2c_gpio5_data = { + .sda_pin = S5PV210_MP05(3), /* XM0ADDR_11 */ + .scl_pin = S5PV210_MP05(2), /* XM0ADDR_10 */ +}; + +static struct platform_device i2c_gpio5 = { + .name = "i2c-gpio", + .id = AP_I2C_GPIO_BUS_5, + .dev = { + .platform_data = &i2c_gpio5_data, + }, +}; + +static struct i2c_board_info i2c_gpio5_devs[] __initdata = { + { + /* CS/ADDR = low 0x34 (FYI: high = 0x36) */ + I2C_BOARD_INFO("wm8994", 0x34 >> 1), + .platform_data = &wm8994_platform_data, + }, +}; + +static void __init goni_sound_init(void) +{ + unsigned int gpio; + + /* CLKOUT[9:8] set to 0x3(XUSBXTI) of 0xE010E000(OTHERS) + * for 24MHZ + */ + writel(readl(S5P_OTHERS) | (0x3 << 8), S5P_OTHERS); + + /* MICBIAS_EN */ + gpio = S5PV210_GPJ4(2); /* XMSMRN */ + gpio_request(gpio, "MICBIAS_EN"); + s3c_gpio_cfgpin(gpio, S3C_GPIO_OUTPUT); + gpio_direction_output(gpio, 1); +} + /* PMIC Power button */ static struct gpio_keys_button goni_gpio_keys_table[] = { { @@ -504,7 +659,11 @@ static struct platform_device *goni_devices[] __initdata = { &samsung_device_keypad, &goni_i2c_gpio_pmic, &goni_device_gpiokeys, + &i2c_gpio5, &i2c_gpio10, + &wm8994_fixed_voltage0, + &wm8994_fixed_voltage1, + &s5pv210_device_iis0, }; static void __init goni_map_io(void) @@ -520,6 +679,12 @@ static void __init goni_machine_init(void) goni_pmic_init(); i2c_register_board_info(AP_I2C_GPIO_PMIC_BUS_4, i2c_gpio_pmic_devs, ARRAY_SIZE(i2c_gpio_pmic_devs)); + + /* SOUND */ + goni_sound_init(); + i2c_register_board_info(AP_I2C_GPIO_BUS_5, i2c_gpio5_devs, + ARRAY_SIZE(i2c_gpio5_devs)); + /* FB */ s3c_fb_set_platdata(&goni_lcd_pdata); -- 1.6.3.3

1 0

[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: FIFO mode updates
by Peter Ujfalusi 28 Jul '10

28 Jul '10

Hello, The main feature of this series is the automatic FIFO configuration of the FIFO, when one of the FIFO mode is selected. The first patch simplifies the handling of the Mode1 in manual mode, and also removes the hardwired latency definition from the code, so it can be adjusted based on the underlying platform. The second patch adds option to select the automatic FIFO configuration. When this mode is selected the driver will configure the FIFO to match the stream's period size with the burst length (in terms of samples bursted out from the host). In Mode1 this is trivial, since I can configure the nSample to match with the period size. However the Mode7 is a bit more complicated. The formula (UTHR_FROM_PERIOD_SIZE) calculates the UTHR configuration for the given number of samples. The patches are generated against: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git:topic/asoc --- Peter Ujfalusi (2): ASoC: tlv320dac33: Revisit the FIFO Mode1 handling ASoC: tlv320dac33: Add support for automatic FIFO configuration include/sound/tlv320dac33-plat.h | 2 + sound/soc/codecs/tlv320dac33.c | 125 ++++++++++++++++++++++++-------------- 2 files changed, 82 insertions(+), 45 deletions(-) -- 1.7.2

2 5

[alsa-devel] using plug for sample format conversion
by Adam Rosenberg 28 Jul '10

28 Jul '10

I am trying to use plug to play audio files that are S16_LE but my hardware only supports S32_LE. It is not working. Here is an example of the problem using speaker-test. First I use S32_LE and it works as expected. I then try S16_LE but it fails. Does this mean the plug plugin cannot convert between these sample formats? Thanks, Adam root:/> speaker-test -l 1 -r 48000 -c 2 -F S32_LE -Dplughw:0,0,0 speaker-test 1.0.12 Playback device is plughw:0,0,0 Stream parameters are 48000Hz, S32_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 1 to 4096 Period size range from 1 to 4096 Using max buffer size 4096 Periods = 4 was set period_size = 1024 was set buffer_size = 4096 0 - Front Left 1 - Front Right Time per period = 5.844000 root:/> speaker-test -l 1 -r 48000 -c 2 -F S16_LE -Dplughw:0,0,0 speaker-test 1.0.12 Playback device is plughw:0,0,0 Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 1 to 4096 Period size range from 1 to 4096 Using max buffer size 4096 Periods = 4 was set period_size = 1024 was set buffer_size = 4096 Unable to set hw params for playback: Invalid argument Setting of hwparams failed: Invalid argument

2 1

[alsa-devel] hda-verb, hda-analyzer, hda-emu and codecgraph
by David Henningsson 28 Jul '10

28 Jul '10

Hi Takashi, Do you plan to make hda-verb, hda-analyzer, hda-emu and/or codecgraph a part of alsa-tools/alsa-utils/pyalsa in the future? I'm asking because I would like to know if it makes sense to package them separately for Debian. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 10

[alsa-devel] detecting the presence of dmix
by dv 27 Jul '10

27 Jul '10

Hello, is it possible to programmatically detect whether or not dmix is enabled? it is important for a software stack where dmix would introduce unwanted resampling artifacts

2 1

[alsa-devel] simple program for capturing and playing pcm
by loody 27 Jul '10

27 Jul '10

Dear all: Is there simple program which can help me to play and capture pcm files from usb speaker and mic? appreciate your help, miloody

3 5

[alsa-devel] [PATCH] ASoC: TWL4030: Capture route DAPM event fix
by Peter Ujfalusi 27 Jul '10

27 Jul '10

There is no need to handle POST_PMU, POST_PMD event with the Capture Route widget. It is enough to handle POST_REG event, since that will come when the user changes the routing, and we will switch the needed bits in the registers. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index bd557c2..d401c59 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1432,11 +1432,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { TX2 Left/Right: either analog Left/Right or Digimic1 */ SND_SOC_DAPM_MUX_E("TX1 Capture Route", SND_SOC_NOPM, 0, 0, &twl4030_dapm_micpathtx1_control, micpath_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| SND_SOC_DAPM_POST_REG), SND_SOC_DAPM_MUX_E("TX2 Capture Route", SND_SOC_NOPM, 0, 0, &twl4030_dapm_micpathtx2_control, micpath_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| SND_SOC_DAPM_POST_REG), /* Analog input mixers for the capture amplifiers */ -- 1.7.1.1

3 2

[alsa-devel] [PATCH] ALSA: hda: Fixup commit 79c944ad (caused codec initialization failure)
by David Henningsson 27 Jul '10

27 Jul '10

A recent commit in current HEAD (79c944ad) caused codecs not to be detected in some cases, found on several Realtek codecs. Please apply the attached fix for that problem. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 3

Re: [alsa-devel] Time to give up on PA
by Raymond Yau 27 Jul '10

27 Jul '10

2010/7/27 Chris <cpollock(a)embarqmail.com> > Raymond, please look at the attached file that was gleamed from a 552k > hourly syslog snippet. From what I can see PA is not working correctly > and it's sucking up all my memory not to mention causing my HD to thrash > for minutes at a time. For instance, the drive was being accessed from > 10pm to around 10:20 continously, I was finally able to log out and shut > down PA through MCC. Also look at the one entry where clamav would not > run. > > http://ez-files.net/646514 > > > Jul 26 20:44:40 localhost spamd[17669]: rules: failed to run CLAMAV > test, skipping: > Jul 26 20:44:40 localhost spamd[17669]: (__alarm__ignore__(172) > Jul 26 20:44:40 localhost spamd[17669]: ) > Jul 26 20:44:42 localhost spamd[17669]: check: exceeded time limit in > Mail::SpamAssassin::Plugin::Check::_eval_tests_type13_prineg100_set3, > skipping further tests > Jul 26 20:44:44 localhost spamd[17669]: plugin: eval failed: > __alarm__ignore__(131) > > I've currently got PA disabled in MCC I'm afraid to restart because of > all the problems, any advice?? I don't see a fix in sight. > > Thanks for all your help > > Chris > It is quite clear that the sound card ens1371 is underrun as appl_ptr is behind hw_ptr It is strange that PA server use 1 period per buffer (16384) which is different from your daemon.conf and also different from your two bug report at mandriva bugzilla http://thread.gmane.org/gmane.linux.alsa.devel/75101/focus=75105 Do this mean that some parameters in daemon.conf is obsolete ? https://qa.mandriva.com/show_bug.cgi?id=56473#c15 https://qa.mandriva.com/show_bug.cgi?id=60169#c0 Although 16384 is multiple of 8 but one period per buffer is unlikely to work on those non HDA sound cards http://thread.gmane.org/gmane.linux.alsa.devel/66376/focus=66484 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.590| 0.062) alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -424360 bytes (-2405 ms). Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.590| 0.062) alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_ens1371'. Please report this issue to the ALSA developers. Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: snd_pcm_dump(): Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: Hardware PCM card 0 'Ensoniq AudioPCI' device 0 subdevice 0 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: Its setup is: Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: stream : PLAYBACK Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: access : MMAP_INTERLEAVED Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: format : S16_LE Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: subformat : STD Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: channels : 2 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: rate : 44100 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: exact rate : 44101 (1445100000/32768) Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: msbits : 16 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: buffer_size : 16384 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: period_size : 16384 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: period_time : 371510 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: tstamp_mode : ENABLE Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: period_step : 1 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: avail_min : 16384 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: period_event : 0 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: start_threshold : -1 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: stop_threshold : 1073741824 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: silence_threshold: 0 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: silence_size : 0 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: boundary : 1073741824 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: appl_ptr : 389015462 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.634| 0.043) alsa-util.c: hw_ptr : 389121552 Jul 26 20:41:13 localhost pulseaudio[26721]: (89776.669| 0.035) alsa-sink.c: Increasing wakeup watermark to 120.11 ms

1 0

Re: [alsa-devel] [Fwd: Re: [pulseaudio-discuss] pulseaudio debug mode]
by Raymond Yau 27 Jul '10

27 Jul '10

2010/7/26 Chris <cpollock(a)embarqmail.com> > Raymond, attached is a post I made today to the pulseaudio list. > > -- > Chris > KeyID 0xE372A7DA98E6705C > > http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/7204/focus=7218 Refer to the log http://pastebin.com/tZNYaqRV What is the meaning of "memblock.c: Pool full" ? How large is this pool ? Is this memblock used for storing audio data ? 1. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.021| 5.573) ratelimit.c: 522 events suppressed 2. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.014) asyncq.c: q overrun, queuing locally 3. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 4. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 5. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 6. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 7. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 8. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 9. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 10. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 11. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 12. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) asyncq.c: q overrun, queuing locally 13. Jul 26 09:05:31 localhost pulseaudio[26721]: (48035.035| 0.000) alsa-sink.c: Increasing wakeup watermark to 140.00 ms 14. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.068| 5.033) ratelimit.c: 1032 events suppressed 15. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.014) asyncq.c: q overrun, queuing locally 16. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) asyncq.c: q overrun, queuing locally 17. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) asyncq.c: q overrun, queuing locally 18. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) asyncq.c: q overrun, queuing locally 19. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) asyncq.c: q overrun, queuing locally 20. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) asyncq.c: q overrun, queuing locally 21. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) asyncq.c: q overrun, queuing locally 22. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) asyncq.c: q overrun, queuing locally 23. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) asyncq.c: q overrun, queuing locally 24. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) memblock.c: Pool full 25. Jul 26 09:05:36 localhost pulseaudio[26721]: (48040.083| 0.000) memblock.c: Pool full 26. Jul 26 09:05:41 localhost pulseaudio[26721]: (48045.075| 4.992) ratelimit.c: 1154 events suppressed 27. Jul 26 09:05:41 localhost pulseaudio[26721]: (48045.107| 0.031) asyncq.c: q overrun, queuing locally 28. Jul 26 09:05:41 localhost pulseaudio[26721]: (48045.107| 0.000) memblock.c: Pool full 29. Jul 26 09:05:41 localhost pulseaudio[26721]: (48045.107| 0.000) asyncq.c: q overrun, queuing locally 30. Jul 26 09:05:41 localhost pulseaudio[26721]: (48045.107| 0.000) asyncq.c: q overrun, queuing locally 31. Jul 26 09:05:41 localhost pulseaudio[26721]: (48045.107| 0.000) asyncq.c: q overrun, queuing locally 32. Jul 26 09:05:41 localhost pulseaudio[26721]: (48045.107| 0.000) asyncq.c: q overrun, queuing locally 33. Jul 26 09:05:41 localhost pulseaudio[26721]: (48045.107| 0.000) memblock.c: Pool full 34. Jul 26 09:05:41 localhost pulseaudio[26721]: (48045.117| 0.009) memblock.c: Pool full 35. Jul 26 09:05:42 localhost pulseaudio[26721]: (48045.127| 0.010) memblock.c: Pool full 36. Jul 26 09:05:42 localhost pulseaudio[26721]: (48045.148| 0.020) asyncq.c: q overrun, queuing locally 37. Jul 26 09:05:42 localhost pulseaudio[26721]: (48045.148| 0.000) asyncq.c: q overrun, queuing locally Look like PA server decreasing wakeup watermark until underrun occur after Freeing input 14 "bell-window-system" this decreasing/increasing wakeup watermark cycle is repeated a few time Are there any PA clients still connected to PA server ? 1. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) alsa-sink.c: Latency set to 150.00ms 2. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) alsa-sink.c: hwbuf_unused=39076 3. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) alsa-sink.c: setting avail_min=11091 4. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) alsa-sink.c: Requested to rewind 65536 bytes. 5. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) alsa-sink.c: Limited to 5252 bytes. 6. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) alsa-sink.c: before: 1313 7. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) alsa-sink.c: after: 1313 8. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) alsa-sink.c: Rewound 5252 bytes. 9. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) sink.c: Processing rewind... 10. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.022| 0.000) source.c: Processing rewind... 11. Jul 26 09:07:13 localhost pulseaudio[26721]: (48137.024| 0.001) sink-input.c: Freeing input 14 "bell-window-system" 1. Jul 26 09:07:33 localhost pulseaudio[26721]: (48157.094| 20.069) alsa-sink.c: Decreasing wakeup watermark to 115.06 ms 2. Jul 26 09:07:53 localhost pulseaudio[26721]: (48177.108| 20.013) alsa-sink.c: Decreasing wakeup watermark to 110.07 ms 3. Jul 26 09:08:13 localhost pulseaudio[26721]: (48197.143| 20.035) alsa-sink.c: Decreasing wakeup watermark to 105.08 ms 4. Jul 26 09:08:34 localhost pulseaudio[26721]: (48217.152| 20.008) alsa-sink.c: Decreasing wakeup watermark to 100.09 ms 5. Jul 26 09:30:19 localhost pulseaudio[26721]: (49522.760|1305.608) alsa-sink.c: Underrun! 6. Jul 26 09:30:19 localhost pulseaudio[26721]: (49522.760| 0.000) alsa-sink.c: Increasing wakeup watermark to 110.09 ms 7. Jul 26 09:30:39 localhost pulseaudio[26721]: (49542.813| 20.052) alsa-sink.c: Decreasing wakeup watermark to 105.10 ms 8. Jul 26 09:31:18 localhost pulseaudio[26721]: (49581.831| 39.018) alsa-sink.c: Decreasing wakeup watermark to 100.11 ms 9. 10. Jul 26 10:11:01 localhost pulseaudio[26721]: (51964.860|2383.029) alsa-sink.c: Underrun! 11. Jul 26 10:11:01 localhost pulseaudio[26721]: (51964.888| 0.028) alsa-sink.c: Increasing wakeup watermark to 110.11 ms 1. Jul 26 10:11:10 localhost pulseaudio[26721]: (51974.048| 9.160) asyncq.c: q overrun, queuing locally 2. Jul 26 10:11:10 localhost pulseaudio[26721]: (51974.049| 0.000) asyncq.c: q overrun, queuing locally 3. Jul 26 10:11:10 localhost pulseaudio[26721]: (51974.049| 0.000) asyncq.c: q overrun, queuing locally 4. Jul 26 10:11:10 localhost pulseaudio[26721]: (51974.089| 0.040) asyncq.c: q overrun, queuing locally 5. Jul 26 10:11:10 localhost pulseaudio[26721]: (51974.089| 0.000) asyncq.c: q overrun, queuing locally 6. Jul 26 10:11:10 localhost pulseaudio[26721]: (51974.129| 0.039) asyncq.c: q overrun, queuing locally 7. Jul 26 10:11:10 localhost pulseaudio[26721]: (51974.129| 0.000) asyncq.c: q overrun, queuing locally 8. Jul 26 10:11:11 localhost pulseaudio[26721]: (51974.168| 0.039) asyncq.c: q overrun, queuing locally 9. Jul 26 10:11:11 localhost pulseaudio[26721]: (51974.169| 0.000) asyncq.c: q overrun, queuing locally 10. Jul 26 10:11:11 localhost pulseaudio[26721]: (51974.208| 0.039) asyncq.c: q overrun, queuing locally 11. Jul 26 10:11:11 localhost pulseaudio[26721]: (51974.208| 0.000) asyncq.c: q overrun, queuing locally 12. Jul 26 10:11:15 localhost pulseaudio[26721]: (51978.185| 3.976) alsa-sink.c: Increasing wakeup watermark to 120.11 ms 13. Jul 26 10:11:15 localhost pulseaudio[26721]: (51979.064| 0.878) ratelimit.c: 258 events suppressed 14. Jul 26 10:11:15 localhost pulseaudio[26721]: (51979.064| 0.000) asyncq.c: q overrun, queuing locally 15. Jul 26 10:11:15 localhost pulseaudio[26721]: (51979.064| 0.000) asyncq.c: q overrun, queuing locally 16. Jul 26 10:11:15 localhost pulseaudio[26721]: (51979.094| 0.029) asyncq.c: q overrun, queuing locally 17. Jul 26 10:11:15 localhost pulseaudio[26721]: (51979.094| 0.000) asyncq.c: q overrun, queuing locally 18. Jul 26 10:11:15 localhost pulseaudio[26721]: (51979.124| 0.029) asyncq.c: q overrun, queuing locally 19. Jul 26 10:11:15 localhost pulseaudio[26721]: (51979.124| 0.000) asyncq.c: q overrun, queuing locally 20. Jul 26 10:11:16 localhost pulseaudio[26721]: (51979.154| 0.029) asyncq.c: q overrun, queuing locally 21. Jul 26 10:11:16 localhost pulseaudio[26721]: (51979.154| 0.000) asyncq.c: q overrun, queuing locally 22. Jul 26 10:11:16 localhost pulseaudio[26721]: (51979.154| 0.000) asyncq.c: q overrun, queuing locally 23. Jul 26 10:11:16 localhost pulseaudio[26721]: (51979.184| 0.029) asyncq.c: q overrun, queuing locally 24. Jul 26 10:11:16 localhost pulseaudio[26721]: (51979.214| 0.030) asyncq.c: q overrun, queuing locally 25. Jul 26 10:11:20 localhost pulseaudio[26721]: (51984.088| 4.874) ratelimit.c: 85 events suppressed 26. Jul 26 10:11:20 localhost pulseaudio[26721]: (51984.088| 0.000) asyncq.c: q overrun, queuing locally 27. Jul 26 10:11:20 localhost pulseaudio[26721]: (51984.088| 0.000) asyncq.c: q overrun, queuing locally 28. Jul 26 10:11:20 localhost pulseaudio[26721]: (51984.118| 0.029) asyncq.c: q overrun, queuing locally 29. Jul 26 10:11:20 localhost pulseaudio[26721]: (51984.118| 0.000) asyncq.c: q overrun, queuing locally 30. Jul 26 10:11:21 localhost pulseaudio[26721]: (51984.148| 0.029) asyncq.c: q overrun, queuing locally 31. Jul 26 10:11:21 localhost pulseaudio[26721]: (51984.148| 0.000) asyncq.c: q overrun, queuing locally 32. Jul 26 10:11:21 localhost pulseaudio[26721]: (51984.148| 0.000) asyncq.c: q overrun, queuing locally 33. Jul 26 10:11:21 localhost pulseaudio[26721]: (51984.178| 0.029) asyncq.c: q overrun, queuing locally 34. Jul 26 10:11:21 localhost pulseaudio[26721]: (51984.178| 0.000) asyncq.c: q overrun, queuing locally 35. Jul 26 10:11:21 localhost pulseaudio[26721]: (51984.208| 0.029) asyncq.c: q overrun, queuing locally 36. Jul 26 10:11:21 localhost pulseaudio[26721]: (51984.208| 0.000) asyncq.c: q overrun, queuing locally 37. Jul 26 10:11:21 localhost pulseaudio[26721]: (51984.738| 0.530) alsa-sink.c: Increasing wakeup watermark to 130.11 ms 38. Jul 26 10:11:25 localhost pulseaudio[26721]: (51988.239| 3.500) alsa-sink.c: Increasing wakeup watermark to 140.00 ms

1 0

[PATCH 1/1] ALSA: hda: Fixup commit 79c944ad (caused codec initialization failure)
by David Henningsson 27 Jul '10

27 Jul '10

Commit 79c944ad caused codecs not to be detected in some cases, found on several Realtek codecs. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 13 +++++++------ 1 files changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8cd5da8..710f61a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -724,23 +724,24 @@ static int get_codec_name(struct hda_codec *codec) */ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec) { - int i, total_nodes, function_id; + int i, total_nodes, function_id, param_val; hda_nid_t nid; total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - function_id = snd_hda_param_read(codec, nid, + param_val = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); + function_id = param_val & 0xff; switch (function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; - codec->afg_function_id = function_id & 0xff; - codec->afg_unsol = (function_id >> 8) & 1; + codec->afg_function_id = function_id; + codec->afg_unsol = (param_val >> 8) & 1; break; case AC_GRP_MODEM_FUNCTION: codec->mfg = nid; - codec->mfg_function_id = function_id & 0xff; - codec->mfg_unsol = (function_id >> 8) & 1; + codec->mfg_function_id = function_id; + codec->mfg_unsol = (param_val >> 8) & 1; break; default: break; -- 1.7.0.4 --------------040206080406060608060907--

1 0

[alsa-devel] asoc multi-component: CPU DAI name is wrong
by Timur Tabi 26 Jul '10

26 Jul '10

Liam, I have this in my machine driver: machine_data->dai[0].cpu_dai_name = strrchr(np->full_name, '/') + 1; printk(KERN_INFO "%s:%u cpu_dai_name=%s\n", __func__, __LINE__, machine_data->dai[0].cpu_dai_name); ... platform_set_drvdata(sound_device, &machine_data->card); ret = platform_device_add(sound_device); which displays this message: mpc8610_hpcd_probe:316 cpu_dai_name=ssi@16000 So my machine driver says the CPU DAI name is "ssi@16000", and my CPU driver does the same thing. However, I put some printks in soc_bind_dai_link, and this is what I see: soc_bind_dai_link:1136 cpu_dai->name=e0016000.ssi dai_link->cpu_dai_name=ssi@16000 soc_bind_dai_link:1136 cpu_dai->name=cs4270-hifi dai_link->cpu_dai_name=ssi@16000 soc_bind_dai_link:1159 codec->name=0-004f dai_link->codec_name=cs4270-codec.0-004f soc_bind_dai_link:1193 platform->name=e0021180.dma-channel dai_link->platform_name=/soc@e0000000/dma@21300/dma-channel@0 soc_bind_dai_link:1193 platform->name=e0021100.dma-channel dai_link->platform_name=/soc@e0000000/dma@21300/dma-channel@0 soc_bind_dai_link:1136 cpu_dai->name=e0016000.ssi dai_link->cpu_dai_name=ssi@16000 soc_bind_dai_link:1136 cpu_dai->name=cs4270-hifi dai_link->cpu_dai_name=ssi@16000 soc_bind_dai_link:1159 codec->name=0-004f dai_link->codec_name=cs4270-codec.0-004f soc_bind_dai_link:1193 platform->name=e0021180.dma-channel dai_link->platform_name=/soc@e0000000/dma@21300/dma-channel@1 soc_bind_dai_link:1193 platform->name=e0021100.dma-channel dai_link->platform_name=/soc@e0000000/dma@21300/dma-channel@1 Where in the world is "e0016000.ssi" coming from? Why is ASoC ignoring my .cpu_dai_name value?

2 4

[alsa-devel] hda_intel: too quiet sound (regression)
by Jiri Slaby 26 Jul '10

26 Jul '10

Hi, after I upgraded from 2.6.34-rc5-mm1 to 2.6.35-rc5-mm1 (there were other issues in between them), the sound from my soundcard is very very quiet. alsa-info: http://www.alsa-project.org/db/?f=dcadb29a2d5d047ecf5545792ac3239dc0c8c7f0 Any ideas? thanks, -- js

3 4

[alsa-devel] announcing envy24control, mudita (*) edition.
by Niels Mayer 26 Jul '10

26 Jul '10

Summary of updates from envy24control 0.6.0 (GIT HEAD) to "1.0.0": (0) After a decade, incremented version to 1.0.0 (**) (1) Implemented missing "Peak Hold" functionality in meters and reimplemented meters for increased efficiency and lower X resource usage. (see http://www.linuxaudio.org/mailarchive/lad/2010/7/12/171535 & https://bugzilla.redhat.com/show_bug.cgi?id=602903 ) (2) All volumes are represented as decibels, including the 0 to -48dB range of the hardware peak-meters, the 0 -to- -144dB attenuation for all inputs to the digital mixer, the 0 -to- -63dB attenuation of the analog DAC, and the +18 -to- -63dB attenuation/amplification of the analog ADC. (3) All gtk "scale" widgets have dB legends; the "PageUp" "PageDown" keys allow rapid movement between the marked levels. (4) Got rid of myriad compile warnings and other minor fixes across codebase. Some screenshots: http://nielsmayer.com/envy24control/Screenshot-Envy24Control-AnalogVolume.p… http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorInputs.… http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorPCM.png ------------ To the ALSA project: please consider this patch to alsa-tools' envy24control (**): http://nielsmayer.com/envy24control/envy24control-0.6-to-1.0.patch (patch to 'envy24control' from GIT trunk/head of alsa-tools) http://nielsmayer.com/envy24control/envy24control-1.0.README (summary of changes from 0.6.0 to 1.0.0) ------------ Those wanting to compile directly, or run a 64 bit linux binary I've built: http://nielsmayer.com/envy24control/envy24control-1.0.tar.gz (full directory, just follow README directions to build/install) http://nielsmayer.com/envy24control/envy24control-1.0-fc12-x86_64.tar.gz (x86_64 binary that should work on fedora12 and equivalent OpenSuse release) I'd appreciate any testing results or comments on this "1.0.0" release. In particular, I'd like some assurance that the dB markings on sliders in "Analog Volume" panel are correct (compared to values reported by 'alsamixer'). I'm looking for testing with following devices (as I think my testing covers code for M-Audio Delta 44 & Delta 66, Terratec Dmx6fire & EWX2496) specifically: M-Audio Delta 1010, M-Audio Audiophile 2496, M-Audio Delta 1010LT TerraTec EWS 88MT, TerraTec EWS 88D, TerraTec Phase 88, Hoontech SoundTrack DSP 24 (all variants). Thanks, Niels http://nielsmayer.com PS: (*) http://en.wikipedia.org/wiki/Envy#In_philosophy :: "mudita, taking joy in the good fortune of another. This virtue is considered the antidote to envy and the opposite of schadenfreude." (**) http://git.alsa-project.org/?p=alsa-tools.git;a=tree;f=envy24control;h=d5a5…

2 2

Re: [alsa-devel] [Fwd: Re: [pulseaudio-discuss] pulseaudio debug mode]
by Raymond Yau 26 Jul '10

26 Jul '10

2010/7/26 Chris <cpollock(a)embarqmail.com> > Raymond, attached is a post I made today to the pulseaudio list. > > From: Chris <cpollock(a)embarqmail.com> > To: pulseaudio-discuss(a)mail.0pointer.de > Date: Sun, 25 Jul 2010 12:15:02 -0500 > Subject: Re: [pulseaudio-discuss] pulseaudio debug mode > On Fri, 2010-07-23 at 08:58 +0100, Colin Guthrie wrote: > > 'Twas brillig, and Chris at 23/07/10 02:04 did gyre and gimble: > > > What is the best way to start PA for debugging and still have all the > > > usual clients running? > > > > If you mean having all the clients connect (e.g. applications with > > libcanberra support or similar for sound events), then there are > > basically two ways. > > > > The first is as Luke suggests. These clients will automatically > > reconnect to PA if they need to (provided you have a vaguely recent > > libcanberra), after it is restarted and run in debug mode. > > > > Alternatively you can simply set debug-level to "debug" in daemon.conf > > (in /etc/pulse or ~/.pulse), and then "grep pulseaudio /var/log/messages" > > > > Col > > > > Colin, link below is for debug output also some other output. Anything > look out of place that would cause the overruns? > > http://pastebin.com/ZWSWmXZt > -- > Most likely you will need to set debug-level to "debug" in daemon.conf logout and login , and check the /var/log/messages I suspect it occur after PA server abort and auto spawn , so please don't just post the PA log starting from "asyncq overrun" , you need to post from the last two PA startup sequence before the error occur since we need to know why PA auto spawn

3 11

[alsa-devel] [PATCH] eukrea-tlv320: add support for cpuimx35sd
by Eric Bénard 26 Jul '10

26 Jul '10

Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 2f0d6d3..52dac5e 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -32,6 +32,6 @@ config SND_SOC_EUKREA_TLV320 select SND_SOC_TLV320AIC23 help Enable I2S based access to the TLV320AIC23B codec attached - to the SSI4 interface + to the SSI interface endif # SND_IMX_SOC diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 45f5e4b..f15dfbd 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -101,7 +101,8 @@ static int __init eukrea_tlv320_init(void) { int ret; - if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()) + if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd() + && !machine_is_eukrea_cpuimx35sd()) /* return happy. We might run on a totally different machine */ return 0; -- 1.6.3.3

4 5

Re: [alsa-devel] "Playback open error: Inappropriate ioctl for device" in alsa-lib-1.0.22/test/*.c
by John Simpson 26 Jul '10

26 Jul '10

On 7/21/10, John Simpson <john(a)swajime.com> wrote: > On 7/21/10, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> John Simpson wrote: >>> Playback open error: Inappropriate ioctl for device >> >> Make sure that both your kernel and your alsa-lib use the correct >> compiler and compiler flags (where "correct" usually means "whatever >> the vendor's distribution uses"). > > You've gone way over my head ... > How do I find out what flags were used to compile the kernel? > >> >>> This time, checking out https://vcs.maemo.org/svn/dsp-alsa >> >> It would be nice to know what the changes from the original alsa-lib >> are; in theory, they should go upstream. > > I agree ... but it looks like that is an abandoned project; at > https://garage.maemo.org/plugins/scmsvn/viewcvs.php/?root=dsp-alsa it > says the last modification was 3 years ago. The text file "version" > says 1.0.10 > > I found that "apt-get source libasound2" gives me a newer version: > [1|user at Nokia-N810-43-7|~/src/alsa-lib-1.0.14sa/test]cat > ../../alsa-lib_1.0.14sa-2maemo2.dsc > Format: 1.0 > Source: alsa-lib > Version: 1.0.14sa-2maemo2 > Binary: lib32asound2-dev, lib64asound2, lib32asound2, libasound2-dev, > libasound2, libasound2-doc, libasound2-dbg, lib64asound2-dev > Maintainer: Debian ALSA Maintainers <pkg-alsa-devel at > lists.alioth.debian.org> > Architecture: any > Standards-Version: 3.7.2 > Build-Depends: debhelper (>= 4.2.32), quilt (>= 0.40), autotools-dev, > doxygen, libc6-dev-powerpc [ppc64], libc6-dev-i386 [amd64], > libc6-dev-ppc64 [powerpc], libc6-dev-s390x [s390], libc6-dev-sparc64 > [sparc], lib32gcc1 [amd64 ppc64], libc6-dev (>= 2.5) [armel] > Vcs-Browser: http://svn.debian.org/wsvn/pkg-alsa/trunk/alsa-lib/ > Uploaders: Jordi Mallach <jordi at debian.org>, Mikael Magnusson > <mikma at users.sourceforge.net>, Elimar Riesebieter <riesebie at > lxtec.de>, > Marc-Andre Lureau <marc-andre.lureau at nokia.com> > Vcs-Svn: svn://svn.debian.org/pkg-alsa/trunk/alsa-lib > Files: > 5542a18a1479e4de4b9890e30e94588e 1099165 alsa-lib_1.0.14sa.orig.tar.gz > fc02aa3dbf2f66fb090e02ada703cc1d 47455 alsa-lib_1.0.14sa-2maemo2.diff.gz > > This version seems to install correctly, but still gives me no sound. > The full log of my latest effort is here -> > http://www.swajime.com/n810/alsa-lib_1.0.14sa-2maemo2.log >> >>> $ cd dsp-alsa/trunk/alsa-lib >>> #### cvscompile does not succeed >> >> Why? > > I honestly don't know. :-( It looks like a bug in m4? Here is the > log for cvscompile: > [1|root@Nokia-N810-43-7|/home/user/src/dsp-alsa/trunk/alsa-lib]cat > cvscompile > #!/bin/bash > > touch ltconfig > libtoolize --force --copy --automake > aclocal $ACLOCAL_FLAGS > autoheader > automake --foreign --copy --add-missing > touch depcomp # seems to be missing for old automake > autoconf > export CFLAGS='-O2 -Wall -W -pipe -g' > echo "CFLAGS=$CFLAGS" > echo "./configure $@" > ./configure $@ || exit 1 > unset CFLAGS > if [ -z "$CVSCOMPILE_NO_MAKE" ]; then > make > fi > [1|user@Nokia-N810-43-7|~/src/dsp-alsa/trunk/alsa-lib]bash -v ./cvscompile > #!/bin/bash > > touch ltconfig > libtoolize --force --copy --automake > aclocal $ACLOCAL_FLAGS > aclocal: configure.in: 41: macro `AM_PROG_LIBTOOL' not found in library > autoheader > configure.in:41: /usr/local/bin/m4: ERROR: Copying inserted file: No > space left on device > autom4te: /usr/local/bin/m4 failed with exit status: 1 > autoheader: /usr/local/bin/autom4te failed with exit status: 1 > automake --foreign --copy --add-missing > configure.in:41: /usr/local/bin/m4: ERROR: Copying inserted file: No > space left on device > autom4te: /usr/local/bin/m4 failed with exit status: 1 > configure.in: no proper invocation of AM_INIT_AUTOMAKE was found. > configure.in: You should verify that configure.in invokes AM_INIT_AUTOMAKE, > configure.in: that aclocal.m4 is present in the top-level directory, > configure.in: and that aclocal.m4 was recently regenerated (using aclocal). > automake: no `Makefile.am' found or specified > touch depcomp # seems to be missing for old automake > autoconf > configure.in:41: /usr/local/bin/m4: ERROR: Copying inserted file: No > space left on device > autom4te: /usr/local/bin/m4 failed with exit status: 1 > export CFLAGS='-O2 -Wall -W -pipe -g' > echo "CFLAGS=$CFLAGS" > CFLAGS=-O2 -Wall -W -pipe -g > echo "./configure $@" > ./configure > ./configure $@ || exit 1 > ./cvscompile: line 13: ./configure: No such file or directory > [1|user@Nokia-N810-43-7|~/src/dsp-alsa/trunk/alsa-lib] > >> >>> $ .libs/pcm -D 'default' -vf 440 >>> Playback device is default >>> Stream parameters are 44100Hz, S16_LE, 1 channels >>> Sine wave rate is 440.0000Hz >>> Using transfer method: write >>> >>> Still no sound :-( >> >> Mixer controls not set? (amixer and alsamixer are in alsa-utils) > > I haven't been able to find alsa-utils in any maemo/diablo repositories. I found that alsa-tools is in the fremantle repo: http://repository.maemo.org/pool/fremantle/free/a/alsa-utils/ It bricks the n810 :-( Fremantle is not compatible with the n810 as I remember.

1 0

[alsa-devel] Realtek alc269 questions
by David Henningsson 26 Jul '10

26 Jul '10

I'm trying to fix the sound on some Dell Optiplex machines, which are running the ALC269 chip. I can "fix" them with some hda-verb commands, but that won't get me auto-mute, so I'll have to make a model (or possibly more than one) for them. Anyway, I have a pair of questions: 1) In dmesg, there is a "SKU not ready" message at startup. What is SKU, and is it a problem that it is not ready? 2) There are two DACs, and all models so far seem to use NID 0x03 for headphones and NID 0x02 for everything else. But is there a single stream going through the controller and then splitted into two on the codec side, or is just one of the DACs activated at a time, depending on the jack sensing state? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 3

[alsa-devel] problems about dummy alsa driver
by longer 26 Jul '10

26 Jul '10

Hi all , I've written an ALSA driver(only support 8K sample rate), which register two dummy sound cards.The main function of this driver is to exchange audio data of that two dummy sound cards. In other words , the driver route audio data of dummy sound card (PLAYBACK) to the other (CAPTURE),and vice versa. And I've written some test programs based on my ALSA driver . When the sample rate in application is set to be 8K , the driver work well. But when requiring rate conversion , the driver can't get any data. And , when tracing the code of alsa-lib (1.0.23), I found that : In case of rate conversion, It mmap an address from my driver, but writing data to another address instead of that mmaped address. I think this is why my driver don't get any audio data. But I don't know why it don't write audio data to that mmaped address. I am looking forward to your reply. And any advices will help me a lot. Thank you, longer zhou

1 0

[alsa-devel] Compiling alsa-utils
by Pavel Hofman 26 Jul '10

26 Jul '10

Hi, after clean git clone of git://git.alsa-project.org/alsa-utils.git I got the following error: pavel@sara:~/install/alsa/alsa-utils$ autoheader configure.in:268: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ../../lib/autoconf/status.m4:311: AC_CONFIG_FILES is expanded from... configure.in:268: the top level autom4te: /usr/bin/m4 failed with exit status: 1 autoheader: '/usr/bin/autom4te' failed with exit status: 1 After deleting po/Makefile.in from the list of AC_OUTPUT in configure.in:268 the compilation proceeded fine. Ubuntu 9.10. Thanks, Pavel Hofman.

2 3

[alsa-devel] Merging alsa-firmware into linux-firmware
by David Henningsson 26 Jul '10

26 Jul '10

Hi, Chase Douglas (@canonical) and I plan to take what's left of alsa-firmware and try to merge it into linux-firmware, this might happen now or in the coming weeks. This will make alsa-firmware obsolete as all firmware will be provided by linux-firmware instead. Any words of disagreement, warnings or encouragement before we do so? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 16

Re: [alsa-devel] [PATCH] oss: au1550_ac97: simplify au1550_delay()
by Takashi Iwai 26 Jul '10

26 Jul '10

At Mon, 26 Jul 2010 12:28:32 +0400, Kulikov Vasiliy wrote: > > au1550_delay() uses loop with schedule_timeout() to unconditionally wait > for msec. Use schedule_timeout_uninteruptible() instead. > > Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> Thanks, applied both patches. Takashi > --- > sound/oss/au1550_ac97.c | 11 +---------- > 1 files changed, 1 insertions(+), 10 deletions(-) > > diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c > index 0fd256c..c4a4cdc 100644 > --- a/sound/oss/au1550_ac97.c > +++ b/sound/oss/au1550_ac97.c > @@ -163,19 +163,10 @@ ld2(unsigned int x) > static void > au1550_delay(int msec) > { > - unsigned long tmo; > - signed long tmo2; > - > if (in_interrupt()) > return; > > - tmo = jiffies + (msec * HZ) / 1000; > - for (;;) { > - tmo2 = tmo - jiffies; > - if (tmo2 <= 0) > - break; > - schedule_timeout(tmo2); > - } > + schedule_timeout_uninterruptible(msecs_to_jiffies(msec)); > } > > static u16 > -- > 1.7.0.4 >

1 0

[alsa-devel] [PATCH] Ensure codec patch files are checked for the correct codec ID
by David Henningsson 26 Jul '10

26 Jul '10

This one just popped out when I read the source code - or is there a reason for parsing vendorid and subid without caring about the result? // David

4 5

Re: [alsa-devel] Which project to choose?
by Raymond Yau 26 Jul '10

26 Jul '10

2010/7/14 Chris <cpollock(a)embarqmail.com> > On Wed, 2010-07-14 at 10:09 +0800, Raymond Yau wrote: > > > > > > > > > > You need to provide a full pulseaudio log and a test case > > which can > > > reproduce your problem > > > > > > open a new terminal > > > > > > pulseaudio -k ; pulseaudio -vvvvv > > > > Where should I post this Raymond? > > > > > > > > AFAIK , this mailing list has a limit of the message size > > > > > > BTW , > > > > did the program abort when you run alsa-time-test and pipe the output > > to "tail -n 500" > > > > > > > > http://thread.gmane.org/gmane.linux.alsa.devel/60371/focus=60520 > > > > Beware that the program generate a log of output > > Raymond, I've ran bzip2 on the output of pulseaudio -k ; pulseaudio > -vvvvv and it went from 129k to 9k. Should I fwd to you or post on the > list? > > -- > appl_ptr is also behind the hw_ptr in 0.9.21-0.1mdv2010.0 https://qa.mandriva.com/show_bug.cgi?id=56473#c15 Apr 12 15:21:43 localhost pulseaudio[13701]: alsa-util.c: appl_ptr : 28465776 Apr 12 15:21:43 localhost pulseaudio[13701]: alsa-util.c: hw_ptr : 28555040 and also appl_ptr is also behind the hw_ptr for snd_via82xx https://qa.mandriva.com/show_bug.cgi?id=56473#c0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: appl_ptr : 767919576 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: hw_ptr : 767990616

4 32

Re: [alsa-devel] [Fwd: Re: [pulseaudio-discuss] pulseaudio debug mode]
by Raymond Yau 26 Jul '10

26 Jul '10

2010/7/26 Chris <cpollock(a)embarqmail.com> > Raymond, attached is a post I made today to the pulseaudio list. > > -- > Chris > KeyID 0xE372A7DA98E6705C > > > > ---------- 轉寄訊息 ---------- > From: Chris <cpollock(a)embarqmail.com> > To: pulseaudio-discuss(a)mail.0pointer.de > Date: Sun, 25 Jul 2010 12:15:02 -0500 > Subject: Re: [pulseaudio-discuss] pulseaudio debug mode > On Fri, 2010-07-23 at 08:58 +0100, Colin Guthrie wrote: > > 'Twas brillig, and Chris at 23/07/10 02:04 did gyre and gimble: > > > What is the best way to start PA for debugging and still have all the > > > usual clients running? > > > > If you mean having all the clients connect (e.g. applications with > > libcanberra support or similar for sound events), then there are > > basically two ways. > > > > The first is as Luke suggests. These clients will automatically > > reconnect to PA if they need to (provided you have a vaguely recent > > libcanberra), after it is restarted and run in debug mode. > > > > Alternatively you can simply set debug-level to "debug" in daemon.conf > > (in /etc/pulse or ~/.pulse), and then "grep pulseaudio /var/log/messages" > > > > Col > > > > Colin, link below is for debug output also some other output. Anything > look out of place that would cause the overruns? > > http://pastebin.com/ZWSWmXZt > ens1371 is ready to enter running state after PA server call snd_pcm_hw_params() The PA server only call snd_pcm_dump() at the probing phase to create working profile for the alsa sound cards, and snd_pcm_dump() indicate that appl_ptr is behind hw_ptr, but the playing thread is started after snd_pcm_dump() . This mean that some PA clients had connected and send audio to PA server before the first sink finished it setup since ens1371 is card 0 ( the fist sink device )

1 0

Re: [alsa-devel] [LAD] announcing envy24control, mudita (*) edition.
by Niels Mayer 25 Jul '10

25 Jul '10

On Sun, Jul 25, 2010 at 12:21 PM, Paul Davis <paul(a)linuxaudiosystems.com> wrote: >> (2) All volumes are represented as decibels, including the 0 to -48dB > i don't own one, so its not of much concern to me, but dB<WHAT> ? The http://en.wikipedia.org/wiki/Decibel is a unitless measurement. :-) However, in this case there are a few different semi-implicit http://en.wikipedia.org/wiki/DBFS and other markings that I left off in the name of saving on screen-real-estate and reduced redundancy. Tooltips across-the board would be a very helpful addition to the envy24control 1.0.0 interface. On the other hand, these are exactly the same dB values as reported by ALSA's "amixer" and "alsamixer" so such complaints should go upstream. :-) Specifically: (1) The db markings in "Analog Volumes" correspond to analog levels that can be set for the DAC or ADC's. e.g. 0 -to- -63dB attenuation of the output of the analog DAC, and the +18 -to- -63dB attenuation/amplification of the analog ADC. These are the exact same dB readings reported by ALSA, e.g. "amixer -c M66": > Simple mixer control 'ADC',0 > Limits: 0 - 163 > Mono: 163 [100%] [18.00dB] > Simple mixer control 'ADC',1 > Limits: 0 - 163 > Mono: 163 [100%] [18.00dB] > Simple mixer control 'ADC',2 > Limits: 0 - 163 > Mono: 152 [93%] [12.50dB] > Simple mixer control 'ADC',3 > Limits: 0 - 163 > Mono: 152 [93%] [12.50dB] (2) The "0 to -48dB" peak metering levels are based on values from hardware peak metering (per http://alsa.cybermirror.org/manuals/icensemble/envy24.pdf ) > Peak data derived from the absolute value of 9 msb. 00h min - FFh max > volume. Reading the register resets the meter to 00h. When "0dBFS" reads on the peak meters in the new envy24control, it means 0dBFS relative to these 9MSB's. (see the red "0dBFS" labels in http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorInputs.… or http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorPCM.png ). Note that the signal would be able to go 7 or 15 LSB's higher before reaching actual full-scale peak. It would be more useful if there was a way for the hardware meters to detect actual clipping, e.g. value=256, whereas normally the meters report values from 0-255. For example "amixer -c M66 cget iface=PCM,name='Multi Track Peak',numid=45" returns: > type=INTEGER,access=r-------,values=22,min=0,max=255,step=0 > : values=0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,255,198,255,198 For proper metering, I suggest using Fons' excellent jkmeter with it's nice (and free) implementation of the K-System http://en.wikipedia.org/wiki/File:K-14_spectrafoo.jpg (caption: "digital audio meters in the Metric Halo application, called SpectraFoo"). (3) For the envy24's built in digital mixer, the 8PCMs + 2*SPDIF out, and up to 8 inputs + 2SPDIF inputs, all send to the digital mixer via 24dB attenuators. Seer http://nielsmayer.com/envy24control/envy24mixer-architecture.png for details. Thus all mixer input sliders have "0 -to- -144dB attenuation." (from http://alsa.cybermirror.org/manuals/icensemble/envy24.pdf ) This is what the above manual says about the Envy24's digital mixer: > 4.5.5 Multi-Track Digital Monitoring > > The Envy24 integrates a 36-bit resolution digital hardware mixer. The > width of the data path is strictly to ensure that during processing of > all the channels, under any condition, no resolution is lost. The > dynamic range of the end user system will be limited by the range of the > physical output devices used. In order to maintain identical gain to the > input stream (i.e. 0dB), the resulting 24-bit is not msb-aligned to the > 36-bit. The overflow bits correspond to the analog distortion due to > saturation. The user would need to reduce the overall attenuation of the > inputs to avoid clipping. Insertion of the digital mixer adds only a > single sample cycle delay with respect to the original data. This > extremely low latency all digital mixer provides monitoring > functionality and can replace a traditional external analog input > mixer. There are 20 independent audio data streams to mix and control > the volume. -- Niels http://nielsmayer.com

1 0

[alsa-devel] ASoC update for 2.6.36
by Mark Brown 25 Jul '10

25 Jul '10

The following changes since commit 49e704279953b652ec6adc7f5dec412ad118a938: Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2010-07-21 15:01:07 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Kuninori Morimoto (1): ASoC: da7210: Add HeadPhone Playback Volume control sound/soc/codecs/da7210.c | 31 +++++++++++++++++++++++++++---- 1 files changed, 27 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH v3 0/1] Codec driver for TI WL1273 FM Radio.
by Matti J. Aaltonen 24 Jul '10

24 Jul '10

Hello all. Thank you for comments and thanks to Peter for advice on constraints and ALSA in general. New in this version: 1. Now constraints are used in the start-up function. It wasn't as big a deal as I thought... Also now the codec is always in full duplex mode. But it still refuses to do the wrong thing with FM as real full duplex is then impossible. 2. Fixed the exported mode support function. The name in now wl1273_get_format and it returns the audio interface format. 3. Removed the remaining obsolete functions. 4. Not actually new but... >> I didn't do this yet. But could you give me the git URL and then >> I'll resend? > git://git.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git I tried the corresponding http URL http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git but got fatal: http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git/info/refs not found: did you run git update-server-info on the server? Cheers, Matti Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec. sound/soc/codecs/wl1273.c | 598 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 43 ++++ 2 files changed, 641 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h

2 5

[alsa-devel] [PATCH 6/6] pcmcia: remove cs_types.h
by Dominik Brodowski 23 Jul '10

23 Jul '10

Remove cs_types.h which is no longer needed: Most definitions aren't used at all, a few can be made away with, and two remaining definitions (typedefs, unfortunatley) may be moved to more specific places. CC: netdev(a)vger.kernel.org CC: linux-wireless(a)vger.kernel.org CC: linux-ide(a)vger.kernel.org CC: linux-usb(a)vger.kernel.org CC: laforge(a)gnumonks.org CC: linux-mtd(a)lists.infradead.org CC: linux-bluetooth(a)vger.kernel.org CC: alsa-devel(a)alsa-project.org CC: linux-serial(a)vger.kernel.org Signed-off-by: Dominik Brodowski <linux(a)dominikbrodowski.net> --- Documentation/pcmcia/driver-changes.txt | 5 +++ drivers/ata/pata_pcmcia.c | 1 - drivers/bluetooth/bluecard_cs.c | 1 - drivers/bluetooth/bt3c_cs.c | 1 - drivers/bluetooth/btuart_cs.c | 1 - drivers/bluetooth/dtl1_cs.c | 1 - drivers/char/pcmcia/cm4000_cs.c | 1 - drivers/char/pcmcia/cm4040_cs.c | 1 - drivers/char/pcmcia/ipwireless/main.h | 1 - drivers/char/pcmcia/ipwireless/tty.h | 1 - drivers/char/pcmcia/synclink_cs.c | 1 - drivers/ide/ide-cs.c | 1 - drivers/isdn/hardware/avm/avm_cs.c | 1 - drivers/isdn/hisax/avma1_cs.c | 1 - drivers/isdn/hisax/elsa_cs.c | 1 - drivers/isdn/hisax/sedlbauer_cs.c | 1 - drivers/isdn/hisax/teles_cs.c | 1 - drivers/mmc/host/sdricoh_cs.c | 1 - drivers/mtd/maps/pcmciamtd.c | 1 - drivers/net/pcmcia/3c574_cs.c | 1 - drivers/net/pcmcia/3c589_cs.c | 1 - drivers/net/pcmcia/axnet_cs.c | 1 - drivers/net/pcmcia/com20020_cs.c | 1 - drivers/net/pcmcia/fmvj18x_cs.c | 1 - drivers/net/pcmcia/ibmtr_cs.c | 1 - drivers/net/pcmcia/nmclan_cs.c | 1 - drivers/net/pcmcia/pcnet_cs.c | 5 +-- drivers/net/pcmcia/smc91c92_cs.c | 1 - drivers/net/pcmcia/xirc2ps_cs.c | 1 - drivers/net/wireless/airo_cs.c | 1 - drivers/net/wireless/atmel_cs.c | 1 - drivers/net/wireless/b43/pcmcia.c | 1 - drivers/net/wireless/hostap/hostap_cs.c | 3 +- drivers/net/wireless/libertas/if_cs.c | 1 - drivers/net/wireless/orinoco/orinoco_cs.c | 1 - drivers/net/wireless/orinoco/spectrum_cs.c | 1 - drivers/net/wireless/ray_cs.c | 1 - drivers/net/wireless/wl3501_cs.c | 10 +----- drivers/parport/parport_cs.c | 1 - drivers/pcmcia/au1000_generic.h | 1 - drivers/pcmcia/au1000_pb1x00.c | 2 - drivers/pcmcia/cistpl.c | 1 - drivers/pcmcia/cs.c | 1 - drivers/pcmcia/db1xxx_ss.c | 1 - drivers/pcmcia/ds.c | 1 - drivers/pcmcia/i82092.c | 1 - drivers/pcmcia/i82365.c | 1 - drivers/pcmcia/m32r_cfc.c | 1 - drivers/pcmcia/m32r_pcc.c | 1 - drivers/pcmcia/m8xx_pcmcia.c | 1 - drivers/pcmcia/pcmcia_cis.c | 1 - drivers/pcmcia/pcmcia_resource.c | 1 - drivers/pcmcia/pd6729.c | 1 - drivers/pcmcia/pxa2xx_base.c | 1 - drivers/pcmcia/rsrc_iodyn.c | 1 - drivers/pcmcia/rsrc_mgr.c | 1 - drivers/pcmcia/rsrc_nonstatic.c | 1 - drivers/pcmcia/sa1100_generic.c | 1 - drivers/pcmcia/soc_common.h | 1 - drivers/pcmcia/socket_sysfs.c | 1 - drivers/pcmcia/tcic.c | 1 - drivers/pcmcia/xxs1500_ss.c | 1 - drivers/pcmcia/yenta_socket.c | 1 - drivers/scsi/pcmcia/aha152x_stub.c | 1 - drivers/scsi/pcmcia/fdomain_stub.c | 1 - drivers/scsi/pcmcia/nsp_cs.c | 1 - drivers/scsi/pcmcia/qlogic_stub.c | 1 - drivers/scsi/pcmcia/sym53c500_cs.c | 1 - drivers/serial/serial_cs.c | 1 - drivers/ssb/main.c | 1 - drivers/ssb/pcmcia.c | 1 - drivers/ssb/scan.c | 1 - drivers/staging/comedi/drivers/cb_das16_cs.c | 1 - drivers/staging/comedi/drivers/das08_cs.c | 1 - drivers/staging/comedi/drivers/ni_daq_700.c | 1 - drivers/staging/comedi/drivers/ni_daq_dio24.c | 1 - drivers/staging/comedi/drivers/ni_labpc_cs.c | 1 - drivers/staging/comedi/drivers/ni_mio_cs.c | 1 - drivers/staging/comedi/drivers/quatech_daqp_cs.c | 1 - drivers/staging/wlags49_h2/wl_cs.c | 1 - drivers/staging/wlags49_h2/wl_internal.h | 1 - drivers/telephony/ixj_pcmcia.c | 1 - drivers/usb/host/sl811_cs.c | 5 +-- include/pcmcia/cistpl.h | 2 + include/pcmcia/cs.h | 10 +----- include/pcmcia/cs_types.h | 40 ---------------------- include/pcmcia/ds.h | 3 +- include/pcmcia/ss.h | 1 - sound/pcmcia/pdaudiocf/pdaudiocf.h | 1 - sound/pcmcia/vx/vxpocket.h | 1 - 90 files changed, 14 insertions(+), 151 deletions(-) delete mode 100644 include/pcmcia/cs_types.h diff --git a/Documentation/pcmcia/driver-changes.txt b/Documentation/pcmcia/driver-changes.txt index 61bc4e9..ff5f0be 100644 --- a/Documentation/pcmcia/driver-changes.txt +++ b/Documentation/pcmcia/driver-changes.txt @@ -1,4 +1,9 @@ This file details changes in 2.6 which affect PCMCIA card driver authors: +* No dev_info_t, no cs_types.h (as of 2.6.36) + dev_info_t and a few other typedefs are removed. No longer use them + in PCMCIA device drivers. Also, do not include pcmcia/cs_types.h, as + this file is gone. + * No dev_node_t (as of 2.6.35) There is no more need to fill out a "dev_node_t" structure. diff --git a/drivers/ata/pata_pcmcia.c b/drivers/ata/pata_pcmcia.c index 118c28e..3dcb2b1 100644 --- a/drivers/ata/pata_pcmcia.c +++ b/drivers/ata/pata_pcmcia.c @@ -34,7 +34,6 @@ #include <linux/ata.h> #include <linux/libata.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/bluetooth/bluecard_cs.c b/drivers/bluetooth/bluecard_cs.c index 6d34f40..eb085de 100644 --- a/drivers/bluetooth/bluecard_cs.c +++ b/drivers/bluetooth/bluecard_cs.c @@ -39,7 +39,6 @@ #include <linux/skbuff.h> #include <linux/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/bluetooth/bt3c_cs.c b/drivers/bluetooth/bt3c_cs.c index 21e05fd..457b603 100644 --- a/drivers/bluetooth/bt3c_cs.c +++ b/drivers/bluetooth/bt3c_cs.c @@ -45,7 +45,6 @@ #include <linux/device.h> #include <linux/firmware.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/bluetooth/btuart_cs.c b/drivers/bluetooth/btuart_cs.c index 4ed7288..e7e0a17 100644 --- a/drivers/bluetooth/btuart_cs.c +++ b/drivers/bluetooth/btuart_cs.c @@ -41,7 +41,6 @@ #include <asm/system.h> #include <asm/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/bluetooth/dtl1_cs.c b/drivers/bluetooth/dtl1_cs.c index ef044d5..7c94aad 100644 --- a/drivers/bluetooth/dtl1_cs.c +++ b/drivers/bluetooth/dtl1_cs.c @@ -41,7 +41,6 @@ #include <asm/system.h> #include <asm/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/char/pcmcia/cm4000_cs.c b/drivers/char/pcmcia/cm4000_cs.c index e7956ac..a8be2a7 100644 --- a/drivers/char/pcmcia/cm4000_cs.c +++ b/drivers/char/pcmcia/cm4000_cs.c @@ -34,7 +34,6 @@ #include <linux/uaccess.h> #include <linux/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/char/pcmcia/cm4040_cs.c b/drivers/char/pcmcia/cm4040_cs.c index c0775c8..44adae9 100644 --- a/drivers/char/pcmcia/cm4040_cs.c +++ b/drivers/char/pcmcia/cm4040_cs.c @@ -29,7 +29,6 @@ #include <asm/uaccess.h> #include <asm/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/char/pcmcia/ipwireless/main.h b/drivers/char/pcmcia/ipwireless/main.h index 96d0ef3..c207be8 100644 --- a/drivers/char/pcmcia/ipwireless/main.h +++ b/drivers/char/pcmcia/ipwireless/main.h @@ -21,7 +21,6 @@ #include <linux/sched.h> #include <linux/types.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/char/pcmcia/ipwireless/tty.h b/drivers/char/pcmcia/ipwireless/tty.h index 4da6c20..3e163d4 100644 --- a/drivers/char/pcmcia/ipwireless/tty.h +++ b/drivers/char/pcmcia/ipwireless/tty.h @@ -21,7 +21,6 @@ #include <linux/types.h> #include <linux/sched.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/char/pcmcia/synclink_cs.c b/drivers/char/pcmcia/synclink_cs.c index 308903e..522992e 100644 --- a/drivers/char/pcmcia/synclink_cs.c +++ b/drivers/char/pcmcia/synclink_cs.c @@ -70,7 +70,6 @@ #include <linux/workqueue.h> #include <linux/hdlc.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/ide/ide-cs.c b/drivers/ide/ide-cs.c index 0b7815d..27dbab8 100644 --- a/drivers/ide/ide-cs.c +++ b/drivers/ide/ide-cs.c @@ -43,7 +43,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/isdn/hardware/avm/avm_cs.c b/drivers/isdn/hardware/avm/avm_cs.c index f410d0e..e804a01 100644 --- a/drivers/isdn/hardware/avm/avm_cs.c +++ b/drivers/isdn/hardware/avm/avm_cs.c @@ -20,7 +20,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/isdn/hisax/avma1_cs.c b/drivers/isdn/hisax/avma1_cs.c index a80a761..49e141e 100644 --- a/drivers/isdn/hisax/avma1_cs.c +++ b/drivers/isdn/hisax/avma1_cs.c @@ -20,7 +20,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/isdn/hisax/elsa_cs.c b/drivers/isdn/hisax/elsa_cs.c index 218927e..425deea 100644 --- a/drivers/isdn/hisax/elsa_cs.c +++ b/drivers/isdn/hisax/elsa_cs.c @@ -46,7 +46,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/isdn/hisax/sedlbauer_cs.c b/drivers/isdn/hisax/sedlbauer_cs.c index 1f4feaa..5dbad96 100644 --- a/drivers/isdn/hisax/sedlbauer_cs.c +++ b/drivers/isdn/hisax/sedlbauer_cs.c @@ -46,7 +46,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/isdn/hisax/teles_cs.c b/drivers/isdn/hisax/teles_cs.c index 5771955..d3fb1b7 100644 --- a/drivers/isdn/hisax/teles_cs.c +++ b/drivers/isdn/hisax/teles_cs.c @@ -27,7 +27,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/mmc/host/sdricoh_cs.c b/drivers/mmc/host/sdricoh_cs.c index e7507af..7aa65bb 100644 --- a/drivers/mmc/host/sdricoh_cs.c +++ b/drivers/mmc/host/sdricoh_cs.c @@ -30,7 +30,6 @@ #include <linux/ioport.h> #include <linux/scatterlist.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/mtd/maps/pcmciamtd.c b/drivers/mtd/maps/pcmciamtd.c index e699e6a..7948816 100644 --- a/drivers/mtd/maps/pcmciamtd.c +++ b/drivers/mtd/maps/pcmciamtd.c @@ -16,7 +16,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/net/pcmcia/3c574_cs.c b/drivers/net/pcmcia/3c574_cs.c index 10ee106..e249b89 100644 --- a/drivers/net/pcmcia/3c574_cs.c +++ b/drivers/net/pcmcia/3c574_cs.c @@ -87,7 +87,6 @@ earlier 3Com products. #include <linux/bitops.h> #include <linux/mii.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/pcmcia/3c589_cs.c b/drivers/net/pcmcia/3c589_cs.c index ce63c37..b0772df 100644 --- a/drivers/net/pcmcia/3c589_cs.c +++ b/drivers/net/pcmcia/3c589_cs.c @@ -41,7 +41,6 @@ #include <linux/bitops.h> #include <linux/jiffies.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/pcmcia/axnet_cs.c b/drivers/net/pcmcia/axnet_cs.c index 33525bf..467fd4b 100644 --- a/drivers/net/pcmcia/axnet_cs.c +++ b/drivers/net/pcmcia/axnet_cs.c @@ -39,7 +39,6 @@ #include <linux/mii.h> #include "../8390.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/net/pcmcia/com20020_cs.c b/drivers/net/pcmcia/com20020_cs.c index 5643f94..99957af 100644 --- a/drivers/net/pcmcia/com20020_cs.c +++ b/drivers/net/pcmcia/com20020_cs.c @@ -43,7 +43,6 @@ #include <linux/arcdevice.h> #include <linux/com20020.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/net/pcmcia/fmvj18x_cs.c b/drivers/net/pcmcia/fmvj18x_cs.c index 7c27c50..95a991b 100644 --- a/drivers/net/pcmcia/fmvj18x_cs.c +++ b/drivers/net/pcmcia/fmvj18x_cs.c @@ -49,7 +49,6 @@ #include <linux/ioport.h> #include <linux/crc32.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/net/pcmcia/ibmtr_cs.c b/drivers/net/pcmcia/ibmtr_cs.c index 67ee985..c36dcd1 100644 --- a/drivers/net/pcmcia/ibmtr_cs.c +++ b/drivers/net/pcmcia/ibmtr_cs.c @@ -57,7 +57,6 @@ #include <linux/trdevice.h> #include <linux/ibmtr.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/net/pcmcia/nmclan_cs.c b/drivers/net/pcmcia/nmclan_cs.c index 9b63dec..c0eacfa 100644 --- a/drivers/net/pcmcia/nmclan_cs.c +++ b/drivers/net/pcmcia/nmclan_cs.c @@ -146,7 +146,6 @@ Include Files #include <linux/ioport.h> #include <linux/bitops.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cisreg.h> #include <pcmcia/cistpl.h> diff --git a/drivers/net/pcmcia/pcnet_cs.c b/drivers/net/pcmcia/pcnet_cs.c index bfdef72..db6dbda 100644 --- a/drivers/net/pcmcia/pcnet_cs.c +++ b/drivers/net/pcmcia/pcnet_cs.c @@ -42,7 +42,6 @@ #include <linux/mii.h> #include "../8390.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> @@ -113,8 +112,6 @@ static int setup_dma_config(struct pcmcia_device *link, int start_pg, static void pcnet_detach(struct pcmcia_device *p_dev); -static dev_info_t dev_info = "pcnet_cs"; - /*====================================================================*/ typedef struct hw_info_t { @@ -956,7 +953,7 @@ static int pcnet_open(struct net_device *dev) set_misc_reg(dev); outb_p(0xFF, nic_base + EN0_ISR); /* Clear bogus intr. */ - ret = request_irq(dev->irq, ei_irq_wrapper, IRQF_SHARED, dev_info, dev); + ret = request_irq(dev->irq, ei_irq_wrapper, IRQF_SHARED, dev->name, dev); if (ret) return ret; diff --git a/drivers/net/pcmcia/smc91c92_cs.c b/drivers/net/pcmcia/smc91c92_cs.c index 307cd17..88f503a 100644 --- a/drivers/net/pcmcia/smc91c92_cs.c +++ b/drivers/net/pcmcia/smc91c92_cs.c @@ -44,7 +44,6 @@ #include <linux/jiffies.h> #include <linux/firmware.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/pcmcia/xirc2ps_cs.c b/drivers/net/pcmcia/xirc2ps_cs.c index b6c3644..a7662f0 100644 --- a/drivers/net/pcmcia/xirc2ps_cs.c +++ b/drivers/net/pcmcia/xirc2ps_cs.c @@ -82,7 +82,6 @@ #include <linux/bitops.h> #include <linux/mii.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/airo_cs.c b/drivers/net/wireless/airo_cs.c index 33bdc6a..9389ba0 100644 --- a/drivers/net/wireless/airo_cs.c +++ b/drivers/net/wireless/airo_cs.c @@ -32,7 +32,6 @@ #include <linux/timer.h> #include <linux/netdevice.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/atmel_cs.c b/drivers/net/wireless/atmel_cs.c index c2746fc..91ee74a 100644 --- a/drivers/net/wireless/atmel_cs.c +++ b/drivers/net/wireless/atmel_cs.c @@ -42,7 +42,6 @@ #include <linux/moduleparam.h> #include <linux/device.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/b43/pcmcia.c b/drivers/net/wireless/b43/pcmcia.c index 0e99b63..f71bc78 100644 --- a/drivers/net/wireless/b43/pcmcia.c +++ b/drivers/net/wireless/b43/pcmcia.c @@ -26,7 +26,6 @@ #include <linux/ssb/ssb.h> #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/net/wireless/hostap/hostap_cs.c b/drivers/net/wireless/hostap/hostap_cs.c index 29b31a6..2f4b6d4 100644 --- a/drivers/net/wireless/hostap/hostap_cs.c +++ b/drivers/net/wireless/hostap/hostap_cs.c @@ -12,7 +12,6 @@ #include <linux/wireless.h> #include <net/iw_handler.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> @@ -23,7 +22,7 @@ #include "hostap_wlan.h" -static dev_info_t dev_info = "hostap_cs"; +static char *dev_info = "hostap_cs"; MODULE_AUTHOR("Jouni Malinen"); MODULE_DESCRIPTION("Support for Intersil Prism2-based 802.11 wireless LAN " diff --git a/drivers/net/wireless/libertas/if_cs.c b/drivers/net/wireless/libertas/if_cs.c index 08e4e39..1d3a7e0 100644 --- a/drivers/net/wireless/libertas/if_cs.c +++ b/drivers/net/wireless/libertas/if_cs.c @@ -28,7 +28,6 @@ #include <linux/firmware.h> #include <linux/netdevice.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/net/wireless/orinoco/orinoco_cs.c b/drivers/net/wireless/orinoco/orinoco_cs.c index b16d5db..41ca4f1 100644 --- a/drivers/net/wireless/orinoco/orinoco_cs.c +++ b/drivers/net/wireless/orinoco/orinoco_cs.c @@ -17,7 +17,6 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/orinoco/spectrum_cs.c b/drivers/net/wireless/orinoco/spectrum_cs.c index b51a9ad..cad30e4 100644 --- a/drivers/net/wireless/orinoco/spectrum_cs.c +++ b/drivers/net/wireless/orinoco/spectrum_cs.c @@ -25,7 +25,6 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/ray_cs.c b/drivers/net/wireless/ray_cs.c index abff893..165beb6 100644 --- a/drivers/net/wireless/ray_cs.c +++ b/drivers/net/wireless/ray_cs.c @@ -46,7 +46,6 @@ #include <linux/ethtool.h> #include <linux/ieee80211.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/net/wireless/wl3501_cs.c b/drivers/net/wireless/wl3501_cs.c index 376c6b9..35f431b 100644 --- a/drivers/net/wireless/wl3501_cs.c +++ b/drivers/net/wireless/wl3501_cs.c @@ -48,7 +48,6 @@ #include <net/iw_handler.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> @@ -89,13 +88,6 @@ static int wl3501_config(struct pcmcia_device *link); static void wl3501_release(struct pcmcia_device *link); -/* - * The dev_info variable is the "key" that is used to match up this - * device driver with appropriate cards, through the card configuration - * database. - */ -static dev_info_t wl3501_dev_info = "wl3501_cs"; - static const struct { int reg_domain; int min, max, deflt; @@ -1421,7 +1413,7 @@ static struct iw_statistics *wl3501_get_wireless_stats(struct net_device *dev) static void wl3501_get_drvinfo(struct net_device *dev, struct ethtool_drvinfo *info) { - strlcpy(info->driver, wl3501_dev_info, sizeof(info->driver)); + strlcpy(info->driver, "wl3501_cs", sizeof(info->driver)); } static const struct ethtool_ops ops = { diff --git a/drivers/parport/parport_cs.c b/drivers/parport/parport_cs.c index fd8cfe9..ee56fd6 100644 --- a/drivers/parport/parport_cs.c +++ b/drivers/parport/parport_cs.c @@ -48,7 +48,6 @@ #include <linux/parport.h> #include <linux/parport_pc.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/pcmcia/au1000_generic.h b/drivers/pcmcia/au1000_generic.h index a324d32..67530ce 100644 --- a/drivers/pcmcia/au1000_generic.h +++ b/drivers/pcmcia/au1000_generic.h @@ -23,7 +23,6 @@ /* include the world */ -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/au1000_pb1x00.c b/drivers/pcmcia/au1000_pb1x00.c index 5a979cb..807f2d7 100644 --- a/drivers/pcmcia/au1000_pb1x00.c +++ b/drivers/pcmcia/au1000_pb1x00.c @@ -31,11 +31,9 @@ #include <linux/proc_fs.h> #include <linux/types.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> -#include <pcmcia/bus_ops.h> #include <asm/io.h> #include <asm/irq.h> diff --git a/drivers/pcmcia/cistpl.c b/drivers/pcmcia/cistpl.c index 8844bc3..ba4a5ac 100644 --- a/drivers/pcmcia/cistpl.c +++ b/drivers/pcmcia/cistpl.c @@ -27,7 +27,6 @@ #include <asm/byteorder.h> #include <asm/unaligned.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cisreg.h> diff --git a/drivers/pcmcia/cs.c b/drivers/pcmcia/cs.c index efa30b8..2ec8ac9 100644 --- a/drivers/pcmcia/cs.c +++ b/drivers/pcmcia/cs.c @@ -32,7 +32,6 @@ #include <asm/system.h> #include <asm/irq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/db1xxx_ss.c b/drivers/pcmcia/db1xxx_ss.c index 0f4cc3f..27575e6 100644 --- a/drivers/pcmcia/db1xxx_ss.c +++ b/drivers/pcmcia/db1xxx_ss.c @@ -29,7 +29,6 @@ #include <linux/slab.h> #include <linux/spinlock.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <asm/mach-au1x00/au1000.h> diff --git a/drivers/pcmcia/ds.c b/drivers/pcmcia/ds.c index 73aeaea..56933eb 100644 --- a/drivers/pcmcia/ds.c +++ b/drivers/pcmcia/ds.c @@ -26,7 +26,6 @@ #include <linux/dma-mapping.h> #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/pcmcia/i82092.c b/drivers/pcmcia/i82092.c index 3003bb3..05d0879 100644 --- a/drivers/pcmcia/i82092.c +++ b/drivers/pcmcia/i82092.c @@ -15,7 +15,6 @@ #include <linux/interrupt.h> #include <linux/device.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/i82365.c b/drivers/pcmcia/i82365.c index 9e2a156..61746bd 100644 --- a/drivers/pcmcia/i82365.c +++ b/drivers/pcmcia/i82365.c @@ -50,7 +50,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/m32r_cfc.c b/drivers/pcmcia/m32r_cfc.c index 7e16ed8..24de499 100644 --- a/drivers/pcmcia/m32r_cfc.c +++ b/drivers/pcmcia/m32r_cfc.c @@ -26,7 +26,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/m32r_pcc.c b/drivers/pcmcia/m32r_pcc.c index 6c5c3f9..8e47238 100644 --- a/drivers/pcmcia/m32r_pcc.c +++ b/drivers/pcmcia/m32r_pcc.c @@ -27,7 +27,6 @@ #include <asm/system.h> #include <asm/addrspace.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/m8xx_pcmcia.c b/drivers/pcmcia/m8xx_pcmcia.c index 25e5e30..f2f90a7 100644 --- a/drivers/pcmcia/m8xx_pcmcia.c +++ b/drivers/pcmcia/m8xx_pcmcia.c @@ -59,7 +59,6 @@ #include <asm/irq.h> #include <asm/fs_pd.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> diff --git a/drivers/pcmcia/pcmcia_cis.c b/drivers/pcmcia/pcmcia_cis.c index 4a65eaf..0ac54da 100644 --- a/drivers/pcmcia/pcmcia_cis.c +++ b/drivers/pcmcia/pcmcia_cis.c @@ -19,7 +19,6 @@ #include <linux/kernel.h> #include <linux/netdevice.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cisreg.h> #include <pcmcia/cistpl.h> #include <pcmcia/ss.h> diff --git a/drivers/pcmcia/pcmcia_resource.c b/drivers/pcmcia/pcmcia_resource.c index a4cd9ad..2394de4 100644 --- a/drivers/pcmcia/pcmcia_resource.c +++ b/drivers/pcmcia/pcmcia_resource.c @@ -25,7 +25,6 @@ #include <asm/irq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/pd6729.c b/drivers/pcmcia/pd6729.c index b61a136..b8a869a 100644 --- a/drivers/pcmcia/pd6729.c +++ b/drivers/pcmcia/pd6729.c @@ -17,7 +17,6 @@ #include <linux/device.h> #include <linux/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/pcmcia/pxa2xx_base.c b/drivers/pcmcia/pxa2xx_base.c index df4532e..66c0225 100644 --- a/drivers/pcmcia/pxa2xx_base.c +++ b/drivers/pcmcia/pxa2xx_base.c @@ -32,7 +32,6 @@ #include <mach/pxa2xx-regs.h> #include <asm/mach-types.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/rsrc_iodyn.c b/drivers/pcmcia/rsrc_iodyn.c index 6ed7bf1..3b1dce2 100644 --- a/drivers/pcmcia/rsrc_iodyn.c +++ b/drivers/pcmcia/rsrc_iodyn.c @@ -16,7 +16,6 @@ #include <linux/module.h> #include <linux/kernel.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/rsrc_mgr.c b/drivers/pcmcia/rsrc_mgr.c index b12ecf7..b433a79 100644 --- a/drivers/pcmcia/rsrc_mgr.c +++ b/drivers/pcmcia/rsrc_mgr.c @@ -16,7 +16,6 @@ #include <linux/module.h> #include <linux/kernel.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/rsrc_nonstatic.c b/drivers/pcmcia/rsrc_nonstatic.c index 13245a2..0cca08f 100644 --- a/drivers/pcmcia/rsrc_nonstatic.c +++ b/drivers/pcmcia/rsrc_nonstatic.c @@ -28,7 +28,6 @@ #include <asm/irq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/sa1100_generic.c b/drivers/pcmcia/sa1100_generic.c index edbd8c4..e098514 100644 --- a/drivers/pcmcia/sa1100_generic.c +++ b/drivers/pcmcia/sa1100_generic.c @@ -35,7 +35,6 @@ #include <linux/slab.h> #include <linux/platform_device.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> diff --git a/drivers/pcmcia/soc_common.h b/drivers/pcmcia/soc_common.h index e40824c..3fba3a6 100644 --- a/drivers/pcmcia/soc_common.h +++ b/drivers/pcmcia/soc_common.h @@ -11,7 +11,6 @@ /* include the world */ #include <linux/cpufreq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/socket_sysfs.c b/drivers/pcmcia/socket_sysfs.c index 80e36bc..cb0d3ac 100644 --- a/drivers/pcmcia/socket_sysfs.c +++ b/drivers/pcmcia/socket_sysfs.c @@ -26,7 +26,6 @@ #include <asm/system.h> #include <asm/irq.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/tcic.c b/drivers/pcmcia/tcic.c index 56004a1..be0d841 100644 --- a/drivers/pcmcia/tcic.c +++ b/drivers/pcmcia/tcic.c @@ -49,7 +49,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include "tcic.h" diff --git a/drivers/pcmcia/xxs1500_ss.c b/drivers/pcmcia/xxs1500_ss.c index 201ccfa..fa88c36 100644 --- a/drivers/pcmcia/xxs1500_ss.c +++ b/drivers/pcmcia/xxs1500_ss.c @@ -17,7 +17,6 @@ #include <linux/slab.h> #include <linux/spinlock.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/ss.h> #include <pcmcia/cistpl.h> diff --git a/drivers/pcmcia/yenta_socket.c b/drivers/pcmcia/yenta_socket.c index f1d4137..414d9a6 100644 --- a/drivers/pcmcia/yenta_socket.c +++ b/drivers/pcmcia/yenta_socket.c @@ -19,7 +19,6 @@ #include <linux/io.h> #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/ss.h> #include <pcmcia/cs.h> diff --git a/drivers/scsi/pcmcia/aha152x_stub.c b/drivers/scsi/pcmcia/aha152x_stub.c index 9d70aef..b07b53e 100644 --- a/drivers/scsi/pcmcia/aha152x_stub.c +++ b/drivers/scsi/pcmcia/aha152x_stub.c @@ -49,7 +49,6 @@ #include <scsi/scsi_host.h> #include "aha152x.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/scsi/pcmcia/fdomain_stub.c b/drivers/scsi/pcmcia/fdomain_stub.c index 21b1411..ee04897 100644 --- a/drivers/scsi/pcmcia/fdomain_stub.c +++ b/drivers/scsi/pcmcia/fdomain_stub.c @@ -46,7 +46,6 @@ #include <scsi/scsi_host.h> #include "fdomain.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/scsi/pcmcia/nsp_cs.c b/drivers/scsi/pcmcia/nsp_cs.c index 0f0e112..d414207 100644 --- a/drivers/scsi/pcmcia/nsp_cs.c +++ b/drivers/scsi/pcmcia/nsp_cs.c @@ -47,7 +47,6 @@ #include <scsi/scsi.h> #include <scsi/scsi_ioctl.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/scsi/pcmcia/qlogic_stub.c b/drivers/scsi/pcmcia/qlogic_stub.c index f0fc6ba..c1cf7f4 100644 --- a/drivers/scsi/pcmcia/qlogic_stub.c +++ b/drivers/scsi/pcmcia/qlogic_stub.c @@ -48,7 +48,6 @@ #include <scsi/scsi_host.h> #include "../qlogicfas408.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/scsi/pcmcia/sym53c500_cs.c b/drivers/scsi/pcmcia/sym53c500_cs.c index a511641..bd79e45 100644 --- a/drivers/scsi/pcmcia/sym53c500_cs.c +++ b/drivers/scsi/pcmcia/sym53c500_cs.c @@ -71,7 +71,6 @@ #include <scsi/scsi.h> #include <scsi/scsi_host.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/serial/serial_cs.c b/drivers/serial/serial_cs.c index ab17c08..2b99c7b 100644 --- a/drivers/serial/serial_cs.c +++ b/drivers/serial/serial_cs.c @@ -45,7 +45,6 @@ #include <asm/io.h> #include <asm/system.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/ssb/main.c b/drivers/ssb/main.c index 51275aa..06f04b4 100644 --- a/drivers/ssb/main.c +++ b/drivers/ssb/main.c @@ -20,7 +20,6 @@ #include <linux/mmc/sdio_func.h> #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/ssb/pcmcia.c b/drivers/ssb/pcmcia.c index e72f404..2152030 100644 --- a/drivers/ssb/pcmcia.c +++ b/drivers/ssb/pcmcia.c @@ -13,7 +13,6 @@ #include <linux/io.h> #include <linux/etherdevice.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ciscode.h> diff --git a/drivers/ssb/scan.c b/drivers/ssb/scan.c index 0d6c028..9738cad 100644 --- a/drivers/ssb/scan.c +++ b/drivers/ssb/scan.c @@ -17,7 +17,6 @@ #include <linux/pci.h> #include <linux/io.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/staging/comedi/drivers/cb_das16_cs.c b/drivers/staging/comedi/drivers/cb_das16_cs.c index cfeb11f..acef29c 100644 --- a/drivers/staging/comedi/drivers/cb_das16_cs.c +++ b/drivers/staging/comedi/drivers/cb_das16_cs.c @@ -37,7 +37,6 @@ Status: experimental #include <linux/delay.h> #include <linux/pci.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/staging/comedi/drivers/das08_cs.c b/drivers/staging/comedi/drivers/das08_cs.c index 8761a6d..3eddb7c 100644 --- a/drivers/staging/comedi/drivers/das08_cs.c +++ b/drivers/staging/comedi/drivers/das08_cs.c @@ -48,7 +48,6 @@ Command support does not exist, but could be added for this board. #include "das08.h" /* pcmcia includes */ -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/staging/comedi/drivers/ni_daq_700.c b/drivers/staging/comedi/drivers/ni_daq_700.c index 6ec77bf..f3c4d2f 100644 --- a/drivers/staging/comedi/drivers/ni_daq_700.c +++ b/drivers/staging/comedi/drivers/ni_daq_700.c @@ -47,7 +47,6 @@ IRQ is assigned but not used. #include <linux/ioport.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/comedi/drivers/ni_daq_dio24.c b/drivers/staging/comedi/drivers/ni_daq_dio24.c index e4865b1..f0c4367 100644 --- a/drivers/staging/comedi/drivers/ni_daq_dio24.c +++ b/drivers/staging/comedi/drivers/ni_daq_dio24.c @@ -48,7 +48,6 @@ the PCMCIA interface. #include "8255.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/comedi/drivers/ni_labpc_cs.c b/drivers/staging/comedi/drivers/ni_labpc_cs.c index 163245e..1ee78f8 100644 --- a/drivers/staging/comedi/drivers/ni_labpc_cs.c +++ b/drivers/staging/comedi/drivers/ni_labpc_cs.c @@ -71,7 +71,6 @@ NI manuals: #include "comedi_fc.h" #include "ni_labpc.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/comedi/drivers/ni_mio_cs.c b/drivers/staging/comedi/drivers/ni_mio_cs.c index 3a46f0c..0bce220 100644 --- a/drivers/staging/comedi/drivers/ni_mio_cs.c +++ b/drivers/staging/comedi/drivers/ni_mio_cs.c @@ -48,7 +48,6 @@ See the notes in the ni_atmio.o driver. #include "ni_stc.h" #include "8255.h" -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/staging/comedi/drivers/quatech_daqp_cs.c b/drivers/staging/comedi/drivers/quatech_daqp_cs.c index a91db6c..67c0fa6 100644 --- a/drivers/staging/comedi/drivers/quatech_daqp_cs.c +++ b/drivers/staging/comedi/drivers/quatech_daqp_cs.c @@ -50,7 +50,6 @@ Devices: [Quatech] DAQP-208 (daqp), DAQP-308 #include "../comedidev.h" #include <linux/semaphore.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/wlags49_h2/wl_cs.c b/drivers/staging/wlags49_h2/wl_cs.c index 10abd40..afe4575 100644 --- a/drivers/staging/wlags49_h2/wl_cs.c +++ b/drivers/staging/wlags49_h2/wl_cs.c @@ -83,7 +83,6 @@ #include <linux/if_arp.h> #include <linux/ioport.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/staging/wlags49_h2/wl_internal.h b/drivers/staging/wlags49_h2/wl_internal.h index d9a0ad0..02f0a20 100644 --- a/drivers/staging/wlags49_h2/wl_internal.h +++ b/drivers/staging/wlags49_h2/wl_internal.h @@ -69,7 +69,6 @@ ******************************************************************************/ #include <linux/version.h> #ifdef BUS_PCMCIA -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> diff --git a/drivers/telephony/ixj_pcmcia.c b/drivers/telephony/ixj_pcmcia.c index 99cb224..f6c7e6f 100644 --- a/drivers/telephony/ixj_pcmcia.c +++ b/drivers/telephony/ixj_pcmcia.c @@ -8,7 +8,6 @@ #include <linux/errno.h> /* error codes */ #include <linux/slab.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/drivers/usb/host/sl811_cs.c b/drivers/usb/host/sl811_cs.c index 58cb73c..acb7e25 100644 --- a/drivers/usb/host/sl811_cs.c +++ b/drivers/usb/host/sl811_cs.c @@ -20,7 +20,6 @@ #include <linux/ioport.h> #include <linux/platform_device.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/cisreg.h> @@ -43,8 +42,6 @@ MODULE_LICENSE("GPL"); /* VARIABLES */ /*====================================================================*/ -static const char driver_name[DEV_NAME_LEN] = "sl811_cs"; - typedef struct local_info_t { struct pcmcia_device *p_dev; } local_info_t; @@ -246,7 +243,7 @@ MODULE_DEVICE_TABLE(pcmcia, sl811_ids); static struct pcmcia_driver sl811_cs_driver = { .owner = THIS_MODULE, .drv = { - .name = (char *)driver_name, + .name = "sl811_cs", }, .probe = sl811_cs_probe, .remove = sl811_cs_detach, diff --git a/include/pcmcia/cistpl.h b/include/pcmcia/cistpl.h index cfdd5af..1c5088c 100644 --- a/include/pcmcia/cistpl.h +++ b/include/pcmcia/cistpl.h @@ -15,6 +15,8 @@ #ifndef _LINUX_CISTPL_H #define _LINUX_CISTPL_H +typedef unsigned char cisdata_t; + #define CISTPL_NULL 0x00 #define CISTPL_DEVICE 0x01 #define CISTPL_LONGLINK_CB 0x02 diff --git a/include/pcmcia/cs.h b/include/pcmcia/cs.h index c943c96..c78d9b1 100644 --- a/include/pcmcia/cs.h +++ b/include/pcmcia/cs.h @@ -43,14 +43,6 @@ typedef struct conf_reg_t { #define CV_COPY_VALUE 0x08 #define CV_EXT_STATUS 0x10 -/* For GetFirst/NextClient */ -typedef struct client_req_t { - socket_t Socket; - u_int Attributes; -} client_req_t; - -#define CLIENT_THIS_SOCKET 0x01 - /* ModifyConfiguration */ typedef struct modconf_t { u_int Attributes; @@ -133,7 +125,7 @@ typedef struct io_req_t { /* For GetMemPage, MapMemPage */ typedef struct memreq_t { u_int CardOffset; - page_t Page; + u_short Page; } memreq_t; /* For ModifyWindow */ diff --git a/include/pcmcia/cs_types.h b/include/pcmcia/cs_types.h deleted file mode 100644 index f5e3b83..0000000 --- a/include/pcmcia/cs_types.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * cs_types.h - * - * 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. - * - * The initial developer of the original code is David A. Hinds - * <dahinds(a)users.sourceforge.net>. Portions created by David A. Hinds - * are Copyright (C) 1999 David A. Hinds. All Rights Reserved. - * - * (C) 1999 David A. Hinds - */ - -#ifndef _LINUX_CS_TYPES_H -#define _LINUX_CS_TYPES_H - -#ifdef __KERNEL__ -#include <linux/types.h> -#else -#include <sys/types.h> -#endif - -typedef u_short socket_t; -typedef u_int event_t; -typedef u_char cisdata_t; -typedef u_short page_t; - -typedef unsigned long window_handle_t; - -struct region_t; -typedef struct region_t *memory_handle_t; - -#ifndef DEV_NAME_LEN -#define DEV_NAME_LEN 32 -#endif - -typedef char dev_info_t[DEV_NAME_LEN]; - -#endif /* _LINUX_CS_TYPES_H */ diff --git a/include/pcmcia/ds.h b/include/pcmcia/ds.h index 7d7721e..e614aa0 100644 --- a/include/pcmcia/ds.h +++ b/include/pcmcia/ds.h @@ -20,7 +20,6 @@ #include <linux/mod_devicetable.h> #endif -#include <pcmcia/cs_types.h> #include <pcmcia/device_id.h> #ifdef __KERNEL__ @@ -37,6 +36,8 @@ struct pcmcia_device; struct config_t; struct net_device; +typedef unsigned long window_handle_t; + /* dynamic device IDs for PCMCIA device drivers. See * Documentation/pcmcia/driver.txt for details. */ diff --git a/include/pcmcia/ss.h b/include/pcmcia/ss.h index aeac271..626b63c 100644 --- a/include/pcmcia/ss.h +++ b/include/pcmcia/ss.h @@ -19,7 +19,6 @@ #include <linux/sched.h> /* task_struct, completion */ #include <linux/mutex.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #ifdef CONFIG_CARDBUS #include <linux/pci.h> diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h index a0a7ec6..5cc3e45 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.h +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h @@ -24,7 +24,6 @@ #include <sound/pcm.h> #include <asm/io.h> #include <linux/interrupt.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/sound/pcmcia/vx/vxpocket.h b/sound/pcmcia/vx/vxpocket.h index ea4df16..d911066 100644 --- a/sound/pcmcia/vx/vxpocket.h +++ b/sound/pcmcia/vx/vxpocket.h @@ -23,7 +23,6 @@ #include <sound/vx_core.h> -#include <pcmcia/cs_types.h> #include <pcmcia/cs.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> -- 1.7.0.4

4 3

[alsa-devel] [PATCH][v2] powerpc: rename immap_86xx.h to fsl_guts.h, and add 85xx support
by Timur Tabi 23 Jul '10

23 Jul '10

The immap_86xx.h header file only defines one data structure: the "global utilities" register set found on Freescale PowerPC SOCs. Rename this file to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS" register set on 85xx chips. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Liam, This patch is a powerpc-patch, but it's much simpler if you apply it to multi-component, since only the SSI audio drivers are affected. .../include/asm/{immap_86xx.h => fsl_guts.h} | 111 +++++++++++++------ sound/soc/fsl/fsl_ssi.c | 2 - sound/soc/fsl/mpc8610_hpcd.c | 10 +- 3 files changed, 81 insertions(+), 42 deletions(-) rename arch/powerpc/include/asm/{immap_86xx.h => fsl_guts.h} (66%) diff --git a/arch/powerpc/include/asm/immap_86xx.h b/arch/powerpc/include/asm/fsl_guts.h similarity index 66% rename from arch/powerpc/include/asm/immap_86xx.h rename to arch/powerpc/include/asm/fsl_guts.h index 0f165e5..bebd124 100644 --- a/arch/powerpc/include/asm/immap_86xx.h +++ b/arch/powerpc/include/asm/fsl_guts.h @@ -1,5 +1,5 @@ /** - * MPC86xx Internal Memory Map + * Freecale 85xx and 86xx Global Utilties register set * * Authors: Jeff Brown * Timur Tabi <timur(a)freescale.com> @@ -10,73 +10,112 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * This header file defines structures for various 86xx SOC devices that are - * used by multiple source files. */ -#ifndef __ASM_POWERPC_IMMAP_86XX_H__ -#define __ASM_POWERPC_IMMAP_86XX_H__ +#ifndef __ASM_POWERPC_FSL_GUTS_H__ +#define __ASM_POWERPC_FSL_GUTS_H__ #ifdef __KERNEL__ -/* Global Utility Registers */ -struct ccsr_guts { +/* + * These #ifdefs are safe because it's not possible to build a kernel that + * runs on e500 and e600 cores. + */ + +#if !defined(CONFIG_PPC_85xx) && !defined(CONFIG_PPC_86xx) +#error Only 85xx and 86xx SOCs are supported +#endif + +/** + * Global Utility Registers. + * + * Not all registers defined in this structure are available on all chips, so + * you are expected to know whether a given register actually exists on your + * chip before you access it. + * + * Also, some registers are similar on different chips but have slightly + * different names. In these cases, one name is chosen to avoid extraneous + * #ifdefs. + */ +#ifdef CONFIG_PPC_85xx +struct ccsr_guts_85xx { +#else +struct ccsr_guts_86xx { +#endif __be32 porpllsr; /* 0x.0000 - POR PLL Ratio Status Register */ __be32 porbmsr; /* 0x.0004 - POR Boot Mode Status Register */ __be32 porimpscr; /* 0x.0008 - POR I/O Impedance Status and Control Register */ __be32 pordevsr; /* 0x.000c - POR I/O Device Status Register */ __be32 pordbgmsr; /* 0x.0010 - POR Debug Mode Status Register */ - u8 res1[0x20 - 0x14]; + __be32 pordevsr2; /* 0x.0014 - POR device status register 2 */ + u8 res018[0x20 - 0x18]; __be32 porcir; /* 0x.0020 - POR Configuration Information Register */ - u8 res2[0x30 - 0x24]; + u8 res024[0x30 - 0x24]; __be32 gpiocr; /* 0x.0030 - GPIO Control Register */ - u8 res3[0x40 - 0x34]; + u8 res034[0x40 - 0x34]; __be32 gpoutdr; /* 0x.0040 - General-Purpose Output Data Register */ - u8 res4[0x50 - 0x44]; + u8 res044[0x50 - 0x44]; __be32 gpindr; /* 0x.0050 - General-Purpose Input Data Register */ - u8 res5[0x60 - 0x54]; + u8 res054[0x60 - 0x54]; __be32 pmuxcr; /* 0x.0060 - Alternate Function Signal Multiplex Control */ - u8 res6[0x70 - 0x64]; + __be32 pmuxcr2; /* 0x.0064 - Alternate function signal multiplex control 2 */ + __be32 dmuxcr; /* 0x.0068 - DMA Mux Control Register */ + u8 res06c[0x70 - 0x6c]; __be32 devdisr; /* 0x.0070 - Device Disable Control */ __be32 devdisr2; /* 0x.0074 - Device Disable Control 2 */ - u8 res7[0x80 - 0x78]; + u8 res078[0x7c - 0x78]; + __be32 pmjcr; /* 0x.007c - 4 Power Management Jog Control Register */ __be32 powmgtcsr; /* 0x.0080 - Power Management Status and Control Register */ - u8 res8[0x90 - 0x84]; + __be32 pmrccr; /* 0x.0084 - Power Management Reset Counter Configuration Register */ + __be32 pmpdccr; /* 0x.0088 - Power Management Power Down Counter Configuration Register */ + __be32 pmcdr; /* 0x.008c - 4Power management clock disable register */ __be32 mcpsumr; /* 0x.0090 - Machine Check Summary Register */ __be32 rstrscr; /* 0x.0094 - Reset Request Status and Control Register */ - u8 res9[0xA0 - 0x98]; + __be32 ectrstcr; /* 0x.0098 - Exception reset control register */ + __be32 autorstsr; /* 0x.009c - Automatic reset status register */ __be32 pvr; /* 0x.00a0 - Processor Version Register */ __be32 svr; /* 0x.00a4 - System Version Register */ - u8 res10[0xB0 - 0xA8]; + u8 res0a8[0xb0 - 0xa8]; __be32 rstcr; /* 0x.00b0 - Reset Control Register */ - u8 res11[0xC0 - 0xB4]; + u8 res0b4[0xc0 - 0xb4]; +#ifdef CONFIG_PPC_85xx + __be32 iovselsr; /* 0x.00c0 - I/O voltage select status register */ +#else __be32 elbcvselcr; /* 0x.00c0 - eLBC Voltage Select Ctrl Reg */ - u8 res12[0x800 - 0xC4]; +#endif + u8 res0c4[0x224 - 0xc4]; + __be32 iodelay1; /* 0x.0224 - IO delay control register 1 */ + __be32 iodelay2; /* 0x.0228 - IO delay control register 2 */ + u8 res22c[0x800 - 0x22c]; __be32 clkdvdr; /* 0x.0800 - Clock Divide Register */ - u8 res13[0x900 - 0x804]; + u8 res804[0x900 - 0x804]; __be32 ircr; /* 0x.0900 - Infrared Control Register */ - u8 res14[0x908 - 0x904]; + u8 res904[0x908 - 0x904]; __be32 dmacr; /* 0x.0908 - DMA Control Register */ - u8 res15[0x914 - 0x90C]; + u8 res90c[0x914 - 0x90c]; __be32 elbccr; /* 0x.0914 - eLBC Control Register */ - u8 res16[0xB20 - 0x918]; + u8 res918[0xb20 - 0x918]; __be32 ddr1clkdr; /* 0x.0b20 - DDR1 Clock Disable Register */ __be32 ddr2clkdr; /* 0x.0b24 - DDR2 Clock Disable Register */ __be32 ddrclkdr; /* 0x.0b28 - DDR Clock Disable Register */ - u8 res17[0xE00 - 0xB2C]; + u8 resb2c[0xe00 - 0xb2c]; __be32 clkocr; /* 0x.0e00 - Clock Out Select Register */ - u8 res18[0xE10 - 0xE04]; + u8 rese04[0xe10 - 0xe04]; __be32 ddrdllcr; /* 0x.0e10 - DDR DLL Control Register */ - u8 res19[0xE20 - 0xE14]; + u8 rese14[0xe20 - 0xe14]; __be32 lbcdllcr; /* 0x.0e20 - LBC DLL Control Register */ - u8 res20[0xF04 - 0xE24]; + __be32 cpfor; /* 0x.0e24 - L2 charge pump fuse override register */ + u8 rese28[0xf04 - 0xe28]; __be32 srds1cr0; /* 0x.0f04 - SerDes1 Control Register 0 */ __be32 srds1cr1; /* 0x.0f08 - SerDes1 Control Register 0 */ - u8 res21[0xF40 - 0xF0C]; - __be32 srds2cr0; /* 0x.0f40 - SerDes1 Control Register 0 */ - __be32 srds2cr1; /* 0x.0f44 - SerDes1 Control Register 0 */ + u8 resf0c[0xf2c - 0xf0c]; + __be32 itcr; /* 0x.0f2c - Internal transaction control register */ + u8 resf30[0xf40 - 0xf30]; + __be32 srds2cr0; /* 0x.0f40 - SerDes2 Control Register 0 */ + __be32 srds2cr1; /* 0x.0f44 - SerDes2 Control Register 0 */ } __attribute__ ((packed)); +#ifdef CONFIG_PPC_86xx + #define CCSR_GUTS_DMACR_DEV_SSI 0 /* DMA controller/channel set to SSI */ #define CCSR_GUTS_DMACR_DEV_IR 1 /* DMA controller/channel set to IR */ @@ -93,7 +132,7 @@ struct ccsr_guts { * ch: The channel on the DMA controller (0, 1, 2, or 3) * device: The device to set as the source (CCSR_GUTS_DMACR_DEV_xx) */ -static inline void guts_set_dmacr(struct ccsr_guts __iomem *guts, +static inline void guts_set_dmacr(struct ccsr_guts_86xx __iomem *guts, unsigned int co, unsigned int ch, unsigned int device) { unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); @@ -129,7 +168,7 @@ static inline void guts_set_dmacr(struct ccsr_guts __iomem *guts, * ch: The channel on the DMA controller (0, 1, 2, or 3) * value: the new value for the bit (0 or 1) */ -static inline void guts_set_pmuxcr_dma(struct ccsr_guts __iomem *guts, +static inline void guts_set_pmuxcr_dma(struct ccsr_guts_86xx __iomem *guts, unsigned int co, unsigned int ch, unsigned int value) { if ((ch == 0) || (ch == 3)) { @@ -152,5 +191,7 @@ static inline void guts_set_pmuxcr_dma(struct ccsr_guts __iomem *guts, #define CCSR_GUTS_CLKDVDR_SSICLK_MASK 0x000000FF #define CCSR_GUTS_CLKDVDR_SSICLK(x) ((x) & CCSR_GUTS_CLKDVDR_SSICLK_MASK) -#endif /* __ASM_POWERPC_IMMAP_86XX_H__ */ -#endif /* __KERNEL__ */ +#endif + +#endif +#endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 31e8e5c..48fc0bd 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -24,8 +24,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <asm/immap_86xx.h> - #include "fsl_ssi.h" /** diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 81ab639..69aeaeb 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -14,7 +14,7 @@ #include <linux/interrupt.h> #include <linux/of_device.h> #include <sound/soc.h> -#include <asm/immap_86xx.h> +#include <asm/fsl_guts.h> #include "../codecs/cs4270.h" #include "fsl_dma.h" @@ -54,9 +54,9 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; + struct ccsr_guts_86xx __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; @@ -139,9 +139,9 @@ static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; + struct ccsr_guts_86xx __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; -- 1.7.0.1

4 3

[alsa-devel] [PATCH] ASoC: multi-component - SAMSUNG - Fix compile error
by Seungwhan Youn 23 Jul '10

23 Jul '10

This patch fixes compile error on smdk_wm9713. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/s3c24xx/smdk_wm9713.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 849aac1..90108a7 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -48,7 +48,7 @@ static struct snd_soc_dai_link smdk_dai = { .stream_name = "AC97 PCM", .platform_name = "s3c24xx-pcm-audio", .cpu_dai_name = "s3c-ac97-dai", - .codec_dai_name = "wm9713-hifi" + .codec_dai_name = "wm9713-hifi", .codec_name = "wm9713-codec", }; -- 1.6.2.5

3 2

[alsa-devel] [PATCH] driver core: remove CONFIG_SYSFS_DEPRECATED
by Greg Kroah-Hartman 23 Jul '10

23 Jul '10

This is no longer needed by any userspace tools, so it's safe to remove. Cc: Jens Axboe <axboe(a)kernel.dk> Cc: Stephen Hemminger <shemminger(a)vyatta.com> Cc: "Eric W. Biederman" <ebiederm(a)xmission.com> Cc: Kay Sievers <kay.sievers(a)vrfy.org> Cc: Alan Stern <stern(a)rowland.harvard.edu> Cc: "James E.J. Bottomley" <James.Bottomley(a)suse.de> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: Alexey Kuznetsov <kuznet(a)ms2.inr.ac.ru> Cc: Randy Dunlap <randy.dunlap(a)oracle.com> Cc: Tejun Heo <tj(a)kernel.org> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Ingo Molnar <mingo(a)elte.hu> Cc: Peter Zijlstra <a.p.zijlstra(a)chello.nl> Cc: David Howells <dhowells(a)redhat.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)suse.de> --- block/genhd.c | 5 - drivers/base/bus.c | 22 ------ drivers/base/class.c | 25 ------- drivers/base/core.c | 164 ----------------------------------------------- drivers/scsi/hosts.c | 2 drivers/scsi/scsi_scan.c | 2 fs/partitions/check.c | 6 - include/sound/core.h | 6 - init/Kconfig | 36 ---------- sound/core/init.c | 11 --- 10 files changed, 7 insertions(+), 272 deletions(-) --- a/block/genhd.c +++ b/block/genhd.c @@ -22,9 +22,7 @@ #include "blk.h" static DEFINE_MUTEX(block_class_lock); -#ifndef CONFIG_SYSFS_DEPRECATED struct kobject *block_depr; -#endif /* for extended dynamic devt allocation, currently only one major is used */ #define MAX_EXT_DEVT (1 << MINORBITS) @@ -803,10 +801,9 @@ static int __init genhd_device_init(void register_blkdev(BLOCK_EXT_MAJOR, "blkext"); -#ifndef CONFIG_SYSFS_DEPRECATED /* create top-level block dir */ block_depr = kobject_create_and_add("block", NULL); -#endif + return 0; } --- a/drivers/base/bus.c +++ b/drivers/base/bus.c @@ -440,22 +440,6 @@ static void device_remove_attrs(struct b } } -#ifdef CONFIG_SYSFS_DEPRECATED -static int make_deprecated_bus_links(struct device *dev) -{ - return sysfs_create_link(&dev->kobj, - &dev->bus->p->subsys.kobj, "bus"); -} - -static void remove_deprecated_bus_links(struct device *dev) -{ - sysfs_remove_link(&dev->kobj, "bus"); -} -#else -static inline int make_deprecated_bus_links(struct device *dev) { return 0; } -static inline void remove_deprecated_bus_links(struct device *dev) { } -#endif - /** * bus_add_device - add device to bus * @dev: device being added @@ -482,15 +466,10 @@ int bus_add_device(struct device *dev) &dev->bus->p->subsys.kobj, "subsystem"); if (error) goto out_subsys; - error = make_deprecated_bus_links(dev); - if (error) - goto out_deprecated; klist_add_tail(&dev->p->knode_bus, &bus->p->klist_devices); } return 0; -out_deprecated: - sysfs_remove_link(&dev->kobj, "subsystem"); out_subsys: sysfs_remove_link(&bus->p->devices_kset->kobj, dev_name(dev)); out_id: @@ -530,7 +509,6 @@ void bus_remove_device(struct device *de { if (dev->bus) { sysfs_remove_link(&dev->kobj, "subsystem"); - remove_deprecated_bus_links(dev); sysfs_remove_link(&dev->bus->p->devices_kset->kobj, dev_name(dev)); device_remove_attrs(dev->bus, dev); --- a/drivers/base/class.c +++ b/drivers/base/class.c @@ -184,13 +184,7 @@ int __class_register(struct class *cls, if (!cls->dev_kobj) cls->dev_kobj = sysfs_dev_char_kobj; -#if defined(CONFIG_SYSFS_DEPRECATED) && defined(CONFIG_BLOCK) - /* let the block class directory show up in the root of sysfs */ - if (cls != &block_class) - cp->class_subsys.kobj.kset = class_kset; -#else cp->class_subsys.kobj.kset = class_kset; -#endif cp->class_subsys.kobj.ktype = &class_ktype; cp->class = cls; cls->p = cp; @@ -276,25 +270,6 @@ void class_destroy(struct class *cls) class_unregister(cls); } -#ifdef CONFIG_SYSFS_DEPRECATED -char *make_class_name(const char *name, struct kobject *kobj) -{ - char *class_name; - int size; - - size = strlen(name) + strlen(kobject_name(kobj)) + 2; - - class_name = kmalloc(size, GFP_KERNEL); - if (!class_name) - return NULL; - - strcpy(class_name, name); - strcat(class_name, ":"); - strcat(class_name, kobject_name(kobj)); - return class_name; -} -#endif - /** * class_dev_iter_init - initialize class device iterator * @iter: class iterator to initialize --- a/drivers/base/core.c +++ b/drivers/base/core.c @@ -203,37 +203,6 @@ static int dev_uevent(struct kset *kset, if (dev->driver) add_uevent_var(env, "DRIVER=%s", dev->driver->name); -#ifdef CONFIG_SYSFS_DEPRECATED - if (dev->class) { - struct device *parent = dev->parent; - - /* find first bus device in parent chain */ - while (parent && !parent->bus) - parent = parent->parent; - if (parent && parent->bus) { - const char *path; - - path = kobject_get_path(&parent->kobj, GFP_KERNEL); - if (path) { - add_uevent_var(env, "PHYSDEVPATH=%s", path); - kfree(path); - } - - add_uevent_var(env, "PHYSDEVBUS=%s", parent->bus->name); - - if (parent->driver) - add_uevent_var(env, "PHYSDEVDRIVER=%s", - parent->driver->name); - } - } else if (dev->bus) { - add_uevent_var(env, "PHYSDEVBUS=%s", dev->bus->name); - - if (dev->driver) - add_uevent_var(env, "PHYSDEVDRIVER=%s", - dev->driver->name); - } -#endif - /* have the bus specific function add its stuff */ if (dev->bus && dev->bus->uevent) { retval = dev->bus->uevent(dev, env); @@ -578,24 +547,6 @@ void device_initialize(struct device *de set_dev_node(dev, -1); } -#ifdef CONFIG_SYSFS_DEPRECATED -static struct kobject *get_device_parent(struct device *dev, - struct device *parent) -{ - /* class devices without a parent live in /sys/class/<classname>/ */ - if (dev->class && (!parent || parent->class != dev->class)) - return &dev->class->p->class_subsys.kobj; - /* all other devices keep their parent */ - else if (parent) - return &parent->kobj; - - return NULL; -} - -static inline void cleanup_device_parent(struct device *dev) {} -static inline void cleanup_glue_dir(struct device *dev, - struct kobject *glue_dir) {} -#else static struct kobject *virtual_device_parent(struct device *dev) { static struct kobject *virtual_dir = NULL; @@ -719,7 +670,6 @@ static void cleanup_device_parent(struct { cleanup_glue_dir(dev, dev->kobj.parent); } -#endif static void setup_parent(struct device *dev, struct device *parent) { @@ -742,53 +692,6 @@ static int device_add_class_symlinks(str if (error) goto out; -#ifdef CONFIG_SYSFS_DEPRECATED - /* stacked class devices need a symlink in the class directory */ - if (dev->kobj.parent != &dev->class->p->class_subsys.kobj && - device_is_not_partition(dev)) { - error = sysfs_create_link(&dev->class->p->class_subsys.kobj, - &dev->kobj, dev_name(dev)); - if (error) - goto out_subsys; - } - - if (dev->parent && device_is_not_partition(dev)) { - struct device *parent = dev->parent; - char *class_name; - - /* - * stacked class devices have the 'device' link - * pointing to the bus device instead of the parent - */ - while (parent->class && !parent->bus && parent->parent) - parent = parent->parent; - - error = sysfs_create_link(&dev->kobj, - &parent->kobj, - "device"); - if (error) - goto out_busid; - - class_name = make_class_name(dev->class->name, - &dev->kobj); - if (class_name) - error = sysfs_create_link(&dev->parent->kobj, - &dev->kobj, class_name); - kfree(class_name); - if (error) - goto out_device; - } - return 0; - -out_device: - if (dev->parent && device_is_not_partition(dev)) - sysfs_remove_link(&dev->kobj, "device"); -out_busid: - if (dev->kobj.parent != &dev->class->p->class_subsys.kobj && - device_is_not_partition(dev)) - sysfs_delete_link(&dev->class->p->class_subsys.kobj, &dev->kobj, - dev_name(dev)); -#else /* link in the class directory pointing to the device */ error = sysfs_create_link(&dev->class->p->class_subsys.kobj, &dev->kobj, dev_name(dev)); @@ -805,7 +708,6 @@ out_busid: out_busid: sysfs_delete_link(&dev->class->p->class_subsys.kobj, &dev->kobj, dev_name(dev)); -#endif out_subsys: sysfs_remove_link(&dev->kobj, "subsystem"); @@ -818,29 +720,10 @@ static void device_remove_class_symlinks if (!dev->class) return; -#ifdef CONFIG_SYSFS_DEPRECATED - if (dev->parent && device_is_not_partition(dev)) { - char *class_name; - - class_name = make_class_name(dev->class->name, &dev->kobj); - if (class_name) { - sysfs_remove_link(&dev->parent->kobj, class_name); - kfree(class_name); - } - sysfs_remove_link(&dev->kobj, "device"); - } - - if (dev->kobj.parent != &dev->class->p->class_subsys.kobj && - device_is_not_partition(dev)) - sysfs_delete_link(&dev->class->p->class_subsys.kobj, &dev->kobj, - dev_name(dev)); -#else if (dev->parent && device_is_not_partition(dev)) sysfs_remove_link(&dev->kobj, "device"); sysfs_delete_link(&dev->class->p->class_subsys.kobj, &dev->kobj, dev_name(dev)); -#endif - sysfs_remove_link(&dev->kobj, "subsystem"); } @@ -1613,41 +1496,23 @@ int device_rename(struct device *dev, ch pr_debug("device: '%s': %s: renaming to '%s'\n", dev_name(dev), __func__, new_name); -#ifdef CONFIG_SYSFS_DEPRECATED - if ((dev->class) && (dev->parent)) - old_class_name = make_class_name(dev->class->name, &dev->kobj); -#endif - old_device_name = kstrdup(dev_name(dev), GFP_KERNEL); if (!old_device_name) { error = -ENOMEM; goto out; } -#ifndef CONFIG_SYSFS_DEPRECATED if (dev->class) { error = sysfs_rename_link(&dev->class->p->class_subsys.kobj, &dev->kobj, old_device_name, new_name); if (error) goto out; } -#endif + error = kobject_rename(&dev->kobj, new_name); if (error) goto out; -#ifdef CONFIG_SYSFS_DEPRECATED - if (old_class_name) { - new_class_name = make_class_name(dev->class->name, &dev->kobj); - if (new_class_name) { - error = sysfs_rename_link(&dev->parent->kobj, - &dev->kobj, - old_class_name, - new_class_name); - } - } -#endif - out: put_device(dev); @@ -1664,40 +1529,13 @@ static int device_move_class_links(struc struct device *new_parent) { int error = 0; -#ifdef CONFIG_SYSFS_DEPRECATED - char *class_name; - class_name = make_class_name(dev->class->name, &dev->kobj); - if (!class_name) { - error = -ENOMEM; - goto out; - } - if (old_parent) { - sysfs_remove_link(&dev->kobj, "device"); - sysfs_remove_link(&old_parent->kobj, class_name); - } - if (new_parent) { - error = sysfs_create_link(&dev->kobj, &new_parent->kobj, - "device"); - if (error) - goto out; - error = sysfs_create_link(&new_parent->kobj, &dev->kobj, - class_name); - if (error) - sysfs_remove_link(&dev->kobj, "device"); - } else - error = 0; -out: - kfree(class_name); - return error; -#else if (old_parent) sysfs_remove_link(&dev->kobj, "device"); if (new_parent) error = sysfs_create_link(&dev->kobj, &new_parent->kobj, "device"); return error; -#endif } /** --- a/drivers/scsi/hosts.c +++ b/drivers/scsi/hosts.c @@ -406,9 +406,7 @@ struct Scsi_Host *scsi_host_alloc(struct device_initialize(&shost->shost_gendev); dev_set_name(&shost->shost_gendev, "host%d", shost->host_no); -#ifndef CONFIG_SYSFS_DEPRECATED shost->shost_gendev.bus = &scsi_bus_type; -#endif shost->shost_gendev.type = &scsi_host_type; device_initialize(&shost->shost_dev); --- a/drivers/scsi/scsi_scan.c +++ b/drivers/scsi/scsi_scan.c @@ -417,9 +417,7 @@ static struct scsi_target *scsi_alloc_ta starget->reap_ref = 1; dev->parent = get_device(parent); dev_set_name(dev, "target%d:%d:%d", shost->host_no, channel, id); -#ifndef CONFIG_SYSFS_DEPRECATED dev->bus = &scsi_bus_type; -#endif dev->type = &scsi_target_type; starget->id = id; starget->channel = channel; --- a/fs/partitions/check.c +++ b/fs/partitions/check.c @@ -500,14 +500,14 @@ void register_disk(struct gendisk *disk) if (device_add(ddev)) return; -#ifndef CONFIG_SYSFS_DEPRECATED + err = sysfs_create_link(block_depr, &ddev->kobj, kobject_name(&ddev->kobj)); if (err) { device_del(ddev); return; } -#endif + disk->part0.holder_dir = kobject_create_and_add("holders", &ddev->kobj); disk->slave_dir = kobject_create_and_add("slaves", &ddev->kobj); @@ -724,8 +724,6 @@ void del_gendisk(struct gendisk *disk) kobject_put(disk->part0.holder_dir); kobject_put(disk->slave_dir); disk->driverfs_dev = NULL; -#ifndef CONFIG_SYSFS_DEPRECATED sysfs_remove_link(block_depr, dev_name(disk_to_dev(disk))); -#endif device_del(disk_to_dev(disk)); } --- a/include/sound/core.h +++ b/include/sound/core.h @@ -133,9 +133,7 @@ struct snd_card { int free_on_last_close; /* free in context of file_release */ wait_queue_head_t shutdown_sleep; struct device *dev; /* device assigned to this card */ -#ifndef CONFIG_SYSFS_DEPRECATED struct device *card_dev; /* cardX object for sysfs */ -#endif #ifdef CONFIG_PM unsigned int power_state; /* power state */ @@ -196,11 +194,7 @@ struct snd_minor { /* return a device pointer linked to each sound device as a parent */ static inline struct device *snd_card_get_device_link(struct snd_card *card) { -#ifdef CONFIG_SYSFS_DEPRECATED - return card ? card->dev : NULL; -#else return card ? card->card_dev : NULL; -#endif } /* sound.c */ --- a/init/Kconfig +++ b/init/Kconfig @@ -651,42 +651,6 @@ endif # CGROUPS config MM_OWNER bool -config SYSFS_DEPRECATED - bool - -config SYSFS_DEPRECATED_V2 - bool "enable deprecated sysfs features to support old userspace tools" - depends on SYSFS - default n - select SYSFS_DEPRECATED - help - This option switches the layout of sysfs to the deprecated - version. Do not use it on recent distributions. - - The current sysfs layout features a unified device tree at - /sys/devices/, which is able to express a hierarchy between - class devices. If the deprecated option is set to Y, the - unified device tree is split into a bus device tree at - /sys/devices/ and several individual class device trees at - /sys/class/. The class and bus devices will be connected by - "<subsystem>:<name>" and the "device" links. The "block" - class devices, will not show up in /sys/class/block/. Some - subsystems will suppress the creation of some devices which - depend on the unified device tree. - - This option is not a pure compatibility option that can - be safely enabled on newer distributions. It will change the - layout of sysfs to the non-extensible deprecated version, - and disable some features, which can not be exported without - confusing older userspace tools. Since 2007/2008 all major - distributions do not enable this option, and ship no tools which - depend on the deprecated layout or this option. - - If you are using a new kernel on an older distribution, or use - older userspace tools, you might need to say Y here. Do not say Y, - if the original kernel, that came with your distribution, has - this option set to N. - config RELAY bool "Kernel->user space relay support (formerly relayfs)" help --- a/sound/core/init.c +++ b/sound/core/init.c @@ -395,12 +395,10 @@ int snd_card_disconnect(struct snd_card snd_printk(KERN_ERR "not all devices for card %i can be disconnected\n", card->number); snd_info_card_disconnect(card); -#ifndef CONFIG_SYSFS_DEPRECATED if (card->card_dev) { device_unregister(card->card_dev); card->card_dev = NULL; } -#endif #ifdef CONFIG_PM wake_up(&card->power_sleep); #endif @@ -573,7 +571,6 @@ void snd_card_set_id(struct snd_card *ca } EXPORT_SYMBOL(snd_card_set_id); -#ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) @@ -630,7 +627,6 @@ card_number_show_attr(struct device *dev static struct device_attribute card_number_attrs = __ATTR(number, S_IRUGO, card_number_show_attr, NULL); -#endif /* CONFIG_SYSFS_DEPRECATED */ /** * snd_card_register - register the soundcard @@ -649,7 +645,7 @@ int snd_card_register(struct snd_card *c if (snd_BUG_ON(!card)) return -EINVAL; -#ifndef CONFIG_SYSFS_DEPRECATED + if (!card->card_dev) { card->card_dev = device_create(sound_class, card->dev, MKDEV(0, 0), card, @@ -657,7 +653,7 @@ int snd_card_register(struct snd_card *c if (IS_ERR(card->card_dev)) card->card_dev = NULL; } -#endif + if ((err = snd_device_register_all(card)) < 0) return err; mutex_lock(&snd_card_mutex); @@ -674,7 +670,6 @@ int snd_card_register(struct snd_card *c if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER); #endif -#ifndef CONFIG_SYSFS_DEPRECATED if (card->card_dev) { err = device_create_file(card->card_dev, &card_id_attrs); if (err < 0) @@ -683,7 +678,7 @@ int snd_card_register(struct snd_card *c if (err < 0) return err; } -#endif + return 0; }

9 20

[alsa-devel] Which project to choose?
by Chris 23 Jul '10

23 Jul '10

I want to submit a bug report on the below. This issue was noted in Mandriva 2010 and is still present in 2010.1 pulseaudio[24553]: alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -161920 bytes (-917 ms). localhost pulseaudio[24553]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_ens1371'. Please report this issue to the ALSA developers. localhost pulseaudio[24553]: alsa-util.c: snd_pcm_dump(): localhost pulseaudio[24553]: alsa-util.c: Hardware PCM card 0 'Ensoniq AudioPCI' device 0 subdevice 0 localhost pulseaudio[24553]: alsa-util.c: Its setup is: localhost pulseaudio[24553]: alsa-util.c: stream : PLAYBACK localhost pulseaudio[24553]: alsa-util.c: access : MMAP_INTERLEAVED localhost pulseaudio[24553]: alsa-util.c: format : S16_LE localhost pulseaudio[24553]: alsa-util.c: subformat : STD localhost pulseaudio[24553]: alsa-util.c: channels : 2 localhost pulseaudio[24553]: alsa-util.c: rate : 44100 localhost pulseaudio[24553]: alsa-util.c: exact rate : 44101 (1445100000/32768) localhost pulseaudio[24553]: alsa-util.c: msbits : 16 localhost pulseaudio[24553]: alsa-util.c: buffer_size : 4408 localhost pulseaudio[24553]: alsa-util.c: period_size : 1102 localhost pulseaudio[24553]: alsa-util.c: period_time : 24988 localhost pulseaudio[24553]: alsa-util.c: tstamp_mode : ENABLE localhost pulseaudio[24553]: alsa-util.c: period_step : 1 localhost pulseaudio[24553]: alsa-util.c: avail_min : 1102 localhost pulseaudio[24553]: alsa-util.c: period_event : 1 localhost pulseaudio[24553]: alsa-util.c: start_threshold : -1 localhost pulseaudio[24553]: alsa-util.c: stop_threshold : 1155530752 localhost pulseaudio[24553]: alsa-util.c: silence_threshold: 0 localhost pulseaudio[24553]: alsa-util.c: silence_size : 0 localhost pulseaudio[24553]: alsa-util.c: boundary : 1155530752 localhost pulseaudio[24553]: alsa-util.c: appl_ptr : 735801008 localhost pulseaudio[24553]: alsa-util.c: hw_ptr : 735846184 localhost pulseaudio[24553]: ratelimit.c: 280 events suppressed localhost pulseaudio[24553]: asyncq.c: q overrun, queuing locally Because of the constant disk access when this is happening my system becomes almost unusable. -- Chris KeyID 0xE372A7DA98E6705C

4 15

[alsa-devel] open sound device from asound.conf
by Peter Lukac 23 Jul '10

23 Jul '10

hello all maybe trivial question... I want open sound device from /etc/asound.conf with API snd_pcm_open() but unsuccessfull. As name i'm used: "plughw:0,0" "plughw:0,1" "plughw:0,2" "plughw:1,0" "plughw:1,1" "plughw:1,2" ... i'm try alternative name with "hw:X,Y" "hw:0,0" "hw:0,1" ... but in this method i'm open only HW device. I'm get follow error: ALSA lib pcm_hw.c:1429:(_snd_pcm_hw_open) Invalid value for card how can i open my "virtual" device in asound.conf here is asound.conf: pcm.!default { type plug slave.pcm "output" } pcm.output { type dmix ipc_key 1024 slave { pcm "hw:0,0" channels 2 rate 44100 format S16_LE } } pcm.dsp0 { type plug slave.pcm "output" } ...exists some universal method how can i open all HW device and "virtual" devices? thanks

3 4

[alsa-devel] [PATCH v2] ASoC: da7210: Add HeadPhone Playback Volume control
by Kuninori Morimoto 23 Jul '10

23 Jul '10

HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 o remove default HeadPhone Playback Volume sound/soc/codecs/da7210.c | 31 +++++++++++++++++++++++++++---- 1 files changed, 27 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a83aa18..3e42d1e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -23,6 +23,7 @@ #include <sound/pcm_params.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "da7210.h" @@ -136,6 +137,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -218,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -647,6 +667,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: -- 1.7.0.4

3 2

[alsa-devel] [PATCH] ASoC: multi-component - Fix Core AC97 set device name
by Seungwhan Youn 23 Jul '10

23 Jul '10

AC97 driver needs to make kobject with id which from codec. But in multi-component, this id was removed, so this patch fix this same as 'sound/pci/ac97/ac97_codec.c'. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 23dad36..d074b2b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -310,7 +310,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", - codec->id, 0, codec->name); + codec->ac97->bus->card->number, 0, codec->name); err = device_register(&codec->ac97->dev); if (err < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); -- 1.6.2.5

2 1

Re: [alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Viliam Kubis 22 Jul '10

22 Jul '10

I have tried the patch on clean (unpatched) patch_via.c. dmesg prints out ***PATCH 08/07/10 as expected, but the surround does not work. Surround and cen/lfe controls are also not created for alsamixer without other patches. What does work is specifying -Dsurround40 for speaker-test but there is no sound from rear speakers. If you want, I can post the whole patched patch_via.c somewhere or we can work on new patches starting with this one. On Thu, 08 Jul 2010 08:33:34 +0200, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> Viliam Kubis wrote: >>> Hello, for quite some time the "Smart 5.1" function in alsa is not >>> working >>> properly when motherboard has only 3 audio jacks. >>> >>> speaker-test -c 4 -t wav -Dsurround40 >>> >>> Channels count (4) not available for playbacks: Invalid argument >> >> Please try the patch below. >> >> (This function uses 0x12 and 0x13 as the NIDs for the center/LFE and >> side DACs. Shouldn't these be 0x24 and 0x25, as per the datasheet?) >> >> --- a/sound/pci/hda/patch_via.c >> +++ b/sound/pci/hda/patch_via.c >> @@ -2265,6 +2265,13 @@ static int vt1708_auto_fill_dac_nids(str >> } >> } >> + /* for Smart 5.1, line/mic inputs double as output pins */ >> + if (cfg->line_outs == 1) { >> + spec->multiout.num_dacs = 3; >> + spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; >> + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x12; >> + } >> + >> return 0; >> } >> > -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

3 3

[alsa-devel] [PATCH] powerpc: rename immap_86xx.h to fsl_guts.h, and add 85xx support
by Timur Tabi 22 Jul '10

22 Jul '10

The immap_86xx.h header file only defines one data structure: the "global utilities" register set found on Freescale PowerPC SOCs. Rename this file to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS" register set on 85xx chips. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Liam, This patch is a powerpc-patch, but it's much simpler if you apply it to multi-component, since only the SSI audio drivers are affected. Kumar Gala will ack this patch. .../include/asm/{immap_86xx.h => fsl_guts.h} | 106 +++++++++++++------- sound/soc/fsl/fsl_ssi.c | 2 - sound/soc/fsl/mpc8610_hpcd.c | 10 +- 3 files changed, 76 insertions(+), 42 deletions(-) rename arch/powerpc/include/asm/{immap_86xx.h => fsl_guts.h} (67%) diff --git a/arch/powerpc/include/asm/immap_86xx.h b/arch/powerpc/include/asm/fsl_guts.h similarity index 67% rename from arch/powerpc/include/asm/immap_86xx.h rename to arch/powerpc/include/asm/fsl_guts.h index 0f165e5..4827900 100644 --- a/arch/powerpc/include/asm/immap_86xx.h +++ b/arch/powerpc/include/asm/fsl_guts.h @@ -1,5 +1,5 @@ /** - * MPC86xx Internal Memory Map + * Freecale 85xx and 86xx Global Utilties register set * * Authors: Jeff Brown * Timur Tabi <timur(a)freescale.com> @@ -10,73 +10,107 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * This header file defines structures for various 86xx SOC devices that are - * used by multiple source files. */ -#ifndef __ASM_POWERPC_IMMAP_86XX_H__ -#define __ASM_POWERPC_IMMAP_86XX_H__ +#ifndef __ASM_POWERPC_FSL_GUTS_H__ +#define __ASM_POWERPC_FSL_GUTS_H__ #ifdef __KERNEL__ -/* Global Utility Registers */ -struct ccsr_guts { +#if !defined(CONFIG_PPC_85xx) && !defined(CONFIG_PPC_86xx) +#error Only 85xx and 86xx SOCs are supported +#endif + +/** + * Global Utility Registers. + * + * Not all registers defined in this structure are available on all chips, so + * you are expected to know whether a given register actually exists on your + * chip before you access it. + * + * Also, some registers are similar on different chips but have slightly + * different names. In these cases, one name is chosen to avoid extraneous + * #ifdefs. + */ +#ifdef CONFIG_PPC_85xx +struct ccsr_guts_85xx { +#else +struct ccsr_guts_86xx { +#endif __be32 porpllsr; /* 0x.0000 - POR PLL Ratio Status Register */ __be32 porbmsr; /* 0x.0004 - POR Boot Mode Status Register */ __be32 porimpscr; /* 0x.0008 - POR I/O Impedance Status and Control Register */ __be32 pordevsr; /* 0x.000c - POR I/O Device Status Register */ __be32 pordbgmsr; /* 0x.0010 - POR Debug Mode Status Register */ - u8 res1[0x20 - 0x14]; + __be32 pordevsr2; /* 0x.0014 - POR device status register 2 */ + u8 res018[0x20 - 0x18]; __be32 porcir; /* 0x.0020 - POR Configuration Information Register */ - u8 res2[0x30 - 0x24]; + u8 res024[0x30 - 0x24]; __be32 gpiocr; /* 0x.0030 - GPIO Control Register */ - u8 res3[0x40 - 0x34]; + u8 res034[0x40 - 0x34]; __be32 gpoutdr; /* 0x.0040 - General-Purpose Output Data Register */ - u8 res4[0x50 - 0x44]; + u8 res044[0x50 - 0x44]; __be32 gpindr; /* 0x.0050 - General-Purpose Input Data Register */ - u8 res5[0x60 - 0x54]; + u8 res054[0x60 - 0x54]; __be32 pmuxcr; /* 0x.0060 - Alternate Function Signal Multiplex Control */ - u8 res6[0x70 - 0x64]; + __be32 pmuxcr2; /* 0x.0064 - Alternate function signal multiplex control 2 */ + __be32 dmuxcr; /* 0x.0068 - DMA Mux Control Register */ + u8 res06c[0x70 - 0x6c]; __be32 devdisr; /* 0x.0070 - Device Disable Control */ __be32 devdisr2; /* 0x.0074 - Device Disable Control 2 */ - u8 res7[0x80 - 0x78]; + u8 res078[0x7c - 0x78]; + __be32 pmjcr; /* 0x.007c - 4 Power Management Jog Control Register */ __be32 powmgtcsr; /* 0x.0080 - Power Management Status and Control Register */ - u8 res8[0x90 - 0x84]; + __be32 pmrccr; /* 0x.0084 - Power Management Reset Counter Configuration Register */ + __be32 pmpdccr; /* 0x.0088 - Power Management Power Down Counter Configuration Register */ + __be32 pmcdr; /* 0x.008c - 4Power management clock disable register */ __be32 mcpsumr; /* 0x.0090 - Machine Check Summary Register */ __be32 rstrscr; /* 0x.0094 - Reset Request Status and Control Register */ - u8 res9[0xA0 - 0x98]; + __be32 ectrstcr; /* 0x.0098 - Exception reset control register */ + __be32 autorstsr; /* 0x.009c - Automatic reset status register */ __be32 pvr; /* 0x.00a0 - Processor Version Register */ __be32 svr; /* 0x.00a4 - System Version Register */ - u8 res10[0xB0 - 0xA8]; + u8 res0a8[0xb0 - 0xa8]; __be32 rstcr; /* 0x.00b0 - Reset Control Register */ - u8 res11[0xC0 - 0xB4]; + u8 res0b4[0xc0 - 0xb4]; +#ifdef CONFIG_PPC_85xx + __be32 iovselsr; /* 0x.00c0 - I/O voltage select status register */ +#else __be32 elbcvselcr; /* 0x.00c0 - eLBC Voltage Select Ctrl Reg */ - u8 res12[0x800 - 0xC4]; +#endif + u8 res0c4[0x224 - 0xc4]; + __be32 iodelay1; /* 0x.0224 - IO delay control register 1 */ + __be32 iodelay2; /* 0x.0228 - IO delay control register 2 */ + u8 res22c[0x800 - 0x22c]; __be32 clkdvdr; /* 0x.0800 - Clock Divide Register */ - u8 res13[0x900 - 0x804]; + u8 res804[0x900 - 0x804]; __be32 ircr; /* 0x.0900 - Infrared Control Register */ - u8 res14[0x908 - 0x904]; + u8 res904[0x908 - 0x904]; __be32 dmacr; /* 0x.0908 - DMA Control Register */ - u8 res15[0x914 - 0x90C]; + u8 res90c[0x914 - 0x90c]; __be32 elbccr; /* 0x.0914 - eLBC Control Register */ - u8 res16[0xB20 - 0x918]; + u8 res918[0xb20 - 0x918]; __be32 ddr1clkdr; /* 0x.0b20 - DDR1 Clock Disable Register */ __be32 ddr2clkdr; /* 0x.0b24 - DDR2 Clock Disable Register */ __be32 ddrclkdr; /* 0x.0b28 - DDR Clock Disable Register */ - u8 res17[0xE00 - 0xB2C]; + u8 resb2c[0xe00 - 0xb2c]; __be32 clkocr; /* 0x.0e00 - Clock Out Select Register */ - u8 res18[0xE10 - 0xE04]; + u8 rese04[0xe10 - 0xe04]; __be32 ddrdllcr; /* 0x.0e10 - DDR DLL Control Register */ - u8 res19[0xE20 - 0xE14]; + u8 rese14[0xe20 - 0xe14]; __be32 lbcdllcr; /* 0x.0e20 - LBC DLL Control Register */ - u8 res20[0xF04 - 0xE24]; + __be32 cpfor; /* 0x.0e24 - L2 charge pump fuse override register */ + u8 rese28[0xf04 - 0xe28]; __be32 srds1cr0; /* 0x.0f04 - SerDes1 Control Register 0 */ __be32 srds1cr1; /* 0x.0f08 - SerDes1 Control Register 0 */ - u8 res21[0xF40 - 0xF0C]; - __be32 srds2cr0; /* 0x.0f40 - SerDes1 Control Register 0 */ - __be32 srds2cr1; /* 0x.0f44 - SerDes1 Control Register 0 */ + u8 resf0c[0xf2c - 0xf0c]; + __be32 itcr; /* 0x.0f2c - Internal transaction control register */ + u8 resf30[0xf40 - 0xf30]; + __be32 srds2cr0; /* 0x.0f40 - SerDes2 Control Register 0 */ + __be32 srds2cr1; /* 0x.0f44 - SerDes2 Control Register 0 */ } __attribute__ ((packed)); +#ifdef CONFIG_PPC_86xx + #define CCSR_GUTS_DMACR_DEV_SSI 0 /* DMA controller/channel set to SSI */ #define CCSR_GUTS_DMACR_DEV_IR 1 /* DMA controller/channel set to IR */ @@ -93,7 +127,7 @@ struct ccsr_guts { * ch: The channel on the DMA controller (0, 1, 2, or 3) * device: The device to set as the source (CCSR_GUTS_DMACR_DEV_xx) */ -static inline void guts_set_dmacr(struct ccsr_guts __iomem *guts, +static inline void guts_set_dmacr(struct ccsr_guts_86xx __iomem *guts, unsigned int co, unsigned int ch, unsigned int device) { unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); @@ -129,7 +163,7 @@ static inline void guts_set_dmacr(struct ccsr_guts __iomem *guts, * ch: The channel on the DMA controller (0, 1, 2, or 3) * value: the new value for the bit (0 or 1) */ -static inline void guts_set_pmuxcr_dma(struct ccsr_guts __iomem *guts, +static inline void guts_set_pmuxcr_dma(struct ccsr_guts_86xx __iomem *guts, unsigned int co, unsigned int ch, unsigned int value) { if ((ch == 0) || (ch == 3)) { @@ -152,5 +186,7 @@ static inline void guts_set_pmuxcr_dma(struct ccsr_guts __iomem *guts, #define CCSR_GUTS_CLKDVDR_SSICLK_MASK 0x000000FF #define CCSR_GUTS_CLKDVDR_SSICLK(x) ((x) & CCSR_GUTS_CLKDVDR_SSICLK_MASK) -#endif /* __ASM_POWERPC_IMMAP_86XX_H__ */ -#endif /* __KERNEL__ */ +#endif + +#endif +#endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 31e8e5c..48fc0bd 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -24,8 +24,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <asm/immap_86xx.h> - #include "fsl_ssi.h" /** diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 81ab639..69aeaeb 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -14,7 +14,7 @@ #include <linux/interrupt.h> #include <linux/of_device.h> #include <sound/soc.h> -#include <asm/immap_86xx.h> +#include <asm/fsl_guts.h> #include "../codecs/cs4270.h" #include "fsl_dma.h" @@ -54,9 +54,9 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; + struct ccsr_guts_86xx __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; @@ -139,9 +139,9 @@ static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; + struct ccsr_guts_86xx __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; -- 1.7.0.1

3 3

[alsa-devel] [PATCH v2 0/1] *** SUBJECT HERE ***
by Matti J. Aaltonen 22 Jul '10

22 Jul '10

Hello. And thank you for all the comments. Comments to comments: >> +#undef DEBUG > > This shouldn't be here. Removed. >> + wl1273->mode = ucontrol->value.integer.value[0]; > > You're not doing any validation of the supplied value here. Validation added. > Don't put your enums in arrays, it just makes things harder to maintain > since you have to reference an index into an array rather than a named > value. Yes using one-element arrays is kind of meaningless, so arrays removed. > My major comment about this driver is that it'd probably make sense to > redo it on top of Liam's multi-component branch, though it shouldn't be > a pressing concern for merge. I didn't do this yet. But could you give me the git URL and then I'll resend? > Wouldn't it be nicer to do this stuff with the ALSA constraint APIs > rather than futzing with the constant data? I principle I of course agree. But it seems - also discussed with the local ALSA specialist - that using static constraints does not work here. HALF_DUPLEX is not one of the SNDRV_PCM_HW_PARAM_'s... or something like that. As a ALSA non-specialist I'd be willing to this leave as it is... >> +static int wl1273_set_dai_fmt(struct snd_soc_dai *codec_dai, >> + unsigned int fmt) >> +{ >> + return 0; >> +} > >Remove unused functions. I tried leaving this out but then the codec stopped working. I guess I should mention that my test environment is 2.6.32. I'm only able to test that the codec compiles under 2.6.35. Anyway I left the function in. >> +enum wl1273_mode wl1273_get_codec_mode(struct snd_soc_codec *codec) >> +{ >> + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); >> + return wl1273->mode; >> +} >> +EXPORT_SYMBOL_GPL(wl1273_get_codec_mode); > > Why is this being exported? This is because the soc_card driver needs to know the codec mode and accessing the internals of the codec struct is ugly. >> +static int wl1273_soc_suspend(struct platform_device *pdev, pm_message_t state) >> +{ >> + return 0; >> +} >> + >> +static int wl1273_soc_resume(struct platform_device *pdev) >> +{ >> + return 0; >> +} > > Remove unused functions. Dropping these caused no problems, so removed... Cheers, Matti A. Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec. sound/soc/codecs/wl1273.c | 586 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 42 ++++ 2 files changed, 628 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h

5 14

[alsa-devel] ASoC: multi-component updates from RFC
by Liam Girdwood 22 Jul '10

22 Jul '10

I've just pushed some updates to multi-component that came out of the RFC discussion. The main update is the use of dev_name (or DAI name for devices with multiple DAIs) to identify ASoC components. This replaces the current method of using a struct codec|dai|platform pointer and ID to define a DAI link. e.g. we can now define a DAI link as follows :- static struct snd_soc_dai_link sdp3430_dai[] = { { .name = "TWL4030 I2S", .stream_name = "TWL4030 Audio", .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }, { .name = "TWL4030 PCM", .stream_name = "TWL4030 Voice", .cpu_dai_name = "omap-mcbsp-dai.2", .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", .init = sdp3430_twl4030_voice_init, .ops = &sdp3430_voice_ops, }, }; This now means we now no longer have to export all the CODEC | DAI | platform structure symbols around and can keep them static. We also manage to delete a few headers as a result of this change. Since we are now using dev_name or DAI name (for single devices with multiple DAIs) I've taken the opportunity to standardise the ASoC component naming. It's currently very ad-hoc and platform specific, so I've now added some useful prefixes to help. e.g. So we now have :- "-codec" prefix for codecs, e.g. "wm8731-codec" "-dai" prefix for CPU DAI's e.g. "pxa-i2s-dai" "-hifi", -"voice", etc for CODEC DAIs e.g. "wm8731-hifi" "-pcm-audio" for platform DMA, e.g. "pxa-pcm-audio" for PXA DMA (although most platform DMA controllers used this anyway). This naming standard will also come in handy for device tree too. The only other change I've made is fixing the twl4030 and DAC33 codec probe and remove (as spotted by Peter). I've also received and applied numerous patches from Jarkko, Timur, Ryan, Chanwoo, Joonyoug, Kyungmin, Jassi and Seungwhan. The multi-component code can be found here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git topic/multi-component As usual, please give this some testing (especially ASoC platform maintainers) and if all is OK we can upstream some time next week (or the week after). Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

2 3

Re: [alsa-devel] Which project to choose?
by Raymond Yau 22 Jul '10

22 Jul '10

2010/7/22 Chris <cpollock(a)embarqmail.com> > On Thu, 2010-07-22 at 11:03 +0800, Raymond Yau wrote: > > > > > > 2010/7/22 Chris <cpollock(a)embarqmail.com> > > > > > > > > > > AFAIK, the most buggy PA client is pavucontrol because > > > > > > 1) it has those vumeters which need to get audio data from > > the > > > playback and capture devices of the sound card , > > > 2) when you move the slider to left ( over 100%) and this > > force PA to > > > provide software gain > > > > > > You should notice that the new gnome-volume-control is > > actually a > > > simplified version of pavucontrol > > > > > > Hmm, I just pulled up PulseAudio Volume Control and see no > > vumeters, I > > can't figure out though why the drive thrashing at various > > times whether > > I have a stream playing or not. > > > > > > Did you click the input device tab page of the gnome volume control ? > > > > The vumeter is "Input level" above the volume control slider ? > > > > You should noticed that the following message appear in the pulseaudio > > log > > > > I: source-output.c : Created output 0 "Peak detect" on alsa_input > > > > I.source-output:c application.id = "org.gnome.volume.control" > > I.source-output:c media.name = "Peak detect" > > I.source-output:c application.name = "GNOME Volume control dialog" > > > > when you using mplayer to play music and open pavucontrol to > > > > pavucontrol sometime abort suddenly when you aborte mplayer but this > > is not 100% reproducible > > > > pavucontrol > > ** ( pavucontrol:3232) : DEBUG: failed to initialize device manager > > extension: No such extension > > ** (pavucontrol:3056) : DEBUG: Ignoring sink-input due to it being > > designated as an event and thuse handled by the event widget > > > > > > The point is after the complete intergration of pulseaudio in to the > > system ( libcanberra, mixer-applet , ... ) > > > > I think "pulseaudio -k; pulseaudio -vvvv" is no longer the correct way > > to debug since there are lots of PA clients running at the desktop > > > > You have to ask the mandriva maintainer how to increase the debug > > level of PA server and have to perform a logout/login to debug the PA > > server > > Thanks Raymond, so I should contact Colin and go from there? I'm not > sure what to even tell him though. I thank you for all your help. > > Chris > There are over 15 PA clients running when you are using gnome desktop running "pulseaudio -k " to shutdown PA server , this kill all the PA clients restart PA server by "pulseaudio -vvvv" , only one or two PA clients (e.g. consolekit, .. ) can connected back to PA server This command "pulseaudio -k;pulseaudio -vvvv" cannot provide a normal environment to debug PA server since most of PA clients are killed

1 0

[alsa-devel] [PATCH] ASoC: multi-component: SAMSUNG: Fix compile error when registering a DAI
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch modify some code which register a DAI with ASoC core according to modified registration API of DAI. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/aquila_wm8994.c | 3 +-- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c index 69c41eb..007ed3a 100644 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -268,8 +268,7 @@ static int __init aquila_init(void) return -ENOMEM; /* register voice DAI here */ - ret = snd_soc_register_dai(&aquila_snd_device->dev, - 0, &voice_dai); + ret = snd_soc_register_dai(&aquila_snd_device->dev, &voice_dai); if (ret) return ret; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index f4fbc0e..b3866d5 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -750,7 +750,7 @@ int s3c_i2sv2_register_dai(struct device *dev, int id, drv->suspend = s3c2412_i2s_suspend; drv->resume = s3c2412_i2s_resume; - return snd_soc_register_dai(dev, id, drv); + return snd_soc_register_dai(dev, drv); } EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); -- 1.6.3.3

4 9

[alsa-devel] multi-component: determine which DAI is active?
by Timur Tabi 22 Jul '10

22 Jul '10

Liam, In order to get fsl_dma.c to work with the WM8776, which has separate DAIs for playback and capture, I've had to do this: if (pcm->streams[0].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); if (ret) { dev_err(card->dev, "can't allocate playback dma buffer\n"); return ret; } } if (pcm->streams[1].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); dev_err(card->dev, "can't allocate capture dma buffer\n"); return ret; } } That is, I need to check if the 'substream' pointer has been initialized. This looks ugly to me. In an earlier version of ASoC, I would have done this: if (dai->playback.channels_min) { ... if (dai->capture.channels_min) { ... But dai->playback and dai->capture no longer exist. How should I handle this?

3 2

[alsa-devel] [PATCH] ASoC: multi-component - Add Goni sound driver
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch add sound support for the Goni board based on S5PV210. The Goni board is based on Samsung SoC(S5PV210) and include WM8994 codec over I2S to support sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Goni board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Goni board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 ++ sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/goni_wm8994.c | 298 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 309 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/goni_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index c09b402..1cdc37b 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -140,3 +140,12 @@ config SND_S5PC110_SOC_AQUILA_WM8994 help Say Y if you want to add support for SoC audio on aquila with the WM8994. + +config SND_S5PV210_SOC_GONI_WM8994 + tristate "SoC I2S Audio support for GONI - WM8994" + depends on SND_S3C24XX_SOC && MACH_GONI + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on goni + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 02dd12c..47ed6d7 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -31,6 +31,7 @@ snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-aquila-wm8994-objs := aquila_wm8994.o +snd-soc-goni-wm8994-objs := goni_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -45,3 +46,4 @@ obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o +obj-$(CONFIG_SND_S5PV210_SOC_GONI_WM8994) += snd-soc-goni-wm8994.o diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c new file mode 100644 index 0000000..20bcaab --- /dev/null +++ b/sound/soc/s3c24xx/goni_wm8994.c @@ -0,0 +1,298 @@ +/* + * goni_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.com> + * + * 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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +static struct snd_soc_card goni; +static struct platform_device *goni_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget goni_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Left Spk", NULL), + SND_SOC_DAPM_SPK("Ext Right Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route goni_dapm_routes[] = { + {"Ext Left Spk", NULL, "SPKOUTLP"}, + {"Ext Left Spk", NULL, "SPKOUTLN"}, + + {"Ext Right Spk", NULL, "SPKOUTRP"}, + {"Ext Right Spk", NULL, "SPKOUTRN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add goni specific widgets */ + snd_soc_dapm_new_controls(codec, goni_dapm_widgets, + ARRAY_SIZE(goni_dapm_widgets)); + + /* set up goni specific audio routes */ + snd_soc_dapm_add_routes(codec, goni_dapm_routes, + ARRAY_SIZE(goni_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&goni, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int goni_hifi_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_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops goni_hifi_ops = { + .hw_params = goni_hifi_hw_params, +}; + +static int goni_voice_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 pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "goni-voice-dai", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops goni_voice_ops = { + .hw_params = goni_voice_hw_params, +}; + +static struct snd_soc_dai_link goni_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_name = "s3c64xx-i2s-v4-dai", + .codec_dai_name = "wm8994-hifi", + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "wm8994-codec", + .init = goni_wm8994_init, + .ops = &goni_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_name = "goni-voice-dai", + .codec_dai_name = "wm8994-voice", + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "wm8994-codec", + .ops = &goni_voice_ops, +}, +}; + +static struct snd_soc_card goni = { + .name = "goni", + .dai_link = goni_dai, + .num_links = ARRAY_SIZE(goni_dai), +}; + +static int __init goni_init(void) +{ + int ret; + + if (!machine_is_goni()) + return -ENODEV; + + goni_snd_device = platform_device_alloc("soc-audio", -1); + if (!goni_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&goni_snd_device->dev, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(goni_snd_device, &goni); + ret = platform_device_add(goni_snd_device); + + if (ret) + platform_device_put(goni_snd_device); + + return ret; +} + +static void __exit goni_exit(void) +{ + platform_device_unregister(goni_snd_device); +} + +module_init(goni_init); +module_exit(goni_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

2 1

[alsa-devel] [PATCH] ASoC: multi-component: SAMSUNG: Fix wrong field name on Aquila board
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch modify the wrong field name on Aquila board. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/aquila_wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c index f6f0e60..69c41eb 100644 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -242,7 +242,7 @@ static struct snd_soc_dai_link aquila_dai[] = { }, { .name = "WM8994 Voice", .stream_name = "Voice", - .cpu_dai_drv = "aquila-voice-dai", + .cpu_dai_name = "aquila-voice-dai", .codec_dai_name = "wm8994-voice", .platform_name = "s3c24xx-pcm-audio", .codec_name = "wm8994-codec", -- 1.6.3.3

2 1

[alsa-devel] problems about converting sample rates when using ALSA-LIB
by longer 22 Jul '10

22 Jul '10

Hi all, I have a problem when using ALSA lib . The problem is : My ALSA driver only support a special sample rate , so ALSA lib must convert sample rates. But My driver don't get any audio data. When tracing the code of ALSA lib (alsa-lib-1.0.23), I found that : It mmap a address from my driver , but writing audio data to another address instead of that mmaped address. I think this is why my driver don't get any audio data . But I don't know why it don't write audio data to that mmaped address. Can you help me ? Any advices would help me a lot. Thank you, longer.zhou

1 0

[alsa-devel] problems about converting sample rate when using ALSA lib
by longer zhou (周龙) 22 Jul '10

22 Jul '10

Hi all, I have a problem when using ALSA lib . The problem is : My ALSA driver only support a special sample rate , so ALSA lib must convert sample rates. But My driver don't get any audio data. When tracing the code of ALSA lib (alsa-driver-1.0.23), I found that : It mmap a address from my driver , but writting audio data to another address instead of that mmaped address. I think this is why my driver don't get any audio data .But I don't know why it don't write audo data to that mmaped address. Can you help me ? Any advices would help me a lot. Thank you, longer.zhou

1 0

[alsa-devel] add support for Codec: Conexant ID 5068 (not supported by ALSA yet)
by ccamacho＠linuxmail.org 22 Jul '10

22 Jul '10

Hi all, The codec (Conexant ID 5068) required by my sound card (Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)) is not present yet in alsa, I would highly appreciate if you guys could add support for it. You guys might find the information collected by the alsa-info.sh script on http://www.alsa-project.org/db/?f=8f8001d873d5c10c293eac1c94f2ede0b73cd865 Thanks a lot in advance for looking into this. Cheers, Cristopher

1 0

[alsa-devel] Small memory leak in snd_pcm_ladspa_init()
by John Lindgren 22 Jul '10

22 Jul '10

Hello, Valgrind shows up a small memory leak in libasound when using a LADSPA plugin. Sorry, I don't have a patch this time. ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,976 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BEFD: snd_pcm_ladspa_init (pcm_ladspa.c:744) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,977 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BF25: snd_pcm_ladspa_init (pcm_ladspa.c:746) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ... ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,980 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BEFD: snd_pcm_ladspa_init (pcm_ladspa.c:744) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== by 0x1B03B2DC: snd_pcm_recover (pcm.c:7246) ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,981 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BF25: snd_pcm_ladspa_init (pcm_ladspa.c:746) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== by 0x1B03B2DC: snd_pcm_recover (pcm.c:7246) ... and so on. -- John

2 2

[alsa-devel] add support for Codec: Conexant ID 5068 (not present in ALSA yet)
by ccamacho＠linuxmail.org 21 Jul '10

21 Jul '10

Hi all, The codec (Conexant ID 5068) required by my sound card (Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)) is not present yet in alsa, I would highly appreciate if you guys could add support for it. You guys might find the information collected by the alsa-info.sh script on http://www.alsa-project.org/db/?f=8f8001d873d5c10c293eac1c94f2ede0b73cd865 Thanks a lot in advance for looking into this. Cheers, Cristopher

1 0

[alsa-devel] Compiler errors in alsa-lib-1.0.22/test/*.c
by John Simpson 21 Jul '10

21 Jul '10

Howdy, I've been trying to use the alsa-lib API, but I haven't been able to get anything to compile. ================= ~/alsa/alsa-lib-1.0.22/test$ for f in *.c; do echo; cc $f 2>&1 | head -n2; done /tmp/ccNjLWLK.o: In function `dump_event_filter': client_event_filter.c:(.text+0x23): undefined reference to `snd_seq_client_info_event_filter_check' /tmp/ccNLDqdU.o: In function `main': control.c:(.text+0x1c): undefined reference to `snd_ctl_card_info_sizeof' /tmp/ccBehb51.o: In function `setparams_stream': latency.c:(.text+0x24): undefined reference to `snd_pcm_hw_params_any' /usr/lib/gcc/x86_64-linux-gnu/4.4.3/../../../../lib/crt1.o: In function `_start': (.text+0x20): undefined reference to `main' midiloop.c: In function ‘main’: midiloop.c:172: warning: format ‘%i’ expects type ‘int’, but argument 2 has type ‘size_t’ namehint.c: In function ‘main’: namehint.c:12: warning: passing argument 3 of ‘snd_device_name_hint’ from incompatible pointer type oldapi.c: In function ‘main’: oldapi.c:40: warning: assignment from incompatible pointer type /tmp/ccmB528p.o: In function `generate_sine': pcm.c:(.text+0x119): undefined reference to `snd_pcm_format_width' /tmp/ccChgWYC.o: In function `main': pcm_min.c:(.text+0x4d): undefined reference to `snd_pcm_open' /tmp/cc9FiONG.o: In function `write_ev': playmidi1.c:(.text+0x1b65): undefined reference to `snd_seq_event_output' /tmp/ccQkntBX.o: In function `main': queue_timer.c:(.text+0x1d0): undefined reference to `snd_seq_open' /tmp/ccmIMCZ4.o: In function `main': rawmidi.c:(.text+0x4e3): undefined reference to `snd_rawmidi_open' /tmp/cc3g8KGc.o: In function `event_decoder_start_timer': seq.c:(.text+0x4e2): undefined reference to `snd_seq_control_queue' seq-decoder.c:6: error: ‘SND_SEQ_EVENT_SYSTEM’ undeclared here (not in a function) seq-decoder.c:6: error: array index in initializer not of integer type seq-sender.c:68: error: expected ‘)’ before ‘*’ token seq-sender.c:100: error: expected ‘)’ before ‘*’ token /tmp/ccxyZfhg.o: In function `show_status': timer.c:(.text+0x1d): undefined reference to `snd_timer_status_sizeof' ================= Did I miss something? TIA -- John Wesley Simpson SwaJime's Cove℠ www.swajime.com

3 11

[alsa-devel] [PATCH 0/3] Removing dead code
by Christian Dietrich 21 Jul '10

21 Jul '10

Hi all! As part of the VAMOS[0] research project at the University of Erlangen we are looking at multiple integrity errors in linux' configuration system. I've been running a check on the sound/ sourcetree for config Items not defined in Kconfig and found 3 such cases. Sourcecode blocks depending on these Items are not reachable from a vanilla kernel -- dead code. I've seen such dead blocks made on purpose e.g. while integrating new features into the kernel but generally they're just useless. Each of the patches in this patchset removes on such dead config Item, I'd be glad if you consider applying them. I've been doing deeper analysis of such issues before and can do so again but I'm not so sure they were fastly usefull. I build the patches against a vanilla kernel in order to try if the kernel compiles with this patches Please keep me informed of this patch getting confirmed / merged so we can keep track of it. Regards Christian Dietrich [0] http://vamos1.informatik.uni-erlangen.de/ Christian Dietrich (3): sound/oss: Remove dead CONFIG_SOFTOSS* sound/soc/atmel: Remove dead CONFIG_AT32_ENHANCED_PORTMUX sound/isa/gus: Remove dead CONFIG_SND_DEBUG_ROM sound/isa/gus/interwave.c | 8 -------- sound/oss/vidc.c | 3 --- sound/soc/atmel/playpaq_wm8510.c | 9 --------- 3 files changed, 0 insertions(+), 20 deletions(-)

3 6

[alsa-devel] [GIT PULL] ASoC updates for 2.6.36
by Liam Girdwood 21 Jul '10

21 Jul '10

The following changes since commit 66b47fdb851924956b6e4696fb43a3496ae2c462: Mark Brown (1): ASoC: Implement WM8994 OPCLK support are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.36 Jorge Eduardo Candelaria (1): ASoC: TWL6040: Correct widget handling for drivers Peter Ujfalusi (3): ASoC: TWL4030: Fix for digital loopback gain range ASoC: TWL4030: DAC power optimization ASoC: TWL4030: Add configurable delay after digimic enable sound/soc/codecs/twl4030.c | 47 ++++++++++++++++++++++++++---------- sound/soc/codecs/twl4030.h | 1 + sound/soc/codecs/twl6040.c | 56 ++++++++++++++++++++++---------------------- 3 files changed, 63 insertions(+), 41 deletions(-)

2 1

[alsa-devel] mmotm 2010-07-19 - more pm_qos woes - audio this time
by Valdis.Kletnieks＠vt.edu 21 Jul '10

21 Jul '10

On Mon, 19 Jul 2010 16:38:09 PDT, akpm(a)linux-foundation.org said: > The mm-of-the-moment snapshot 2010-07-19-16-37 has been uploaded to > > http://userweb.kernel.org/~akpm/mmotm/ I've accumulated about 1,000 of these overnight. Seems to be consistently landing in pairs based on the timestamps. [42875.544006] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42875.544006] Hardware name: Latitude E6500 [42875.544006] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42875.544006] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42875.544006] Call Trace: [42875.544006] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42875.544006] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42875.544006] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42875.544006] [<ffffffff812052c7>] plist_add+0x1b/0xb1 [42875.544006] [<ffffffff81057b2e>] update_target+0x94/0xd0 [42875.544006] [<ffffffff81057d54>] pm_qos_add_request+0x79/0x7b [42875.544006] [<ffffffff8141d199>] snd_pcm_hw_params+0x25a/0x28a [42875.544006] [<ffffffff8141da29>] snd_pcm_common_ioctl1+0x23b/0x980 [42875.544006] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42875.544006] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42875.544006] [<ffffffff810d9178>] ? fd_install+0x36/0xd5 [42875.544006] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42875.544006] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42875.544006] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42875.544006] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42875.544006] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42875.544006] ---[ end trace 23d666d936caf587 ]--- [42875.544006] ------------[ cut here ]------------ [42875.544006] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42875.544006] Hardware name: Latitude E6500 [42875.544006] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42875.544006] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42875.544006] Call Trace: [42875.544006] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42875.544006] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42875.544006] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42875.544006] [<ffffffff81205354>] plist_add+0xa8/0xb1 [42875.544006] [<ffffffff81057b2e>] update_target+0x94/0xd0 [42875.544006] [<ffffffff81057d54>] pm_qos_add_request+0x79/0x7b [42875.544006] [<ffffffff8141d199>] snd_pcm_hw_params+0x25a/0x28a [42875.544006] [<ffffffff8141da29>] snd_pcm_common_ioctl1+0x23b/0x980 [42875.544006] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42875.544006] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42875.544006] [<ffffffff810d9178>] ? fd_install+0x36/0xd5 [42875.544006] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42875.544006] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42875.544006] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42875.544006] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42875.544006] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42875.544006] ---[ end trace 23d666d936caf588 ]--- [42882.427310] ------------[ cut here ]------------ [42882.428016] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42882.428016] Hardware name: Latitude E6500 [42882.428016] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42882.428016] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42882.428016] Call Trace: [42882.428016] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42882.428016] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42882.428016] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42882.428016] [<ffffffff8120537c>] plist_del+0x1f/0x8b [42882.428016] [<ffffffff81057b21>] update_target+0x87/0xd0 [42882.428016] [<ffffffff81057bb6>] pm_qos_remove_request+0x4c/0x5c [42882.428016] [<ffffffff8141dad9>] snd_pcm_common_ioctl1+0x2eb/0x980 [42882.428016] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42882.428016] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42882.428016] [<ffffffff8109589d>] ? trace_preempt_on+0x15/0x28 [42882.428016] [<ffffffff810c727e>] ? unmap_region+0xbd/0xcc [42882.428016] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42882.428016] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42882.428016] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42882.428016] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42882.428016] [<ffffffff810c746a>] ? unlink_file_vma+0x2d/0x4d [42882.428016] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42882.428016] ---[ end trace 23d666d936caf589 ]--- [42882.428016] ------------[ cut here ]------------ [42882.428016] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42882.428016] Hardware name: Latitude E6500 [42882.428016] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42882.428016] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42882.428016] Call Trace: [42882.428016] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42882.428016] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42882.428016] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42882.428016] [<ffffffff812053df>] plist_del+0x82/0x8b [42882.428016] [<ffffffff81057b21>] update_target+0x87/0xd0 [42882.428016] [<ffffffff81057bb6>] pm_qos_remove_request+0x4c/0x5c [42882.428016] [<ffffffff8141dad9>] snd_pcm_common_ioctl1+0x2eb/0x980 [42882.428016] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42882.428016] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42882.428016] [<ffffffff8109589d>] ? trace_preempt_on+0x15/0x28 [42882.428016] [<ffffffff810c727e>] ? unmap_region+0xbd/0xcc [42882.428016] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42882.428016] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42882.428016] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42882.428016] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42882.428016] [<ffffffff810c746a>] ? unlink_file_vma+0x2d/0x4d [42882.428016] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42882.428016] ---[ end trace 23d666d936caf58a ]---

2 1

[alsa-devel] Missing config option for SFFSDR_FPGA
by Christian Dietrich 21 Jul '10

21 Jul '10

Hi all! As part of the VAMOS[0] research project at the University of Erlangen we are looking at multiple integrity errors in linux' configuration system. I've been running a check on the sound/ sourcetree for config Items not defined in Kconfig and found such a case. Sourcecode blocks depending on these Items are not reachable from a vanilla kernel -- dead code. I've seen such dead blocks made on purpose e.g. while integrating new features into the kernel but generally they're just useless. There are two ifdef blocks, which do a check on CONFIG_SFFSDR_FPGA, but this flag isn't defined anywhere. There is an select in sound/soc/davinci[1], but Kconfig doesn't define the symbol, if there isn't a config option for it. So this blocks can never be selected. I don't think this is intended, so i didn't wrote a patch. Please just notice it. Regards Christian Dietrich [0] http://vamos1.informatik.uni-erlangen.de/ [1] sound/soc/davinci/Kconfig: select SFFSDR_FPGA sound/soc/davinci/davinci-sffsdr.c:#ifdef CONFIG_SFFSDR_FPGA sound/soc/davinci/davinci-sffsdr.c:#ifndef CONFIG_SFFSDR_FPGA sound/soc/davinci/davinci-sffsdr.c:#ifndef CONFIG_SFFSDR_FPGA -- (λ x . x x) (λ x . x x) -- See how beatiful the lambda is No documentation is better than bad documentation -- Das Ausdrucken dieser Mail wird urheberrechtlich verfolgt.

3 3

[alsa-devel] [PATCH] ASoC: TWL4030: Add configurable delay after digimic enable
by Peter Ujfalusi 21 Jul '10

21 Jul '10

When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, I have been looking after this problem for a year now (with low priority), and finally I did figured out what is caused the glitch at the beginning of the recording via the digimic interface. Adding platform dependent delay after enabling the digimic interface fixes the issue I am tracking, in other HW setups with different components the delay might not be needed. Peter sound/soc/codecs/twl4030.c | 25 +++++++++++++++++++++---- sound/soc/codecs/twl4030.h | 1 + 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6fd6d0b..bd557c2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -143,6 +143,9 @@ struct twl4030_priv { u8 earpiece_enabled; u8 predrivel_enabled, predriver_enabled; u8 carkitl_enabled, carkitr_enabled; + + /* Delay needed after enabling the digimic interface */ + unsigned int digimic_delay; }; /* @@ -312,6 +315,8 @@ static void twl4030_init_chip(struct platform_device *pdev) if (!setup) return; + twl4030->digimic_delay = setup->digimic_delay; + /* Configuration for headset ramp delay from setup data */ if (setup->sysclk != twl4030->sysclk) dev_warn(codec->dev, @@ -855,6 +860,16 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, return 0; } +static int digimic_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); + + if (twl4030->digimic_delay) + mdelay(twl4030->digimic_delay); + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -1439,10 +1454,12 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_PGA("ADC Physical Right", TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic0 Enable", - TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic1 Enable", - TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("Digimic0 Enable", + TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Digimic1 Enable", + TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 788e3d1..6c57430 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -41,6 +41,7 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; + unsigned int digimic_delay; /* in ms */ unsigned int sysclk; unsigned int offset_cncl_path; unsigned int check_defaults:1; -- 1.7.1.1

3 2

[alsa-devel] [PATCHv2] ASoC: multi-component - Add Aquila sound driver
by Chanwoo Choi 21 Jul '10

21 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2S to support sound. This uses the I2Sv4 driver compatible with I2Sv5 to run sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 301 +++++++++++++++++++++++++++++++++++++ 3 files changed, 312 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..c09b402 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..375f4c0 --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,301 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.com> + * + * 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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define WM8994_DAI_HIFI 0 +#define WM8994_DAI_VOICE 1 + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_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_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_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 pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "Voice", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], + .codec_dai_id = WM8994_DAI_HIFI, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_drv = &voice_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], + .codec_dai_id = WM8994_DAI_VOICE, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + aquila_snd_device = platform_device_alloc("soc-audio", -1); + if (!aquila_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&aquila_snd_device->dev, + 0, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(aquila_snd_device, &aquila); + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

2 1

[alsa-devel] [PATCH] Memory leak in namehint.c
by John Lindgren 21 Jul '10

21 Jul '10

get_card_name() can be called more than once on the same list, so it must free the previous list->cardname before replacing it. Signed-off-by: John Lindgren <john.lindgren(a)tds.net> --- src/control/namehint.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/src/control/namehint.c b/src/control/namehint.c index 78572d8..fc1cb27 100644 --- a/src/control/namehint.c +++ b/src/control/namehint.c @@ -471,6 +471,8 @@ static int get_card_name(struct hint_list *list, int card) char scard[16], *s; int err; + if (list->cardname) + free(list->cardname); err = snd_card_get_name(card, &list->cardname); if (err <= 0) return 0;

2 2

[alsa-devel] Question for ALSA I/O Plugin
by Raul Xiong 21 Jul '10

21 Jul '10

Hi all, I know that there are two kinds of ALSA external Plugin: filter-type and I/O type. Now I have an I/O Plugin named phone. Can it be a slave of a internal Plugin? E.g. can I use this way: aplay -Dplug:phone wav. Thanks!

1 0

[alsa-devel] snd-hda-intel acl269 toshiba r700 speakers work, headphones do not
by Thomas Hoffmann 21 Jul '10

21 Jul '10

Dear Alsa developers, Sound works from the built in laptop speakers. When headphones are plugged in, the speakers properly mute, but there is no sound coming from the headphones. I've tried some of the options to the alsa-base.conf file to no avail; generally they made it so sound came out of the speakers always, even when headphones were plugged in. I previously sent an e-mail to this list server: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5064 where they suggested I e-mail you. My system info is available from http://www.alsa-project.org/db/?f=f9aaf6e43b669cf2c4b2215781a414b12843e2a3 Best, Tom

3 3

[alsa-devel] [PATCH 1/3] ASoC: multi-component - Add Aquila sound driver
by Chanwoo Choi 21 Jul '10

21 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2S to support sound. This uses the I2Sv4 driver compatible with I2Sv5 to run sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 300 +++++++++++++++++++++++++++++++++++++ 3 files changed, 311 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..c09b402 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..6ff5068 --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,300 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.com> + * + * 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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define WM8994_DAI_HIFI 0 +#define WM8994_DAI_VOICE 1 + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_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_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_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 pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "Voice", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], + .codec_dai_id = WM8994_DAI_HIFI, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_drv = &voice_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], + .codec_dai_id = WM8994_DAI_VOICE, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&aquila_snd_device->dev, + 0, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(aquila_snd_device, &aquila); + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

3 4

[alsa-devel] Audio dropouts with pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32).
by VDR User 20 Jul '10

20 Jul '10

Hi. I've been experiencing audio drops on dvb streams that use ac3 audio (h264+ac3 hdtv streams) and occasionally mp2 audio (sdtv channels) . When the ac3 audio drops out, it usually recovers after a few seconds though sometimes it never does. When mp2 drops out, it almost never recovers. Retuning the channel always fixes the drop out. When this happens there's usually no interruption in the video so I question whether stream errors are causing the problem. When a dropout occurs, I always see the following in my xine log: pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). I (and others apparently) have been experiencing this problem for quite some time. I have read about and tried adding notsc to no kernel boot line, which has had no affect. Here's a clip from the most recent drop outs: audio jump, diff=-90720 pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). audio_out: inserting 21713 0-frames to fill a gap of 40723 pts video_out: throwing away image with pts 176336725 because it's too old (diff : 10842). video_out: throwing away image with pts 176339939 because it's too old (diff : 16618). video jump 200 frames delivered, 0 frames skipped, 2 frames discarded video jump audio jump, diff=-103680 pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). audio_out: inserting 21233 0-frames to fill a gap of 39823 pts video_out: throwing away image with pts 183068800 because it's too old (diff : 6699). video jump video_out: throwing away image with pts 183071835 because it's too old (diff : 12728). pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed (-32). 200 frames delivered, 0 frames skipped, 2 frames discarded Some system specs are: debian testing kernel 2.6.34.1 alsa-base 1.0.23+dfsg-1 alsa-utils 1.0.23-2 alsa drivers compiled from today's snapshot (2010-07-20). in /etc/modprobe.d/alsa-base.conf: options snd-hda-intel enable_msi=1 probe_mask=0xffff,0xfff2 lsmod: snd_hda_codec_nvhdmi 9750 1 snd_hda_intel 14764 0 snd_hda_codec 48881 2 snd_hda_codec_nvhdmi,snd_hda_intel snd_hwdep 3566 1 snd_hda_codec snd_pcm 44172 2 snd_hda_intel,snd_hda_codec snd_timer 11991 1 snd_pcm snd 28664 6 snd_hda_codec_nvhdmi,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer soundcore 548 1 snd snd_page_alloc 4669 2 snd_hda_intel,snd_pcm I'd greatly appreciate any help in getting this problem worked out and hopefully solved once and for all. Thanks in advance for _anything_. Best regards, Derek

1 0

[alsa-devel] [PATCH] wm8960: fix compile error
by Axel Lin 20 Jul '10

20 Jul '10

This patch fixes compile error: sound/soc/codecs/wm8960.c: In function 'wm8960_hw_params': sound/soc/codecs/wm8960.c:511: error: 'codec' undeclared (first use in this function) sound/soc/codecs/wm8960.c:511: error: (Each undeclared identifier is reported only once sound/soc/codecs/wm8960.c:511: error: for each function it appears in.) make[3]: *** [sound/soc/codecs/wm8960.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> --- This patch is against Liam's multi-component branch. Axel sound/soc/codecs/wm8960.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e6df6e3..599c559 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -508,8 +508,8 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); struct snd_soc_codec *codec = rtd->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; int i; -- 1.5.4.3

2 3

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 20 Jul '10

20 Jul '10

The following changes since commit 0c74a939d84730818bc9bf1efa5ad7184bb2f248: ASoC: au1x: fix section mismatch in psc-i2s.c (2010-07-13 12:39:14 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Axel Lin (2): ASoC: wm8727: add a missing return in wm8727_platform_probe ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register Chanwoo Choi (2): ASoC: Select wm_hubs automatically for WM8994 ASoC:Support Samsung SoC(S5P) in I2Sv2 Eric Bénard (1): ASoC: eukrea-tlv320: add support for cpuimx35sd Kulikov Vasiliy (3): ASoC: davinci: check kzalloc() result (typo) ASoC: imx: check kzalloc() result and fix memory leak ASoC: imx: check kzalloc() result and fix memory leak Kuninori Morimoto (1): ASoC: fsi: Add specified ID for soc-audio Mark Brown (8): Merge branch 'for-2.6.35' into for-2.6.36 ASoC: Handle read failures in codec_reg ASoC: Remove duplicate AUX definition from WM8776 Merge branch 'for-2.6.35' into for-2.6.36 ASoC: Explicitly disable DC servo on WM hubs headphone powerdown ASoC: Implement WM8994 AIF1ADC2 paths Merge branch 'for-2.6.35' into for-2.6.36 ASoC: Unconditionally enable WM8994 AIF1ADC TDM mode Sekhar Nori (1): ASoC: davinci: let platform data define edma queue numbers arch/arm/mach-davinci/board-da830-evm.c | 2 +- arch/arm/mach-davinci/board-da850-evm.c | 2 +- arch/arm/mach-davinci/board-dm646x-evm.c | 4 +- arch/arm/mach-davinci/include/mach/asp.h | 3 +- include/sound/sh_fsi.h | 3 ++ sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/wm8727.c | 2 + sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8994.c | 32 ++++++++++++++++++++++++++++++ sound/soc/codecs/wm_hubs.c | 2 + sound/soc/davinci/davinci-i2s.c | 10 +++++++++ sound/soc/davinci/davinci-mcasp.c | 6 +++- sound/soc/davinci/davinci-pcm.c | 5 ++- sound/soc/davinci/davinci-pcm.h | 3 +- sound/soc/davinci/davinci-vcif.c | 2 +- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 +- sound/soc/imx/imx-pcm-dma-mx2.c | 6 ++++- sound/soc/imx/imx-pcm-fiq.c | 6 ++++- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 +- sound/soc/sh/fsi-ak4642.c | 4 +- sound/soc/sh/fsi-da7210.c | 4 +- sound/soc/soc-core.c | 22 +++++++++++++++---- 24 files changed, 103 insertions(+), 29 deletions(-)

2 1

[alsa-devel] [PATCH RFC 0/1] ASoC: WL1273 FM Radio codes
by Matti J. Aaltonen 20 Jul '10

20 Jul '10

Hello. This is the ALSA part of our WL1273 FM Radio driver. I've sent the complete driver to the linux-media list for review. But as people on that list are not in general ALSA specialists I'm sending the codec part here. So all comments are welcome. Thank you, Matti A. Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec. sound/soc/codecs/wl1273.c | 593 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 42 ++++ 2 files changed, 635 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h

2 3

[alsa-devel] ASoC update for 2.6.35
by Mark Brown 20 Jul '10

20 Jul '10

The following changes since commit 637727838a5e82bc9285ab078a793eaae590bacb: ASoC: fsi: fixup wrong value setting order of TDM (2010-07-13 12:26:26 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Axel Lin (2): ASoC: wm8727: add a missing return in wm8727_platform_probe ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register Chanwoo Choi (1): ASoC: Select wm_hubs automatically for WM8994 Mark Brown (1): ASoC: Remove duplicate AUX definition from WM8776 Another series of small, driver-specific fixes. sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/wm8727.c | 2 ++ sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8988.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH - alsa-lib 0/3] Fixes for snd_tlv_convert_from_dB
by Peter Ujfalusi 20 Jul '10

20 Jul '10

Hello, One cosmetic fix, and two real fixes for alsa-lib's tlv handling. There were two separate issue with the snd_tlv_convert_from_dB function: 1. When user asked for out of range dB value, and the control had SND_CTL_TLVT_DB_RANGE, the control's full range was not checked against the requested value, which resulted failure. 2. When there are 'holes' in the SND_CTL_TLVT_DB_RANGE array, snd_tlv_convert_from_dB will fail to find the apropriate raw value. For issue #2, I have sent patches for the drivers I maintain, but Mark Brown suggested to fix the issue in alsa-lib instead of converting the driver's non-overlapping to overlapping mapping: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-July/029483.html So the following series for alsa-lib, will fix the SND_CTL_TLVT_DB_RANGE handling. --- Peter Ujfalusi (3): tlv: Check out of range dB with SND_CTL_TLVT_DB_RANGE tlv: Handle 'holes' in SND_CTL_TLVT_DB_RANGE array tlv: Remove tailing tab after snd_ctl_get_dB_range function src/control/tlv.c | 17 +++++++++++++++-- 1 files changed, 15 insertions(+), 2 deletions(-)

4 18

[alsa-devel] [PATCH 0/12] sound/alsa/soc/codec: fix memory leak and resource relaim in error path
by Axel Lin 20 Jul '10

20 Jul '10

This serial of patches fixes memory leak and resource relaim in error path [PATCH 1/12] ad1836: fix a memory leak if another ad1836 is registered [PATCH 2/12] ak4642: fix a memory leak if failed to initialise AK4642 [PATCH 3/12] da7210: fix a memory leak if failed to initialise da7210 audio codec [PATCH 4/12] wm8523: fix resource reclaim in wm8523_register error path [PATCH 5/12] wm8711: fix a memory leak if another WM8711 is registered [PATCH 6/12] wm8904: fix resource reclaim in wm8904_register error path [PATCH 7/12] wm8940: fix resource reclaim in wm8940_register error path [PATCH 8/12] wm8955: fix resource reclaim in wm8955_register error path [PATCH 9/12] wm8961: fix resource reclaim in wm8961_register error path [PATCH 10/12] wm8974: fix a memory leak if another WM8974 is registered [PATCH 11/12] wm8978: fix a memory leak if another WM8978 is registered [PATCH 12/12] wm9081: fix resource reclaim in wm9081_register error path Regards, Axel

6 25

Re: [alsa-devel] [PATCH v2] asoc: davinci: let platform data define edma queue numbers
by Mark Brown 20 Jul '10

20 Jul '10

On Mon, Jul 19, 2010 at 12:31:16PM +0530, Sekhar Nori wrote: > Currently the EDMA queue to be used by for servicing ASP through > internal RAM is fixed to EDMAQ_0 and that to service internal RAM > from external RAM is fixed to EDMAQ_1. Applied, thanks. It's "ASoC", BTW.

1 0

[alsa-devel] [PATCH 2/3] ASoC: multi-component - Support Samsung SoC(S5P) in I2Sv2
by Chanwoo Choi 20 Jul '10

20 Jul '10

This patch modify I2Sv2 driver to support Samsung SoC(S5PV210). Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 6575410..f4fbc0e 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,7 +32,8 @@ #undef S3C_IIS_V2_SUPPORTED -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \ + || defined(CONFIG_CPU_S5PV210) #define S3C_IIS_V2_SUPPORTED #endif -- 1.6.3.3

3 3

[alsa-devel] [PATCH 3/3] ASoC: multi-component - Enable WM8993/4 common code to support WM8994
by Chanwoo Choi 20 Jul '10

20 Jul '10

This patch enable WM8993/4 common code which is used in WM8994 driver. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/codecs/Kconfig | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea1f5ed..17e28eb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -85,8 +85,8 @@ config SND_SOC_ALL_CODECS config SND_SOC_WM_HUBS tristate - default y if SND_SOC_WM8993=y - default m if SND_SOC_WM8993=m + default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y + default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m config SND_SOC_AC97_CODEC tristate -- 1.6.3.3

3 2

[alsa-devel] [PATCH] ASoC: da7210: Add HeadPhone Playback Volume control
by Kuninori Morimoto 20 Jul '10

20 Jul '10

HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. This patch add HeadPhone Playback Volume 40 as default, because DA7210 doesn't have default volume. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/da7210.c | 37 +++++++++++++++++++++++++++++++++---- 1 files changed, 33 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 75af2d6..f0af2c8 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -145,6 +145,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -227,10 +250,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -572,6 +591,13 @@ static int da7210_init(struct da7210_priv *da7210) /* Activate all enabled subsystem */ da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + /* + * HeadPhone PlayBack Volume 40 is given here + * because DA7210 doesn't have default volume. + */ + snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); + snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); + return ret; init_err: @@ -656,6 +682,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: -- 1.7.0.4

2 2

[alsa-devel] [PATCH v2 - alsa-lib 0/3] Fixes for snd_tlv_convert_from_dB
by Peter Ujfalusi 20 Jul '10

20 Jul '10

Hello, Changes since the initial version: - In patch one the range check has been converted to be more intuitive Intro part from the first version: One cosmetic fix, and two real fixes for alsa-lib's tlv handling. There were two separate issue with the snd_tlv_convert_from_dB function: 1. When user asked for out of range dB value, and the control had SND_CTL_TLVT_DB_RANGE, the control's full range was not checked against the requested value, which resulted failure. 2. When there are 'holes' in the SND_CTL_TLVT_DB_RANGE array, snd_tlv_convert_from_dB will fail to find the apropriate raw value. For issue #2, I have sent patches for the drivers I maintain, but Mark Brown suggested to fix the issue in alsa-lib instead of converting the driver's non-overlapping to overlapping mapping: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-July/029483.html So the following series for alsa-lib, will fix the SND_CTL_TLVT_DB_RANGE handling. --- Peter Ujfalusi (3): tlv: Check out of range dB with SND_CTL_TLVT_DB_RANGE tlv: Handle 'holes' in SND_CTL_TLVT_DB_RANGE array tlv: Remove tailing tab after snd_ctl_get_dB_range function src/control/tlv.c | 20 ++++++++++++++++++-- 1 files changed, 18 insertions(+), 2 deletions(-)

1 3

[alsa-devel] [PATCH 0/3] ASoC: twl4030/tpa6130a2: DB_RANGE mapping fixes
by Peter Ujfalusi 20 Jul '10

20 Jul '10

Hello, There were 'holes' in the DB_RANGE and SCALE_ITEM mapping, which results failure, when user space requests dB value in these holes, since alsa-lib will fail to find a range, and can not round the requested dB to raw value. Fixing the drivers I'm maintaining to correctly fill the tlv struct. Peter --- Peter Ujfalusi (3): ASoC: TWL4030: Fix DB_RANGE tlv mapping for digital loopback ASoC: tpa6130a2: Fix DB_RANGE mapping for tpa6130a2 chip ASoC: tpa6130a2: Fix DB_RANGE mapping for tpa6140a2 chip sound/soc/codecs/tpa6130a2.c | 22 ++++++++++++++-------- sound/soc/codecs/twl4030.c | 4 ++-- 2 files changed, 16 insertions(+), 10 deletions(-)

3 15

[alsa-devel] [PATCH v2] ASoC: da7210: Add HeadPhone Playback Volume control
by Kuninori Morimoto 20 Jul '10

20 Jul '10

HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. This patch add HeadPhone Playback Volume 40 as default, because DA7210 doesn't have default volume. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 o based on ASoC's git tree (add #include <sound/tlv.h>) sound/soc/codecs/da7210.c | 38 ++++++++++++++++++++++++++++++++++---- 1 files changed, 34 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a83aa18..ec292cc 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -23,6 +23,7 @@ #include <sound/pcm_params.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "da7210.h" @@ -136,6 +137,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -218,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -563,6 +583,13 @@ static int da7210_init(struct da7210_priv *da7210) /* Activate all enabled subsystem */ da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + /* + * HeadPhone PlayBack Volume 40 is given here + * because DA7210 doesn't have default volume. + */ + snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); + snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); + return ret; init_err: @@ -647,6 +674,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: -- 1.7.0.4

1 0

[alsa-devel] [PATCH] Add conexant quirk for AMD based Lenovo G series machines
by Jerone Young 19 Jul '10

19 Jul '10

This is a follow on patch adds support for AMD based Lenovo G series machines, such as the Lenovo G555. Signed-off-by: Jerone Young <jerone.young(a)canonical.com> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index c49030b..774edb9 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -3000,6 +3000,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), {} };

2 2

[alsa-devel] DMA Physical Addr Issue.
by Adish Kuvelker 19 Jul '10

19 Jul '10

Hello, I have allocated the DMA memory using the function "snd_pcm_lib_preallocate_pages_for_all" passing SNDRV_DMA_TYPE_DEV and "card->dev" as inputs to the function. Later, in the hw_params call out function I allocate pages to the requested DMA memory region using the function "snd_pcm_lib_malloc_pages". I play a wav file "Noise.wav" from command line by giving the following command ./aplay --file-type wav -f dat Noise.wav And, when I print the DMA region physical address in the "prepare" call out through "dma_addr" member of the runtime structure I get a weird decimal value of "-1036562432" as the physical address. Can anyone point where I might be going wrong in the whole DMA memory allocation? Regards; Aadish

2 1

[alsa-devel] [PATCH 12/15] pci: riptide: check kzalloc() result
by Kulikov Vasiliy 19 Jul '10

19 Jul '10

If kzalloc() fails exit with -ENOMEM. Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/pci/riptide/riptide.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 59d7996..f64fb7d 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1615,7 +1615,10 @@ static int snd_riptide_playback_open(struct snd_pcm_substream *substream) chip->playback_substream[sub_num] = substream; runtime->hw = snd_riptide_playback; + data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; data->paths = lbus_play_paths[sub_num]; data->id = play_ids[sub_num]; data->source = play_sources[sub_num]; @@ -1635,7 +1638,10 @@ static int snd_riptide_capture_open(struct snd_pcm_substream *substream) chip->capture_substream = substream; runtime->hw = snd_riptide_capture; + data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; data->paths = lbus_rec_path; data->id = PADC; data->source = ACLNK2PADC; -- 1.7.0.4

2 1

[alsa-devel] [PATCH 11/15] pci: echoaudio: check kmalloc() result
by Kulikov Vasiliy 19 Jul '10

19 Jul '10

If kmalloc() fails exit with -ENOMEM. Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/pci/echoaudio/echoaudio.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 668a5ec..20763dd 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2250,6 +2250,8 @@ static int snd_echo_resume(struct pci_dev *pci) DE_INIT(("resume start\n")); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + if (commpage_bak == NULL) + return -ENOMEM; commpage = chip->comm_page; memcpy(commpage_bak, commpage, sizeof(struct comm_page)); -- 1.7.0.4

3 2

[alsa-devel] XRUN handling
by melwyn lobo 19 Jul '10

19 Jul '10

Hi all, I have noticed when an Xrun occurs, the bytes between the hw_ptr and appl_ptr are not copied when both hw and appl pointers are reset in snd_pcm_prepare(). Consider the following parameters: buffer_size : 22352 period_size : 1016 period_time : 23038 tstamp_mode : NONE period_step : 1 avail_min : 1016 period_event : 0 start_threshold : 22352 stop_threshold : 22352 silence_threshold: 0 silence_size : 0 In case snd_pcm_playback_avail() is less than stop threshold but greater than 0, then these bytes are missing after Xrun prepare and start stage. Is my analysis correct? If so how do we ensure correct behaviour ie byte accuracy after Xrun. Thanks, M.

4 11

[alsa-devel] [PATCH 2/2]ASoC: multi-component - Fix S5P compile error
by Chanwoo Choi 19 Jul '10

19 Jul '10

This patch fix compile error on using I2S and WM8994 common code. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/codecs/Kconfig | 4 ++-- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 ++- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea1f5ed..55bc2be 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -85,8 +85,8 @@ config SND_SOC_ALL_CODECS config SND_SOC_WM_HUBS tristate - default y if SND_SOC_WM8993=y - default m if SND_SOC_WM8993=m + default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y + default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m config SND_SOC_AC97_CODEC tristate diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 6575410..f4fbc0e 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,7 +32,8 @@ #undef S3C_IIS_V2_SUPPORTED -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \ + || defined(CONFIG_CPU_S5PV210) #define S3C_IIS_V2_SUPPORTED #endif

2 2

[alsa-devel] [PATCH 1/2]ASoC: multi-component - Add Aquila sound driver
by Chanwoo Choi 19 Jul '10

19 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2S to support sound. This uses the I2Sv4 driver compatible with I2Sv5 to run sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 300 +++++++++++++++++++++++++++++++++++++ 3 files changed, 311 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..c09b402 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..6ff5068 --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,300 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.com> + * + * 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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define WM8994_DAI_HIFI 0 +#define WM8994_DAI_VOICE 1 + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_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_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_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 pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "Voice", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], + .codec_dai_id = WM8994_DAI_HIFI, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_drv = &voice_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], + .codec_dai_id = WM8994_DAI_VOICE, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&aquila_snd_device->dev, + 0, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(aquila_snd_device, &aquila); + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

1 0

[alsa-devel] Clicking and stuttering audio when playing two or more files
by Adam Rosenberg 19 Jul '10

19 Jul '10

I have audio output perfectly from my second CS42448 CODEC now, but only when one file is playing. When I attempt to play two files, one or both of the files play with clicking and stuttering. I am also noticing that the PCM prepare operation is occurring more than once. What could be causing this problem? HW settings are: buffer_bytes_max = 32768 period_bytes min = 8 max = 32768 periods min = 1 max = 4096 PCM runtime is reporting: frames per period = 1114 periods = 2 total frames = 2228 frame bytes = 8 Thank you, Adam

2 1

[alsa-devel] [PATCH] Add conexant quirk for Lenovo G series support
by Jerone Young 19 Jul '10

19 Jul '10

This patch adds quirk add support for Lenovo G series laptops. Launchpad bug can be found here: https://bugs.launchpad.net/ubuntu/+source/linux-backports-modules-2.6.32/+b… Signed-off-by: Jerone Young <jerone.young(a)canonical.com> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index 54f7419..c49030b 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -2999,6 +2999,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), {} };

1 0

[alsa-devel] [PATCH] asoc: davinci: let platform data define edma queue numbers
by Sekhar Nori 17 Jul '10

17 Jul '10

Currently the EDMA queue to be used by for servicing ASP through internal RAM is fixed to EDMAQ_0 and that to service internal RAM from external RAM is fixed to EDMAQ_1. This may not be the desirable configuration on all platforms. For example, on DM365, queue 0 has large fifo size and is more suitable for video transfers. Having audio and video transfers on the same queue may lead to starvation on audio side. platform data as defined currently passes a queue number to the driver but that remains unused inside the driver. Fix this by defining one queue each for ASP and RAM transfers in the platform data and using it inside the driver. Since EDMAQ_0 maps to 0, thats the queue that will be used if the asp queue number is not initialized. None of the platforms currently utilize ping-pong transfers through internal RAM so that functionality remains unchanged too. This patch has been tested on DM644x and OMAP-L138 EVMs. Signed-off-by: Sekhar Nori <nsekhar(a)ti.com> --- This patch applies to latest of Linus's tree. arch/arm/mach-davinci/board-da830-evm.c | 2 +- arch/arm/mach-davinci/board-da850-evm.c | 2 +- arch/arm/mach-davinci/board-dm646x-evm.c | 4 ++-- arch/arm/mach-davinci/include/mach/asp.h | 3 ++- sound/soc/davinci/davinci-i2s.c | 10 ++++++++++ sound/soc/davinci/davinci-mcasp.c | 6 ++++-- sound/soc/davinci/davinci-pcm.c | 5 +++-- sound/soc/davinci/davinci-pcm.h | 3 ++- 8 files changed, 25 insertions(+), 10 deletions(-) diff --git a/arch/arm/mach-davinci/board-da830-evm.c b/arch/arm/mach-davinci/board-da830-evm.c index 212d970..bc384d3 100644 --- a/arch/arm/mach-davinci/board-da830-evm.c +++ b/arch/arm/mach-davinci/board-da830-evm.c @@ -208,7 +208,7 @@ static struct snd_platform_data da830_evm_snd_data = { .num_serializer = ARRAY_SIZE(da830_iis_serializer_direction), .tdm_slots = 2, .serial_dir = da830_iis_serializer_direction, - .eventq_no = EVENTQ_0, + .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_2, .txnumevt = 1, .rxnumevt = 1, diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index 2ec3095..d4ec18d 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -342,7 +342,7 @@ static struct snd_platform_data da850_evm_snd_data = { .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), .tdm_slots = 2, .serial_dir = da850_iis_serializer_direction, - .eventq_no = EVENTQ_1, + .asp_chan_q = EVENTQ_1, .version = MCASP_VERSION_2, .txnumevt = 1, .rxnumevt = 1, diff --git a/arch/arm/mach-davinci/board-dm646x-evm.c b/arch/arm/mach-davinci/board-dm646x-evm.c index 6d88893..87521f2 100644 --- a/arch/arm/mach-davinci/board-dm646x-evm.c +++ b/arch/arm/mach-davinci/board-dm646x-evm.c @@ -323,7 +323,7 @@ static struct snd_platform_data dm646x_evm_snd_data[] = { .num_serializer = ARRAY_SIZE(dm646x_iis_serializer_direction), .tdm_slots = 2, .serial_dir = dm646x_iis_serializer_direction, - .eventq_no = EVENTQ_0, + .asp_chan_q = EVENTQ_0, }, { .tx_dma_offset = 0x400, @@ -332,7 +332,7 @@ static struct snd_platform_data dm646x_evm_snd_data[] = { .num_serializer = ARRAY_SIZE(dm646x_dit_serializer_direction), .tdm_slots = 32, .serial_dir = dm646x_dit_serializer_direction, - .eventq_no = EVENTQ_0, + .asp_chan_q = EVENTQ_0, }, }; diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..5d0a4c7 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -52,7 +52,8 @@ struct snd_platform_data { u32 tx_dma_offset; u32 rx_dma_offset; - enum dma_event_q eventq_no; /* event queue number */ + enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ + enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..4c1a7d2 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -526,6 +526,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; + enum dma_event_q asp_chan_q = EVENTQ_0; + enum dma_event_q ram_chan_q = EVENTQ_1; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -552,7 +554,15 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + asp_chan_q = pdata->asp_chan_q; + ram_chan_q = pdata->ram_chan_q; } + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; + dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d395509..b247208 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -890,7 +890,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->rxnumevt = pdata->rxnumevt; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; - dma_data->eventq_no = pdata->eventq_no; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + io_v2p(dev->base)); @@ -904,7 +905,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; - dma_data->eventq_no = pdata->eventq_no; + dma_data->asp_chan_q = pdata->asp_chan_q; + dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + io_v2p(dev->base)); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2dc406f..f026ac5 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -381,7 +381,7 @@ static int request_ping_pong(struct snd_pcm_substream *substream, /* Request ram master channel */ link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, davinci_pcm_dma_irq, substream, - EVENTQ_1); + prtd->params->ram_chan_q); if (link < 0) goto exit1; @@ -477,7 +477,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) /* Request asp master DMA channel */ link = prtd->asp_channel = edma_alloc_channel(params->channel, - davinci_pcm_dma_irq, substream, EVENTQ_0); + davinci_pcm_dma_irq, substream, + prtd->params->asp_chan_q); if (link < 0) goto exit1; diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0764944..b799a02 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -21,7 +21,8 @@ struct davinci_pcm_dma_params { unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ unsigned sram_size; - enum dma_event_q eventq_no; /* event queue number */ + enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ + enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; -- 1.6.2.4

3 2

[alsa-devel] [PATCH] ASoC: fsi: Add specified ID for soc-audio
by Kuninori Morimoto 17 Jul '10

17 Jul '10

Specified ID is necessary, when some codecs are used with FSI. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark. Thank you about your advice. it solved my issue. This patch is v2 of "ASoC: FSI - codecs settings" include/sound/sh_fsi.h | 3 +++ sound/soc/sh/fsi-ak4642.c | 4 ++-- sound/soc/sh/fsi-da7210.c | 4 ++-- 3 files changed, 7 insertions(+), 4 deletions(-) diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index c022736..6463fd6 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -12,6 +12,9 @@ * published by the Free Software Foundation. */ +#define FSI_PORT_A 0 +#define FSI_PORT_B 1 + /* flags format * 0xABCDEEFF diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index be01854..2c61ed2 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -38,7 +38,7 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) static struct snd_soc_dai_link fsi_dai_link = { .name = "AK4642", .stream_name = "AK4642", - .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .cpu_dai = &fsi_soc_dai[FSI_PORT_A], .codec_dai = &ak4642_dai, .init = fsi_ak4642_dai_init, .ops = NULL, @@ -62,7 +62,7 @@ static int __init fsi_ak4642_init(void) { int ret = -ENOMEM; - fsi_snd_device = platform_device_alloc("soc-audio", -1); + fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A); if (!fsi_snd_device) goto out; diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index 33b4d17..5774449 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -33,7 +33,7 @@ static int fsi_da7210_init(struct snd_soc_codec *codec) static struct snd_soc_dai_link fsi_da7210_dai = { .name = "DA7210", .stream_name = "DA7210", - .cpu_dai = &fsi_soc_dai[1], /* FSI B */ + .cpu_dai = &fsi_soc_dai[FSI_PORT_B], .codec_dai = &da7210_dai, .init = fsi_da7210_init, }; @@ -56,7 +56,7 @@ static int __init fsi_da7210_sound_init(void) { int ret; - fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); + fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); if (!fsi_da7210_snd_device) return -ENOMEM; -- 1.7.0.4

3 2

[alsa-devel] [PATCH 13/15] sound/soc: davinci: check kzalloc() result (typo)
by Kulikov Vasiliy 17 Jul '10

17 Jul '10

The code checks 'davinci_vc' after kzalloc() and do not checks 'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that it is a typo (autocompletion?). Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/soc/davinci/davinci-vcif.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9aa980d..4867853 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -203,7 +203,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) int ret; davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); - if (!davinci_vc) { + if (!davinci_vcif_dev) { dev_dbg(&pdev->dev, "could not allocate memory for private data\n"); return -ENOMEM; -- 1.7.0.4

3 2

[alsa-devel] [PATCH v2] ASoC: TWL6040: Correct widget handling for drivers
by Jorge Eduardo Candelaria 17 Jul '10

17 Jul '10

In order to reduce pop-noise at powering up/down of the DACs and Drivers, these components have to be handled in a specific sequence. Headset, Handsfree, and Earphone drivers are now registered as PGA components to ensure DACs are enabled first. Also, add a delay to leave time for DACs to settle before continuing power up/down sequence. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 56 ++++++++++++++++++++++---------------------- 1 files changed, 28 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 715caef..345d86a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -360,6 +360,13 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) return 0; } +static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + msleep(1); + return 0; +} + static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -371,6 +378,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, else priv->non_lp--; + msleep(1); + return 0; } @@ -471,20 +480,6 @@ static const struct snd_kcontrol_new hfdacl_switch_controls = static const struct snd_kcontrol_new hfdacr_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); -/* Headset driver switches */ -static const struct snd_kcontrol_new hsl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); - -static const struct snd_kcontrol_new hsr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); - -/* Handsfree driver switches */ -static const struct snd_kcontrol_new hfl_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); - -static const struct snd_kcontrol_new hfr_driver_switch_controls = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); - static const struct snd_kcontrol_new ep_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); @@ -548,10 +543,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0), /* DACs */ - SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", - TWL6040_REG_HSLCTL, 0, 0), - SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", - TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0, + twl6040_hs_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", TWL6040_REG_HFLCTL, 0, 0, twl6040_power_mode_event, @@ -571,18 +570,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HFDAC Right Playback", SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Left Driver", - SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), - SND_SOC_DAPM_SWITCH("Headset Right Driver", - SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), - SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", - SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + /* Analog playback drivers */ + SND_SOC_DAPM_PGA_E("Handsfree Left Driver", + TWL6040_REG_HFLCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", - SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + SND_SOC_DAPM_PGA_E("Handsfree Right Driver", + TWL6040_REG_HFRCTL, 4, 0, NULL, 0, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA("Headset Left Driver", + TWL6040_REG_HSLCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Driver", + TWL6040_REG_HSRCTL, 2, 0, NULL, 0), SND_SOC_DAPM_SWITCH_E("Earphone Driver", SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, twl6040_power_mode_event, @@ -616,8 +616,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSDAC Left Playback", "Switch", "HSDAC Left"}, {"HSDAC Right Playback", "Switch", "HSDAC Right"}, - {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, - {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + {"Headset Left Driver", NULL, "HSDAC Left Playback"}, + {"Headset Right Driver", NULL, "HSDAC Right Playback"}, {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, -- 1.7.0.4

3 2

[alsa-devel] [PATCHv2 0/4] ASoC: multi-component - Samsung S3C Platform
by Seungwhan Youn 17 Jul '10

17 Jul '10

Hi, This is a second version of patch series adding support for multi component model on Samsung S3C Platform. This patch series can be apply upon "ASoC: multi-component - Fix S3C compile error" in the first patch of V1 that already applied by Liam. Changes since V1: - Support for GPIO, DMA and SFR settings pull from platform device resource. - Modify description of add WM9713 platform device on smdk board. The following patch series contains: [PATCHv2 1/4] ASoC: S3C64XX: Move GPIO settings to platform [PATCHv2 2/4] ASoC: S3C64XX: Move DMA settings to platform [PATCHv2 3/4] ASoC: multi-component - S3C - Get original functionality [PATCHv2 4/4] ASoC: multi-component - S3C - Add WM9713 platform device Thanks, Claude

3 7

[alsa-devel] [PATCH] asoc: multi-component: mpc8610: fix typo in mpc8610_hpcd_data
by Timur Tabi 17 Jul '10

17 Jul '10

The dma_id[] variable in struct mpc8610_hpcd_data is a zero-length array, when it should be of length 2. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Liam, please squash this into your repository. sound/soc/fsl/mpc8610_hpcd.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 81ab639..8665cc2 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -37,7 +37,7 @@ struct mpc8610_hpcd_data { unsigned int cpu_clk_direction; unsigned int clk_frequency; unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ - unsigned int dma_id[0]; /* 0 = DMA1, 1 = DMA2, etc */ + unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ char dai_name[32]; }; -- 1.7.0.1

3 2

Re: [alsa-devel] [PlanetCCRMA] [LAD] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 17 Jul '10

17 Jul '10

On Fri, Jul 16, 2010 at 5:27 AM, Geoff King <gsking1(a)gmail.com> wrote: > Niels, Thank you for taking the time to work on improving Envy24Ctl. > I tried your patch for the peaks and it seemed to work fine for me. I > can't comment on the code (as I'm much more musician than programmer), > but had no problem patching and installing. Looking forward to trying > this new patch and whatever else you come up with. Geoff Geoff -- thank you for trying out the patch! (And, OT, thanks for working w/ Rui to help fix http://sourceforge.net/tracker/?func=detail&atid=733076&aid=3021645&group_i… :: "help fix qtractor crash on bus changing/configuration (3021645)" -- Envy24 and multichannel qtractor users, yes, it's time for a "svn up" ). FYI, to try out the new levelmeters "easy", if you're running Fedora x86_64, use the "envy24control" binary directly, or drop the levelmeters.c file into your build: http://nielsmayer.com/npm/Efficient-Meters-Envy24Control.tgz ( for details http://nielsmayer.com/npm/Efficient-Meters-Envy24Control.README ) ........ On a completely different "note" the envy 24 manual ( http://alsa.cybermirror.org/manuals/icensemble/envy24.pdf ) has interesting info on the envy24 digital mixer that I've snapshotted: Diagram: http://nielsmayer.com/npm/envy24mixer-architecture.png Note the way it truncates in the mixer: the more inputs you "mix" at once, the fewer bits each input source gets, and it's not clear what kind of dither, or it's a straight truncate of 24... ultimately the envy24 seems oriented towards producing a 16 bit, and not 24 bit master (which makes sense given that the chip is well over a decade old and HD audio production for "prosumer" was rare): .......... 4.5.5 Multi-Track Digital Monitoring The Envy24 integrates a 36-bit resolution digital hardware mixer. The width of the data path is strictly to ensure that during processing of all the channels, under any condition, no resolution is lost. The dynamic range of the end user system will be limited by the range of the physical output devices used. In order to maintain identical gain to the input stream (i.e. 0dB), the resulting 24-bit is not msb-aligned to the 36-bit. The overflow bits correspond to the analog distortion due to saturation. The user would need to reduce the overall attenuation of the inputs to avoid clipping. Insertion of the digital mixer adds only a single sample cycle delay with respect to the original data. This extremely low latency all digital mixer provides monitoring functionality and can replace a traditional external analog input mixer. There are 20 independent audio data streams to mix and control the volume. ... .............. --Niels http://nielsmayer.com

1 0

[alsa-devel] [PATCH 15/15] sound/soc: imx: check kzalloc() result and fix memory leak
by Kulikov Vasiliy 16 Jul '10

16 Jul '10

If kzalloc() fails we must exit with -ENOMEM. Also we must free allocated runtime->private_data on error as it would be lost on next call to snd_imx_open(). Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/soc/imx/imx-pcm-fiq.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 6b518e0..b2bf272 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -192,6 +192,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream) int ret; iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + if (iprtd == NULL) + return -ENOMEM; runtime->private_data = iprtd; iprtd->substream = substream; @@ -202,8 +204,10 @@ static int snd_imx_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) + if (ret < 0) { + kfree(iprtd); return ret; + } snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); return 0; -- 1.7.0.4

1 0

[alsa-devel] [PATCH 14/15] sound/soc: imx: check kzalloc() result and fix memory leak
by Kulikov Vasiliy 16 Jul '10

16 Jul '10

If kzalloc() fails we must exit with -ENOMEM. Also we must free allocated runtime->private_data on error as it would be lost on next call to snd_imx_open(). Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/soc/imx/imx-pcm-dma-mx2.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 05f19c9..0a595da 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -292,12 +292,16 @@ static int snd_imx_open(struct snd_pcm_substream *substream) int ret; iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + if (iprtd == NULL) + return -ENOMEM; runtime->private_data = iprtd; ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) + if (ret < 0) { + kfree(iprtd); return ret; + } snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); return 0; -- 1.7.0.4

1 0

[alsa-devel] [PATCH] twl6040: fix wrong kfree in twl6040_remove and twl6040_codec_remove
by Axel Lin 16 Jul '10

16 Jul '10

Memory allocation part: We allocate a memory in twl6040_codec_probe(): priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); twl6040_codec = codec = &priv->codec; Memory release part: In twl6040_codec_remove() we should kfree(priv) instead of kfree(twl6040_codec). In twl6040_remove(), no need and should not kfree(codec). Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/twl6040.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index af36346..9d4a85f 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1068,7 +1068,6 @@ static int twl6040_remove(struct platform_device *pdev) twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - kfree(codec); return 0; } @@ -1214,7 +1213,7 @@ static int __devexit twl6040_codec_remove(struct platform_device *pdev) snd_soc_unregister_dai(&twl6040_dai); snd_soc_unregister_codec(twl6040_codec); - kfree(twl6040_codec); + kfree(priv); twl6040_codec = NULL; return 0; -- 1.5.4.3

2 2

[alsa-devel] ASoC: Davinci Voice codec compliant to arago kernel for dm365
by Raffaele Recalcati 16 Jul '10

16 Jul '10

Hi Miguel, I've used successfully the patches below on top of 2.6.34 Kevin Hilman kernel for davinci http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git Now I need to backport your patches to arago kernel http://arago-project.org/git/projects/linux-davinci.git that means 2.6.32-rc1 I use able adapt your patches to it, but I guess that there are many changes from 2.6.32-rc1 to 2.6.34 ... Can you tell me what is the best thing to do? I'm afraid you'll tell me that I need to rewrite completely your patches to be really compliant to 2.6.32-rc1. Here you are a not working log: aplay tone1KHzMono_8000.wav Playing WAVE 'tone1KHzMono_8000.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono davinci_pcm_prepare - 203 davinci_pcm_enqueue_dma - 77: period=2000,dma_offset=0,dma_pos=-2044854272 davinci_pcm: audio_set_dma_params_play channel = 79 dma_ptr = 861e0000 period_size=7d0 davinci_pcm_enqueue_dma - 84: data_type=0 davinci_pcm_enqueue_dma - 77: period=2000,dma_offset=2000,dma_pos=-2044852272 davinci_pcm: audio_set_dma_params_play channel = 79 dma_ptr = 861e07d0 period_size=7d0 davinci_pcm_enqueue_dma - 84: data_type=0 aplay: pcm_write:1528: write error: Input/output error Thanks, Raffaele --------------- Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Date: Thu Mar 11 09:32:21 2010 -0600 MFD: DaVinci Voice Codec This is the MFD driver for the DaVinci Voice codec, it has two clients: * Voice codec interface * Voice codec CQ93VC Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> commit 94a532f7156582ec7459f21baa4de6665773974a Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Date: Thu Mar 11 09:32:59 2010 -0600 ASoC: DaVinci: CQ93VC Voice Codec Currently the DM365 is the only SoC that includes this Voice Codec. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> commit 80a6609eeb0995732318c7ab672767e39a111f3b Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Date: Thu Mar 11 09:32:42 2010 -0600 ASoC: DaVinci: Voice Codec Interface This patch adds the support for the interface needed by the DaVinci Voice Codec CQ93VC. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> -- www.opensurf.it

1 0

[alsa-devel] HPI version 4.04.01
by linux＠audioscience.com 16 Jul '10

16 Jul '10

HPI stable driver 4.04.01 has just been released. Only very minor patches since the last beta version. Please also update alsa-firmware/asihpi with firmware from http://audioscience.com/internet/download/drivers/released/v4/04/01/asihpi-…

3 6

[alsa-devel] Accessing snd_soc_read/snd_soc_write directly for diagnostics
by Mike Crowe 16 Jul '10

16 Jul '10

Hi folks, I know this is a strange request, however, for diagnostic purposes in our embedded system, we would like to talk directly to the codec registers of our sound chip, bypassing alsa. I have the driver compiled into the kernel (not loaded as a module). Is it possible to create a program which can talk to the snd_soc_read and snd_soc_write functions directly? I'm not sure if this is an option, but if it is, I'd appreciate a pointer to something similar to use as a guide. TIA Mike

2 2

Re: [alsa-devel] [LAD] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 16 Jul '10

16 Jul '10

On Tue, Jul 13, 2010 at 3:46 PM, Tim E. Real <termtech(a)rogers.com> wrote: > But nothing was ever done about the slider markings. > Having them go from 0 to 164 is not really helpful. > It's not clear where the 0dB mark is. > At the very least a 0dB mark should be added. > Uh, do you feel like taking a stab at this? > IIRC it would be a bit tricky because you have to look up > the corresponding values in the IC data sheets, I think. This makes sense... I was thinking of a simple ruler with a 0dB label at 127 and a + dB label above and a - dB label below. Some thoughts on further usability improvements to envy24control: (1) In "alsamixer" the snd-ice1712-card's input levels are represented by (versus envy24control's value): 100% (163) -> ADC [dB gain: 18.00] 78% (127) -> ADC [dB gain: 0.00] 1% (001) -> ADC [dB gain: -63.00] 0% (000) -> ADC [dB gain: mute] Mixers like envy24control, kmix, alsamixergui, gst-mixer, etc, are a lot less useful than they could be because they just present sliders. At least envy24control gives numeric values, -- which is essential if you want to know stereo channels are balenced, or what unity-gain levels are for inputs. In "alsamixer", the snd-ice1712-card's output levels are represented by (versus envy24control value): 100% (127) -> DAC [dB gain: 0.00] 1% (001) -> DAC [dB gain: -63.00] 0% (000) -> DAC [dB gain: mute] As this is a separate issue from the meters, it makes sense to do the changes to the "Analog Volume" panel of envy24control as a separate patch. (2) The "0-96" sliders, feeding the digital mixer, need better labeling and mappings too: The "96" actually being "-96dB" and only valid for 16 bit depths. On the other hand, alsamixer is showing the 24 bit dB values, even when 16 bits are being used: 100% (0) -> H/W Multi [dB gain: 0.00, ...] 1% (95) -> H/W Multi [dB gain: -142.50,...] 0% (96) -> H/W Multi [dB gain: -144.00,...] Worse than the labelling, but associated -- the scaling: Right now the digital mixer's input sliders all have linear response -- which wastes the lower half of the slider and makes setting levels difficult because all the "action" is near the top. If you use an external MIDI controller for the mixer, you can use this existing option: -M, --midienhanced Use an enhanced mapping from midi controller to db slider However this does not apply to the GUI controller, which stays linear. Clearly the GUIs digital mixer controls also need a logarithmic scale (perhaps as default settings); matching that, each mixer input's meters also need to have their scale expressed "logarithmically" so as to provide, for example 50% of the display resolution for 0 to -20dB, and remaining 50% for -20dB to -96dB/-144dB. Alongside the numbers indicating the 0 to -96db L or R gain on each mixer input channel, I would also add the current peak value held in the meters in dB, which would be based on hardware metering level "0xFF" representing 0db, and hardware meter level "0x01" representing all but the last of the 9 MSB's being 0 (per http://alsa.cybermirror.org/manuals/icensemble/envy24.pdf: "Peak data derived from the absolute value of 9 msb. 00h * min - FFh max volume. Reading the register resets the meter to 00h." Other missing features: (3) Due to the hardware metering employed by envy24control, the peak meters don't actually detect "clipping" ; rather, they just detect "full scale" as 255. Is there a hardware clipped indicator on the ice1712? It doesn't make sense to turn on a big red "clipped" light for 0dB output, so the current peak-meters are "subtle," and just indicate that a peak occurred at the given level... And color-code the peak towards red as it reaches 255. (4) The "IEC958 Input Status" isn't presented, even though it's available and working, e.g from the command-line,. with SPDIF input going to the Delta66 soundcard: ..... gnulem-176-~> amixer -c M66 cget numid=50,iface=MIXER,name='Delta IEC958 Input Status' numid=50,iface=MIXER,name='Delta IEC958 Input Status' ; type=BOOLEAN,access=r-------,values=1 : values=on ..... And without any SPDIF input: ..... gnulem-177-~> amixer -c M66 cget numid=50,iface=MIXER,name='Delta IEC958 Input Status' numid=50,iface=MIXER,name='Delta IEC958 Input Status' ; type=BOOLEAN,access=r-------,values=1 : values=off Perhaps this isn't available in envy24control because it has differing levels of support across envy24-based cards? I only have a Delta66's and a Terratec Dmx6Fire 24/96 for testing, and both support the SPDIF input status. -- Niels http://nielsmayer.com PS: ALSA tip of the day. To read ice1712's hardware level-meter values from the command line: "amixer -c M66 cget iface=PCM,name='Multi Track Peak',numid=45" type=INTEGER,access=r-------,values=22,min=0,max=255,step=0 : values=0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,255,198,255,198

1 0

[alsa-devel] [PATCH] uda134x: properly free allocated memory if uda134x_soc_probe fail with unknown codec type
by Axel Lin 16 Jul '10

16 Jul '10

If uda134x_soc_probe fail with unknown codec type, it should properly free allocated memory before return -EINVAL. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This case won't be executed in current implementation because there is a model checking in the begin of uda134x_soc_probe. But since the code is there, I think it's good to make it correct. Otherwise, we can simply remove the default case. sound/soc/codecs/uda134x.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 28aac53..9bff3c6 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -566,9 +566,10 @@ static int uda134x_soc_probe(struct platform_device *pdev) ARRAY_SIZE(uda1345_snd_controls)); break; default: + ret = -EINVAL; printk(KERN_ERR "%s unknown codec type: %d", __func__, pd->model); - return -EINVAL; + break; } if (ret < 0) { -- 1.5.4.3

1 0

[alsa-devel] [PATCH 8/8] sound: asihpi: fix sign bug
by Kulikov Vasiliy 15 Jul '10

15 Jul '10

bytes_per_sec is unsigned, so if snd_pcm_format_width() return error we would not see it. Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/pci/asihpi/asihpi.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 91218f7..c80b0b8 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -460,6 +460,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); int err; u16 format; + int width; unsigned int bytes_per_sec; print_hwparams(params); @@ -512,9 +513,10 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, dpcm->hpi_buffer_attached); } bytes_per_sec = params_rate(params) * params_channels(params); - bytes_per_sec *= snd_pcm_format_width(params_format(params)); + width = snd_pcm_format_width(params_format(params)); + bytes_per_sec *= width; bytes_per_sec /= 8; - if (bytes_per_sec <= 0) + if (width < 0 || bytes_per_sec == 0) return -EINVAL; dpcm->bytes_per_sec = bytes_per_sec; -- 1.7.0.4

1 0

[alsa-devel] ALSA Help?
by Matthew Hudson 15 Jul '10

15 Jul '10

Hello, I posted similar messages to the alsa-user mail list and to ubuntu forums ( http://ubuntuforums.org/showthread.php?t=1528442 ), but haven't been able to get help. Could any of you help with this? So, I've got a great Acer Aspire 7736. Seems to have a RealTek ALC888 card & Intel HDA. I've upgraded Alsa. I've used the Drivers at the RealTek site. I've checked alsamixer over and over and over. Yet, currently I cannot get sound through the headphone jack. Originally, I had problems w/ headphone/speakers playing simlutaneously. Somewhere in my adventures I fixed that, but then only one app would use sound during a login (UGH). So, I kept poking...as all good linux users do...now I'm afraid I've totally wonked it. I'll put all pertinent alsa-info up here for anyone who will help!! http://www.alsa-project.org/db/?f=58850a92a9ea0edbf28a8d014a875cdbd81b621d I guess I'm bright enough to really mess it up, but dumb enough not to have a clue how to fix it. It has NEVER been able to recognize a headphone plugged in the jack automatically. I hope someone knows... -- Soli Deo Gloria! Matt Hudson "There is a kind of happiness and wonder that makes you serious." C.S. Lewis

2 1

[alsa-devel] [PATCH 1/3] ASoC: multi-component - Fix S3C compile error
by Seungwhan Youn 15 Jul '10

15 Jul '10

This patch fix compile error on using S3C I2S. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index ad92e10..5017e31 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -56,11 +56,11 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, switch (clk_id) { case S3C64XX_CLKSRC_PCLK: - iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX; + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_MUX: - iismod |= S3C64XX_IISMOD_IMS_SYSMUX; + iismod |= S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_CDCLK: @@ -90,7 +90,7 @@ struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(dai); u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) + if (iismod & S3C2412_IISMOD_IMS_SYSMUX) return i2s->iis_cclk; else return i2s->iis_pclk; -- 1.6.2.5

4 9

Re: [alsa-devel] [PATCH 1/2] Kconfig: SND_AC97_POWER_SAVE description improvement
by Jiri Kosina 15 Jul '10

15 Jul '10

On Wed, 14 Jul 2010, Michael Witten wrote: > The description has been expanded to explain the time-out > value provided by the power_save module parameter. Michael, thanks a lot for the patch. I'd rather let it go through Takashi still anyway (added to CC). Thanks. > > Signed-off-by: Michael Witten <mfwitten(a)gmail.com> > --- > sound/drivers/Kconfig | 24 +++++++++++++++++++++--- > 1 files changed, 21 insertions(+), 3 deletions(-) > > diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig > index 84714a6..3264600 100644 > --- a/sound/drivers/Kconfig > +++ b/sound/drivers/Kconfig > @@ -170,9 +170,25 @@ config SND_AC97_POWER_SAVE > AC97 codecs. In this mode, the power-mode is dynamically > controlled at each open/close. > > - The mode is activated by passing power_save=1 option to > - snd-ac97-codec driver. You can toggle it dynamically over > - sysfs, too. > + The mode is activated by passing 'power_save=X' to the > + snd-ac97-codec driver module, where 'X' is the time-out > + value, a nonnegative integer that specifies how many > + seconds of idle time the driver must count before it may > + put the AC97 into power-save mode; a value of 0 (zero) > + disables the use of this power-save mode. > + > + After the snd-ac97-codec driver module has been loaded, > + the 'power_save' parameter can be set via sysfs as follows: > + > + echo 10 > /sys/module/snd_ac97_codec/parameters/power_save > + > + In this case, the time-out is set to 10 seconds; setting > + the time-out to 1 second (the minimum activation value) > + isn't recommended because many applications try to reopen > + the device frequently. A value of 10 seconds would be a > + good choice for normal operations. > + > + See Documentation/sound/alsa/powersave.txt for more details. > > config SND_AC97_POWER_SAVE_DEFAULT > int "Default time-out for AC97 power-save mode" > @@ -182,4 +198,6 @@ config SND_AC97_POWER_SAVE_DEFAULT > The default time-out value in seconds for AC97 automatic > power-save mode. 0 means to disable the power-save mode. > > + See SND_AC97_POWER_SAVE for more details. > + > endif # SND_DRIVERS > -- > 1.7.2.rc1.9.g5b527 > -- Jiri Kosina SUSE Labs, Novell Inc.

3 3

[alsa-devel] [RFC][PATCH 0/3] ASoC: FSI - codecs settings
by Kuninori Morimoto 15 Jul '10

15 Jul '10

Dear Mark, Liam These patches are FSI - codecs settings Kuninori Morimoto (3): ASoC: ak4642: make sure snd_soc_codec value for ak4642 ASoC: da7210: make sure snd_soc_codec value for da7210 ASoC: Add fsi-codec support But I'm not sure this type of fix is correct or not. So, please check and teach me. Now, FSI driver has 2 connections like below. fsi - ak4642 fsi - da7210 In future, I will send "fsi - HDMI" pair. On AP4 board, I would like to use "fsi - ak4642" and "fsi - HDMI" in same time. To use it, I guess I should modify snd_soc_card :: dai_link correct ? I expected this is correct, I tried to use "fsi - ak4642" and "fsi - da7210" in same time for test. But when I used "soc_codec_dev_ak4642" for snd_soc_device :: codec_dev, da7210 didn't work. because i2c address of ak4642 is used for da7210. Same things happen on "soc_codec_dev_da7210" - ak4642. My patches solve above issue for me. But I'm not sure this is correct way. Best regards -- Kuninori Morimoto

2 6

[alsa-devel] controlling access to the sound card
by Jason Butler 14 Jul '10

14 Jul '10

I have three applications that want to use the sound card, two audio stream players, and a voip phone. Is it possible to use alsa so that if a call comes in on the phone alsa will disconnect the audio players, give exclusive access to the voip phone, and then when the phone is done reconnect the audio players to the sound card. Can this be done using alsa and if so how? Could it be done using just the alsa command line tools and config files and if so how?

1 0

[alsa-devel] [PATCH] wm8727: add a missing return in wm8727_platform_probe
by Axel Lin 14 Jul '10

14 Jul '10

otherwise the error path will always be executed. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8727.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 1072621..9d1df26 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -127,6 +127,8 @@ static __devinit int wm8727_platform_probe(struct platform_device *pdev) goto err_codec; } + return 0; + err_codec: snd_soc_unregister_codec(codec); err: -- 1.7.0.4

3 2

Re: [alsa-devel] [PATCH 4/4] ASoC: DaVinci: pcm, fix underrun by using sram
by troy.kisky＠boundarydevices.com 14 Jul '10

14 Jul '10

BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; } On Tue 13/07/10 6:00 AM , "Nori, Sekhar" nsekhar(a)ti.com sent: Hi Troy, On Tue, Nov 17, 2009 at 05:22:34, Troy Kisky wrote: > Fix underruns by using dma to copy 1st to sram > in a ping/pong buffer style and then copying from > the sram to the ASP. This also has the advantage > of tolerating very long interrupt latency on dma > completion. > > Signed-off-by: Troy Kisky [...] > +static int request_ping_pong(struct snd_pcm_substream *substream, > + struct davinci_runtime_data *prtd, > + struct snd_dma_buffer *iram_dma) > +{ > + dma_addr_t asp_src_ping; > + dma_addr_t asp_dst_ping; > + int link; > + struct davinci_pcm_dma_params *dma_data = prtd->params; > + > + /* Request ram master channel */ > + link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, > + davinci_pcm_dma_irq, substream, > + EVENTQ_1); What is the reason for choosing EVENTQ_1 for this channel? EVENTQ_0 is already being used for ASP channel. I imagine it will be much easier to tune the queue usage in the system if all of audio data was using the same queue. I am working on a patch which lets platform specify the event queues for audio DMA. I am not sure if I really need to make a provision for two different queues to be specified - that's why I ask. Thanks, Sekhar The reason is so that the IRAM data can be fetched and used while EVENTQ_1 fetches the next buffer of data from sdram into IRAM. Troy

3 2

[alsa-devel] [PATCH] ASoC: Handle read failures in codec_reg
by Mark Brown 14 Jul '10

14 Jul '10

When a device is powered down volatile registers can't be read so attempts to display codec_reg will show error values, and obviously it is also possible for there to be hardware errors too. Check for errors from reads and display them more clearly when formatting codec_reg. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 22 +++++++++++++++++----- 1 files changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96f9a8e..844ae82 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static int run_delayed_work(struct delayed_work *dwork) /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { - int i, step = 1, count = 0; + int ret, i, step = 1, count = 0; if (!codec->reg_cache_size) return 0; @@ -101,12 +101,24 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (count >= PAGE_SIZE - 1) break; - if (codec->display_register) + if (codec->display_register) { count += codec->display_register(codec, buf + count, PAGE_SIZE - count, i); - else - count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); + } else { + /* If the read fails it's almost certainly due to + * the register being volatile and the device being + * powered off. + */ + ret = codec->read(codec, i); + if (ret >= 0) + count += snprintf(buf + count, + PAGE_SIZE - count, + "%4x", ret); + else + count += snprintf(buf + count, + PAGE_SIZE - count, + "<no data: %d>", ret); + } if (count >= PAGE_SIZE - 1) break; -- 1.7.1

2 1

[alsa-devel] [PATCH] Asoc Davinci Voicecodec: Added support based on copy_from_user instead of DMA
by Raffaele Recalcati 14 Jul '10

14 Jul '10

From: Davide Bonfanti <davide.bonfanti(a)bticino.it> Since DM365 has the same DMA event for McBSP and Voicecodec this two peripherals cannot be used at the same time. This driver implements Voicecodec without the use of a DMA but with a copy_from_user. There's a buffer of BUF_SIZE (tested with 2KB) bytes in the driver that is filled with davinci_pcm_copy. When pcm is running, a TIMER interrupt is activated each 1.36ms in order to fill HW FIFO (dm_vc_irq). BUG: It happens sometimes that the peripheral stops working so there's a trap. This driver was tested with timer1 and timer4. The measures made using a GPIO and scope give a time for copy_from_user about 8/10us (triggered by alsa layer) while TIMER interrupt takes 1.5us. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and tested on bmx board. Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> --- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-pcm-copyfromuser.c | 333 ++++++++++++++++++++++++++ sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/soc-core.c | 31 ++- 4 files changed, 354 insertions(+), 12 deletions(-) create mode 100644 sound/soc/davinci/davinci-pcm-copyfromuser.c diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a93679d..7d6a9a1 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,5 +1,6 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o +snd-soc-davinci-objs += davinci-pcm-copyfromuser.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o diff --git a/sound/soc/davinci/davinci-pcm-copyfromuser.c b/sound/soc/davinci/davinci-pcm-copyfromuser.c new file mode 100644 index 0000000..df37873 --- /dev/null +++ b/sound/soc/davinci/davinci-pcm-copyfromuser.c @@ -0,0 +1,333 @@ +/* + * + * Copyright (C) 2010 Bticino S.p.a + * Author: Davide Bonfanti <davide.bonfanti(a)bticino.it> + * + * Contributors: + * Raffaele Recalcati <raffaele.recalcati(a)bticino.it> + * + * 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. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; without even the implied warranty + * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/kernel.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/gpio.h> + +#include "davinci-pcm.h" + +/* Timer register offsets */ +#define PID12 0x00 +#define TIM12 0x10 +#define TIM34 0x14 +#define PRD12 0x18 +#define PRD34 0x1c +#define TCR 0x20 +#define TGCR 0x24 + +/* Timer register bitfields */ +#define TCR_ENAMODE_DISABLE 0x0 +#define TCR_ENAMODE_ONESHOT 0x1 +#define TCR_ENAMODE_PERIODIC 0x2 +#define TCR_ENAMODE_MASK 0x3 + +#define TGCR_TIMMODE_SHIFT 2 +#define TGCR_TIMMODE_64BIT_GP 0x0 +#define TGCR_TIMMODE_32BIT_UNCHAINED 0x1 +#define TGCR_TIMMODE_64BIT_WDOG 0x2 +#define TGCR_TIMMODE_32BIT_CHAINED 0x3 + +#define TGCR_TIM12RS_SHIFT 0 +#define TGCR_TIM34RS_SHIFT 1 +#define TGCR_RESET 0x0 +#define TGCR_UNRESET 0x1 +#define TGCR_RESET_MASK 0x3 + +#define TIMER1_BASE 0x01C21800 +#define TIMER4_BASE 0x01C23800 + +/* choose the timer to use */ +#define TIMER_BASE TIMER4_BASE +/* driver buffer dimension */ +#define BUF_SIZE 2048 + +#if (TIMER_BASE == TIMER1_BASE) +#define TINT IRQ_TINT1_TINT12 +#define TIMER "timer1" +#elif (TIMER_BASE == TIMER4_BASE) +#define TINT IRQ_PWMINT2 +#define TIMER "timer4" +#endif + +int pointer_sub; +u16 local_buffer[BUF_SIZE/2]; + +static struct snd_pcm_hardware pcm_hardware_playback = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000), + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = BUF_SIZE, + .period_bytes_min = 512, + .period_bytes_max = 512, + .periods_min = BUF_SIZE / 512, + .periods_max = BUF_SIZE / 512, + .fifo_size = 0, +}; + +/* + * How this driver works... + * + * Since DM365 have the same DMA event for I2S and Voicecodec this two + * peripheralscannot be used at the same time. + * This driver implements voicecodec without the use of a DMA but with + * a copy_from_user. + * There's a buffer of BUF_SIZE bytes in the driver that is filled with + * davinci_pcm_copy. + * When pcm is running, a TIMER interrupt is activated each 1.36ms in + * order to fill HW FIFO (dm_vc_irq). + * It happens that the peripheral stop working so there's a trap... + * Driver was proved with timer1 and timer4. if timer2 is used the device + * resets during power-up. Timer3 have a structure a bit different and, + * in fact, doesn't work. + * Measures using a GPIO and scope give a time for copy_from_user about + * 8/10us while interrupt takes 1.5us + */ + +static snd_pcm_uframes_t +davinci_pcm_pointer(struct snd_pcm_substream *substream) +{ + return pointer_sub; +} + +static int davinci_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd; + struct snd_pcm_hardware *ppcm; + int ret = 0; + struct davinci_pcm_dma_params *pa; + struct davinci_pcm_dma_params *params; + struct clk *clk; + + pointer_sub = 0; + +/* + * ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + * &pcm_hardware_playback : &pcm_hardware_capture; + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ppcm = &pcm_hardware_playback; + else + return -ENODEV; + + snd_soc_set_runtime_hwparams(substream, ppcm); + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + __raw_writel(0x0, IO_ADDRESS(0x01D0C008)); + __raw_writel(0x7400, IO_ADDRESS(0x01D0C004)); + clk = clk_get(NULL, TIMER); + if (!IS_ERR(clk)) + clk_enable(clk); + return 0; +} + +static int davinci_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct clk *clk; + + clk = clk_get(NULL, TIMER); + if (!IS_ERR(clk)) + clk_disable(clk); + return 0; +} + +static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int davinci_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + if (copy_from_user(local_buffer + hwoff, buf, + frames_to_bytes(runtime, frames))) { + printk(KERN_ERR "ERROR COPY_FROM_USER\n"); + return -EFAULT; + } + return 0; +} + +static struct snd_pcm_ops davinci_pcm_ops = { + .open = davinci_pcm_open, + .close = davinci_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = davinci_pcm_hw_params, + .hw_free = davinci_pcm_hw_free, + .pointer = davinci_pcm_pointer, + .copy = davinci_pcm_copy, +}; + +static u64 davinci_pcm_dmamask = 0xffffffff; + +static irqreturn_t dm_vc_irq(int irq, void *sbst) +{ + struct snd_pcm_substream *substream = + (struct snd_pcm_substream *) sbst; + int fifo, diff, per_size, buf_size; + static int last_ptr; + + if (substream->runtime && substream->runtime->status && + snd_pcm_running(substream)) { + fifo = __raw_readl(IO_ADDRESS(0x01D0C028)); + fifo = (fifo & 0x1f00) >> 8; + if (fifo == 15) { + /* peripheral blocked! restart */ + __raw_writel(0, IO_ADDRESS(0x01D0C004)); + __raw_writel(0x5400, IO_ADDRESS(0x01D0C004)); + } + buf_size = substream->runtime->buffer_size; + per_size = substream->runtime->period_size; + for (; fifo < 0x10; fifo++) { + __raw_writew(local_buffer[pointer_sub++], + IO_ADDRESS(0x01D0C024)); + pointer_sub %= buf_size; + do { + diff = __raw_readl(IO_ADDRESS(0x01D0C00C)); + } while (!(diff & 0x8)); + } + if (last_ptr >= pointer_sub) + diff = buf_size + pointer_sub - last_ptr; + else + diff = pointer_sub - last_ptr; + if (diff >= per_size) { + snd_pcm_period_elapsed(substream); + last_ptr += per_size; + if (last_ptr >= buf_size) + last_ptr -= buf_size; + } + } else + last_ptr = 0; + return IRQ_HANDLED; +} + +static int davinci_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int tcr, tgcr; + struct snd_dma_buffer *buf; + struct snd_pcm_substream *substream; + struct clk *clk; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &davinci_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + buf = &substream->dma_buffer; + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->bytes = pcm_hardware_playback.buffer_bytes_max; + } + clk = clk_get(NULL, TIMER); + if (IS_ERR(clk)) + printk(KERN_ERR "ERROR getting timer\n"); + + /* Disabled, Internal clock source */ + __raw_writel(0, IO_ADDRESS(TIMER_BASE + TCR)); + + /* reset both timers, no pre-scaler for timer34 */ + tgcr = 0; + __raw_writel(tgcr, IO_ADDRESS(TIMER_BASE + TGCR)); + + /* Set both timers to unchained 32-bit */ + tgcr = TGCR_TIMMODE_32BIT_UNCHAINED << TGCR_TIMMODE_SHIFT; + __raw_writel(tgcr, IO_ADDRESS(TIMER_BASE + TGCR)); + + /* Unreset timers */ + tgcr |= (TGCR_UNRESET << TGCR_TIM12RS_SHIFT); + __raw_writel(tgcr, IO_ADDRESS(TIMER_BASE + TGCR)); + + /* Init both counters to zero */ + __raw_writel(0, IO_ADDRESS(TIMER_BASE + TIM12)); + tcr = __raw_readl(IO_ADDRESS(TIMER_BASE + TCR)); + + /* disable timer */ + tcr &= ~(TCR_ENAMODE_MASK << 6); + __raw_writel(tcr, IO_ADDRESS(TIMER_BASE + TCR)); + + /* reset counter to zero, set new period */ + __raw_writel(0, IO_ADDRESS(TIMER_BASE + TIM12)); + __raw_writel(0x8000, IO_ADDRESS(TIMER_BASE + PRD12)); + + /* Set enable mode */ + tcr |= TCR_ENAMODE_PERIODIC << 6; + __raw_writel(tcr, IO_ADDRESS(TIMER_BASE + TCR)); + +#if (TIMER_BASE == TIMER4_BASE) + __raw_writel(__raw_readl(IO_ADDRESS(0x01c40018)) | + 0x1000, IO_ADDRESS(0x01c40018)); +#endif + if (request_irq(TINT, dm_vc_irq, IRQF_TIMER, "VOICE-TOUT", substream)) + printk(KERN_ERR "%s ERROR requesting Interrupt\n", __func__); + return 0; +} + +struct snd_soc_platform davinci_soc_platform_copy = { + .name = "davinci-audio-copy", + .pcm_ops = &davinci_pcm_ops, + .pcm_new = davinci_pcm_new, +}; EXPORT_SYMBOL_GPL(davinci_soc_platform_copy); + +static int __init davinci_soc_copy_platform_init(void) +{ + return snd_soc_register_platform(&davinci_soc_platform_copy); +} +module_init(davinci_soc_copy_platform_init); + +static void __exit davinci_soc_copy_platform_exit(void) +{ + snd_soc_unregister_platform(&davinci_soc_platform_copy); +} +module_exit(davinci_soc_copy_platform_exit); + +MODULE_AUTHOR("bticino s.p.a."); +MODULE_DESCRIPTION("TI DAVINCI PCM copy_from_user module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0764944..cb7c2aa 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -29,5 +29,6 @@ struct davinci_pcm_dma_params { extern struct snd_soc_platform davinci_soc_platform; +extern struct snd_soc_platform davinci_soc_platform_copy; #endif diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad7f952..696e8b3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -801,7 +801,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } return 0; } - +#if 0 /* ASoC PCM operations */ static struct snd_pcm_ops soc_pcm_ops = { .open = soc_pcm_open, @@ -811,7 +811,7 @@ static struct snd_pcm_ops soc_pcm_ops = { .prepare = soc_pcm_prepare, .trigger = soc_pcm_trigger, }; - +#endif #ifdef CONFIG_PM /* powers down audio subsystem for suspend */ static int soc_suspend(struct device *dev) @@ -1306,6 +1306,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + struct snd_pcm_ops *soc_pcm_ops; rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); if (rtd == NULL) @@ -1335,19 +1336,25 @@ static int soc_new_pcm(struct snd_soc_device *socdev, dai_link->pcm = pcm; pcm->private_data = rtd; - soc_pcm_ops.mmap = platform->pcm_ops->mmap; - soc_pcm_ops.pointer = platform->pcm_ops->pointer; - soc_pcm_ops.ioctl = platform->pcm_ops->ioctl; - soc_pcm_ops.copy = platform->pcm_ops->copy; - soc_pcm_ops.silence = platform->pcm_ops->silence; - soc_pcm_ops.ack = platform->pcm_ops->ack; - soc_pcm_ops.page = platform->pcm_ops->page; + soc_pcm_ops = kmalloc(sizeof(struct snd_pcm_ops), GFP_KERNEL); + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_codec_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->mmap = platform->pcm_ops->mmap; + soc_pcm_ops->pointer = platform->pcm_ops->pointer; + soc_pcm_ops->ioctl = platform->pcm_ops->ioctl; + soc_pcm_ops->copy = platform->pcm_ops->copy; + soc_pcm_ops->silence = platform->pcm_ops->silence; + soc_pcm_ops->ack = platform->pcm_ops->ack; + soc_pcm_ops->page = platform->pcm_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); ret = platform->pcm_new(codec->card, codec_dai, pcm); if (ret < 0) { -- 1.7.0.4

2 1

Re: [alsa-devel] [LAD] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 14 Jul '10

14 Jul '10

On Tue, Jul 13, 2010 at 1:19 AM, <fons(a)kokkinizita.net> wrote: >On Mon, Jul 12, 2010 at 06:18:29PM -0700, Niels Mayer wrote: >> Speaking of making my brain hurt -- I really would not have done the >> meters as done in envy24control. I really hate the concept of >> emulating the look of real gear down to the level that you have to >> "quantize" to individual LED's... > > That is indeed quite absurd. Yes, the meters in envy24control -- http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=envy24control/… -- are absurd! What's even more absurd, is to have the benefits of the ice1712's hardware metering squandered by drawing thousands of rectangles per second (each individual "LED" on each meter is a separate x draw-rectangle call). The existing envy24control meters are an abomination of inefficiency! Especially when used with a remote X display. As a second patch (coming soon), I've rewritten the meters in a more sensible fashion, drawing a single rectangle to represent the instantaneous level (2-3 X-primitive draws per meter total and one blit, versus hundreds of draws and a blit per value change per meter in the original code). I'm also being a little more careful about not needing to redraw the entire meter for each change -- using the monotonicity of the audio signal being represented to optimize the way you draw it; not redrawing if nothing has changed, etc. And since the hardware metering already quantizes to 256 levels -- there's no point in re-quantizing those to individual "virtual LEDs" as is done w/ the current envy24control. The result so far are promising: Here's 'top' output (*) using the new envy24control. Note the load is lower, and 'envy24control' and 'X' both use much less CPU. In fact running this meter lowers the system load. whereas running the old envy24control raises it (and makes the fans run faster!). top - 21:30:13 up 11:30, 9 users, load average: 0.02, 0.11, 0.06 2664 root 20 0 544m 134m 34m S 2.6 3.4 26:03.51 X 9404 npm 20 0 192m 9340 6908 S 0.7 0.2 0:01.72 envy24control The new metering gives 2.6% CPU for X, and 0.7% CPU for the new-metered envy24control; Versus 6.3% CPU for X and 2.7% for the old metering in envy24control (the one from my first patch): top - 21:25:57 up 11:25, 9 users, load average: 0.10, 0.09, 0.03 2664 root 20 0 544m 134m 34m S 6.3 3.4 25:50.67 X 9398 npm 20 0 192m 9368 6940 S 2.7 0.2 0:16.81 envy24control Now granted this is not a scientific test, and "top" is not exactly the best way to gauge performance. But it does suggest that the computational complexity caused by the bad metering algorithm directly translates to higher-power consumption, higher loads, and poorer performance. Compared to the existing envy24control metering -- It would probably me more efficient to actually compute RMS levels off the audio stream -- or even hook up jkmeter through jack -- instead of using the hardware metering data through these computationally wasteful level meters. -- Niels http://nielsmayer.com

2 2

[alsa-devel] [PATCH] wm8988: remove a redundant snd_soc_unregister_codec call in wm8988_register
by Axel Lin 14 Jul '10

14 Jul '10

snd_soc_unregister_codec is called twice if snd_soc_register_dai fail. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8988.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 0417dae..19ad590 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -885,7 +885,6 @@ static int wm8988_register(struct wm8988_priv *wm8988, ret = snd_soc_register_dai(&wm8988_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); goto err_codec; } -- 1.7.0.4

1 0

[alsa-devel] ASoC updates for-2.6.36
by Mark Brown 13 Jul '10

13 Jul '10

The following changes since commit 66b47fdb851924956b6e4696fb43a3496ae2c462: ASoC: Implement WM8994 OPCLK support (2010-07-09 08:50:12 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Arnaud Patard (2): ASoC: Fix kirkwood i2s mono playback ASoC: kirkwood-i2s: Handle mute/unmute playback/record Joe Perches (2): ASoC: Remove unnecessary casts of private_data ASoC: Remove unnecessary casts of private_data Kuninori Morimoto (8): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM ASoC: fsi: modify format area definition on flags ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode Manuel Lauss (1): ASoC: au1x: fix section mismatch in psc-i2s.c Mark Brown (1): Merge branch 'for-2.6.35' into for-2.6.36 include/sound/sh_fsi.h | 44 +++++++- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 6 +- sound/soc/blackfin/bf5xx-tdm.c | 6 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 27 ++++-- sound/soc/kirkwood/kirkwood.h | 3 + sound/soc/sh/fsi.c | 213 ++++++++++++++++++++++++++----------- 8 files changed, 219 insertions(+), 84 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 13 Jul '10

13 Jul '10

The following changes since commit 1c5474a65bf15a4cb162dfff86d6d0b5a08a740c: Linux 2.6.35-rc5 (2010-07-12 14:55:33 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Kuninori Morimoto (2): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM Small, targetted fixes for a specific platform. sound/soc/sh/fsi.c | 27 +++++++++++---------------- 1 files changed, 11 insertions(+), 16 deletions(-)

2 1

[alsa-devel] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"
by Niels Mayer 13 Jul '10

13 Jul '10

I've been long annoyed by alsa-tools' envy24control (*) lack of peak-level indication in it's metering, and the non-implementation of its "Reset Peaks" button: https://bugzilla.redhat.com/show_bug.cgi?id=602903 I've fixed that annoyance. Now, a small white line appears above the highest level on each meter, and can be reset via "Reset Peaks" button. Original source: http://git.alsa-project.org/?p=alsa-tools.git;a=blob_plain;f=envy24control/… Patch: http://nielsmayer.com/npm/602903.patch To run this version of envy24control grab the most recent stable release ( ftp://ftp.alsa-project.org/pub/tools/alsa-tools-1.0.23.tar.bz2 ) or git pull from trunk of the "alsa-tools" project ( http://alsa-project.org/main/index.php/GIT_Server ). After unpacking and assuming you've got the patch in ~/602903.patch do: >> cd alsa-tools-1.0.23 >> cat ~/602903.patch | patch -p1 it should give message "patching file envy24control/levelmeters.c" ... follow the directions to compile alsa-tools.... Can whoever's in charge of these things commit this change "upstream" and get it to propagate to the distros? Alternately, give me commit access and I'll do it myself, if people agree this patch is worth having. Next steps: (1) Think about building-in meter ballistics from Fons Adriaensen's jkmeter ( http://www.kokkinizita.net/linuxaudio/downloads/jkmeter-0.4.0.tar.bz2 ). Which helps implent Bob Katz' "K System" ( http://www.digido.com/level-practices-part-2-includes-the-k-system.html ). Although it probably most makes sense to implement such metering on the main "digital mixer" meter of envy24control, if it's not too inefficient, I'd want it available on all meters, despite the warning from Fons ( http://old.nabble.com/First-release-of-jkmeter-td18798950.html ). > This is the type of meter you want for live recording, > mixing and mastering. It probably makes no sense to > use it on all tracks of a DAW, where keeping digital > level within limits is the main purpose of metering. (2) Fix https://bugzilla.redhat.com/show_bug.cgi?id=602900 which causes 'envy24control' to crash when given "-D" argument. and behave incorrectly when given a correct non-numeric name for an audio device: e.g. >> envy24control --card M66 >> invalid card type (driver is USB-Audio) >> envy24control --card M66 --device M66 >> Segmentation fault (core dumped) (3) Apply more reasonable defaults and automatically size to correct dimensions. I use, for example: >> envy24control --card 2 --outputs 4 --input 4 --pcm_output 8 --view_spdif_playback --midichannel 1 --midienhanced --window_width 1275 --tall_eq_mixer_heights 1 & (4) Add midi output, in addition to midi input, which would monitor the hardware peak metering from the ice1712 ("amixer -c M66 cget iface=PCM,name='Multi Track Peak',numid=45") and output a MIDI note-on on the corresponding MIDI channel whenever a specified threshold value was exceeded. This could be used to implement "automatic gain control" in association with the midi-controllable ADC gain provided by envy24control. Niels http://nielsmayer.com PS: (*) http://alsa.opensrc.org/index.php/Envy24control Cards supported by this fix: * M Audio Delta 1010 * M Audio Delta 1010LT * M Audio Delta DiO 2496 * M Audio Delta 66 * M Audio Delta 44 * M Audio Delta 410 * M Audio Audiophile 2496 * TerraTec EWS 88MT * TerraTec EWS 88D * TerraTec EWX 24/96 * TerraTec DMX 6Fire * TerraTec Phase 88 * Hoontech SoundTrack DSP 24 * Hoontech SoundTrack DSP 24 Value * Hoontech SoundTrack DSP 24 Media 7.1 * Event Electronics, EZ8 * Digigram VX442 * Lionstracs, Mediastaton * Terrasoniq TS 88

3 2

[alsa-devel] [PATCH] ASoC: au1x: fix section mismatch in psc-i2s.c
by Manuel Lauss 13 Jul '10

13 Jul '10

Annotate platform probe callback with __devinit instead of plain __init. Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- sound/soc/au1x/psc-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 495be6e..24454c9 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -300,7 +300,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = { }; EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev) +static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) { struct resource *r; unsigned long sel; -- 1.7.1.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 2

[alsa-devel] [PATCH 34/36] sound/soc/blackfin: Remove unnecessary casts of private_data
by Joe Perches 13 Jul '10

13 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/blackfin/bf5xx-ac97.c | 6 ++---- sound/soc/blackfin/bf5xx-tdm.c | 6 ++---- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 523b7fc..c0eba51 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -255,8 +255,7 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); #ifdef CONFIG_PM static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) @@ -271,8 +270,7 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) static int bf5xx_ac97_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 4b36012..24c1426 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -210,8 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; if (!dai->active) return 0; @@ -225,8 +224,7 @@ static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) static int bf5xx_tdm_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; + struct sport_device *sport = dai->private_data; if (!dai->active) return 0; -- 1.7.1.337.g6068.dirty

3 2

[alsa-devel] [patch 0/2] Kirkwood audio fixes
by arnaud.patard＠rtp-net.org 13 Jul '10

13 Jul '10

Hi, While working on the t5325 audio support, I found two small buglets in the kirkwood ASoC support. The first one is aiming at fixing mono audio playback and should fix mono playback for all platforms. The second is adding support for muting/unmuting kirkwood audio. It's mainly usefull for the t5325 which has a bootloader playing with this (hardware default is not muted). Arnaud

3 5

[alsa-devel] [PATCH 1/6 v3] ASoC: fsi: modify format area definition on flags
by Kuninori Morimoto 13 Jul '10

13 Jul '10

There is no necessity that each bit in this area has the meaning. This patch modify it to sequence number Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 o don't use << 0 shift 2/6 - 6/6 doesn't depend on 1/6 patch include/sound/sh_fsi.h | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index c022736..3ecdb96 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -55,12 +55,12 @@ #define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) #define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) -#define SH_FSI_FMT_MONO (1 << 0) -#define SH_FSI_FMT_MONO_DELAY (1 << 1) -#define SH_FSI_FMT_PCM (1 << 2) -#define SH_FSI_FMT_I2S (1 << 3) -#define SH_FSI_FMT_TDM (1 << 4) -#define SH_FSI_FMT_TDM_DELAY (1 << 5) +#define SH_FSI_FMT_MONO 0 +#define SH_FSI_FMT_MONO_DELAY 1 +#define SH_FSI_FMT_PCM 2 +#define SH_FSI_FMT_I2S 3 +#define SH_FSI_FMT_TDM 4 +#define SH_FSI_FMT_TDM_DELAY 5 #define SH_FSI_IFMT_TDM_CH(x) \ (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) -- 1.7.0.4

3 2

[alsa-devel] [PATCH 0/2] ASoC: fsi: fixup patches
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These are v2 bug fix patches for FSI driver. Kuninori Morimoto (2): ASoC: fsi: fixup clock inversion operation ASoC: fsi: fixup wrong value setting order of TDM v1 -> v2 It is bug fix only Best regards -- Kuninori Morimoto

3 4

[alsa-devel] [PATCH 1/4] ASoC: DaVinci: remove requirement that dma_params is 1st in structure
by Troy Kisky 13 Jul '10

13 Jul '10

Remove requirement that dma_params is 1st in the structures davinci_audio_dev and davinci_mcbsp_dev. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 6 +----- sound/soc/davinci/davinci-mcasp.c | 1 + sound/soc/davinci/davinci-mcasp.h | 5 ----- sound/soc/davinci/davinci-pcm.c | 7 ++++--- 4 files changed, 6 insertions(+), 13 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae7070..9e69a4e 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -98,11 +98,6 @@ enum { }; struct davinci_mcbsp_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -547,6 +542,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; + davinci_i2s_dai.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a..5023915 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -907,6 +907,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9d179cc..582c924 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,11 +39,6 @@ enum { }; struct davinci_audio_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index c73a915..c2c2e85 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -238,10 +238,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; - struct davinci_pcm_dma_params *params = &pa[substream->stream]; - if (!params) + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *params; + if (!pa) return -ENODEV; + params = &pa[substream->stream]; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ -- 1.5.6.3

3 5

[alsa-devel] [PATCH] ASoC: TWL4030: Fix for digital loopback gain range
by Peter Ujfalusi 13 Jul '10

13 Jul '10

When the gain is configured using dB value it was not possible to use -24dB since the loopback got muted instead of -24dB. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 8d36bfa..d7f0048 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -540,10 +540,11 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); -/* Digital bypass gain, 0 mutes the bypass */ +/* Digital bypass gain, mute instead of -30dB */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { - TLV_DB_RANGE_HEAD(2), - 0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1), + TLV_DB_RANGE_HEAD(3), + 0, 1, TLV_DB_SCALE_ITEM(-3000, 600, 1), + 2, 3, TLV_DB_SCALE_ITEM(-2400, 0, 0), 4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0), }; -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: TWL4030: DAC power optimization
by Peter Ujfalusi 13 Jul '10

13 Jul '10

Restructure the DAPM connections in order to enable only the needed DAC (out of four in twl4030 series). I need to keep the 'AIF Enable' supply connected to the L2/R2 digital path, since the digital loopback needs AIF and APLL running. If no valid route available, than none of the DAC will be powered, but the AIF and APLL is going to be enabled. Furthermore, if only one audio path have valid route, than only the corresponding DAC will be powered. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 15 +++++++++------ 1 files changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index d7f0048..6fd6d0b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1460,8 +1460,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Supply for the digital part (APLL) */ {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, - {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, - {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"DAC Left1", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"DAC Left2", NULL, "AIF Enable"}, + {"DAC Right1", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, @@ -1532,10 +1535,10 @@ static const struct snd_soc_dapm_route intercon[] = { /* outputs */ /* Must be always connected (for AIF and APLL) */ - {"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"}, - {"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"}, + {"Virtual HiFi OUT", NULL, "DAC Left1"}, + {"Virtual HiFi OUT", NULL, "DAC Right1"}, + {"Virtual HiFi OUT", NULL, "DAC Left2"}, + {"Virtual HiFi OUT", NULL, "DAC Right2"}, /* Must be always connected (for APLL) */ {"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"}, /* Physical outputs */ -- 1.7.1.1

3 2

[alsa-devel] [PATCH 0/8] ASoC: fsi: fixup patches
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These patches are fixup for FSI driver Kuninori Morimoto (8): ASoC: fsi: modify format area definition on flags ASoC: fsi: fixup clock inversion operation ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: modify wrong value setting order of TDM ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode Some of these are not so important for now. But it is used from "FSI - HDMI" patches which I will send in future

2 13

[alsa-devel] issue with TWL4030
by Grazvydas Ignotas 13 Jul '10

13 Jul '10

Hi, there seems to be a problem with TWL4030 hw_param programming, which is easy to reproduce over OSS emulation with this simple program: int oss_dev = open("/dev/dsp", O_WRONLY); int stereo = 1, bits = 16, rate = 44100; ioctl(oss_dev, SNDCTL_DSP_SETFMT, &bits); ioctl(oss_dev, SNDCTL_DSP_SPEED, &rate); ioctl(oss_dev, SNDCTL_DSP_STEREO, &stereo); write(oss_dev, somebuff, sizeof(somebuff)); The above program ends up playing at 8kHz, instead of expected 44kHz. I did some call tracing and got this: [ 1092.246032] soc_pcm_hw_params, rate 8000 [ 1092.250122] twl4030_hw_params [ 1092.253234] set rate 8000 [ 1092.256103] configured 1 [ 1092.259368] soc_pcm_hw_params, rate 8000 [ 1092.263427] twl4030_hw_params [ 1092.266845] soc_pcm_hw_params, rate 44100 [ 1092.270965] twl4030_hw_params [ 1094.459106] twl4030_shutdown [ 1094.462036] configured 0 So it looks like twl4030->configured is the culprit of this, removing it solves the problem, but I guess that will break something else. Any ideas how to properly fix this? Gražvydas

6 11

[alsa-devel] [PATCH 0/6] ASoC: fsi: modify patch
by Kuninori Morimoto 13 Jul '10

13 Jul '10

Dear Mark These are v2 patches for FSI driver almost all are same as v1 Kuninori Morimoto (6): ASoC: fsi: modify format area definition on flags ASoC: fsi: remove un-used variable on fsi_dai_startup ASoC: fsi: remove noisy CR_FMT macro ASoC: fsi: Change struct fsi_regs to fsi_core ASoC: fsi: Add pr_err for noticing unsupported access ASoC: fsi: Fixup for master mode v1 -> v2 o modiry straight start number as 0 o based on bug fix patches Best regards -- Kuninori Morimoto

1 6

[alsa-devel] [PATCH 0/3] further BKL removal
by Arnd Bergmann 13 Jul '10

13 Jul '10

With the ioctl, block, tty, vfs and llseek series on their way into linux-next, these three patches are attacking the hardest remaining issues. If we get these ready for 2.6.36, the kernel should be almost usable with the BKL disabled. In all three cases, I'm cheating a bit, because the BKL still remains lurking in the dark corners of the three subsystems (i810/i830 for drm, OSS for sound and lockd for fs/locks.c). Arnd Cc: Christoph Hellwig <hch(a)lst.de> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Frederic Weisbecker <fweisbec(a)gmail.com> Cc: Ingo Molnar <mingo(a)redhat.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: "J. Bruce Fields" <bfields(a)fieldses.org> Cc: John Kacur <jkacur(a)redhat.com> Cc: Matthew Wilcox <willy(a)linux.intel.com> Cc: Miklos Szeredi <mszeredi(a)suse.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Trond Myklebust <trond.myklebust(a)fys.uio.no> Cc: alsa-devel(a)alsa-project.org Cc: linux-fsdevel(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Arnd Bergmann (2): drm: kill BKL from common code sound: push BKL into open functions Matthew Wilcox (1): Remove BKL from fs/locks.c arch/um/drivers/hostaudio_kern.c | 6 ++ drivers/gpu/drm/drm_drv.c | 4 +- drivers/gpu/drm/drm_fops.c | 23 ++++---- drivers/gpu/drm/i810/i810_dma.c | 44 +++++++++----- drivers/gpu/drm/i810/i810_drv.c | 2 +- drivers/gpu/drm/i810/i810_drv.h | 1 + drivers/gpu/drm/i830/i830_dma.c | 42 +++++++++---- drivers/gpu/drm/i830/i830_drv.c | 2 +- drivers/gpu/drm/i830/i830_drv.h | 1 + fs/Kconfig | 4 + fs/locks.c | 116 ++++++++++++++++++++++-------------- include/drm/drmP.h | 2 +- sound/core/hwdep.c | 14 +++- sound/core/oss/mixer_oss.c | 19 ++++--- sound/oss/au1550_ac97.c | 26 +++++--- sound/oss/dmasound/dmasound_core.c | 28 +++++++-- sound/oss/msnd_pinnacle.c | 10 ++- sound/oss/sh_dac_audio.c | 9 ++- sound/oss/soundcard.c | 20 ++++--- sound/oss/swarm_cs4297a.c | 17 +++++- sound/oss/vwsnd.c | 8 +++ sound/sound_core.c | 6 +-- 22 files changed, 267 insertions(+), 137 deletions(-)

3 7

[alsa-devel] [PATCH 36/36] sound/core: Remove unnecessary casts of private_data
by Joe Perches 12 Jul '10

12 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/core/oss/mixer_oss.c | 22 +++++++++++----------- sound/core/pcm.c | 3 +-- sound/core/timer.c | 4 ++-- 3 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index f50ebf2..86afb13 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -77,7 +77,7 @@ static int snd_mixer_oss_release(struct inode *inode, struct file *file) struct snd_mixer_oss_file *fmixer; if (file->private_data) { - fmixer = (struct snd_mixer_oss_file *) file->private_data; + fmixer = file->private_data; module_put(fmixer->card->module); snd_card_file_remove(fmixer->card, file); kfree(fmixer); @@ -368,7 +368,7 @@ static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int static long snd_mixer_oss_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - return snd_mixer_oss_ioctl1((struct snd_mixer_oss_file *) file->private_data, cmd, arg); + return snd_mixer_oss_ioctl1(file->private_data, cmd, arg); } int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg) @@ -582,7 +582,7 @@ static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *left, int *right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; *left = *right = 100; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { @@ -691,7 +691,7 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int left, int right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); @@ -740,7 +740,7 @@ static int snd_mixer_oss_get_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -753,7 +753,7 @@ static int snd_mixer_oss_get_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -766,7 +766,7 @@ static int snd_mixer_oss_put_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], active, active, 0); return 0; @@ -776,7 +776,7 @@ static int snd_mixer_oss_put_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], active, active, 1); return 0; @@ -813,7 +813,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -861,7 +861,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -925,7 +925,7 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl static void snd_mixer_oss_slot_free(struct snd_mixer_oss_slot *chn) { - struct slot *p = (struct slot *)chn->private_data; + struct slot *p = chn->private_data; if (p) { if (p->allocated && p->assigned) { kfree(p->assigned->name); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cbe815d..de5a0eb 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -358,8 +358,7 @@ static void snd_pcm_stream_proc_info_read(struct snd_info_entry *entry, static void snd_pcm_substream_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - snd_pcm_proc_info_read((struct snd_pcm_substream *)entry->private_data, - buffer); + snd_pcm_proc_info_read(entry->private_data, buffer); } static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, diff --git a/sound/core/timer.c b/sound/core/timer.c index 13afb60..5ab1014 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -962,7 +962,7 @@ static int snd_timer_s_start(struct snd_timer * timer) struct snd_timer_system_private *priv; unsigned long njiff; - priv = (struct snd_timer_system_private *) timer->private_data; + priv = timer->private_data; njiff = (priv->last_jiffies = jiffies); if (priv->correction > timer->sticks - 1) { priv->correction -= timer->sticks - 1; @@ -981,7 +981,7 @@ static int snd_timer_s_stop(struct snd_timer * timer) struct snd_timer_system_private *priv; unsigned long jiff; - priv = (struct snd_timer_system_private *) timer->private_data; + priv = timer->private_data; del_timer(&priv->tlist); jiff = jiffies; if (time_before(jiff, priv->last_expires)) -- 1.7.1.337.g6068.dirty

1 0

[alsa-devel] [PATCH 35/36] sound/soc/davinci: Remove unnecessary casts of private_data
by Joe Perches 12 Jul '10

12 Jul '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/davinci/davinci-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2dc406f..def454e 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -800,7 +800,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; - iram_dma = (struct snd_dma_buffer *)buf->private_data; + iram_dma = buf->private_data; if (iram_dma) { sram_free(iram_dma->area, iram_dma->bytes); kfree(iram_dma); -- 1.7.1.337.g6068.dirty

1 0

[alsa-devel] How does struct snd_pcm_hardware relate to the actual hardware
by Adam Rosenberg 12 Jul '10

12 Jul '10

I am using DMA to transfer data to SPORT1 on my Blackfin BF537. I am using a CS42448 CODEC that has 8 DACs. I am using TDM mode and have set the following: DMA Buffer Size For Each SPORT Data Line (Rx or Tx) = 131072 bytes DMA Periods = 8 static struct snd_pcm_hardware snd_pcm_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME), .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, .rate_max = 48000, .buffer_bytes_max = 32768, .channels_min = 2, .channels_max = 2, .period_bytes_min = 8, .period_bytes_max = 4096, .periods_min = 8, .periods_max = 8, }; Everytime I receive a DMA interrupt from the SPORT driver I call snd_pcm_period_elapsed for all of the substreams that are currently active. In the PCM Copy callback I calculate the position within the DMA buffer to write to based on the TDM slot for the channel and the position passed into the function. The audio is not being placed into the DMA buffer properly. The audio plays faster than it should and has a constant stutter. I have read the "writing an ALSA driver" guide multiple times but it does not explain the relationship between struct snd_pcm_hardware, snd_pcm_period_elapsed, and the copy callback clearly enough for me to understand what I am doing wrong. Please explain how these interact in more detail to help me fix this problem. I can also send the full source code to the CS42448 driver I am working on if more details are needed. Thank you, Adam

2 2

[alsa-devel] [PATCH] ASoC: SAMSUNG: Add Aquila sound driver
by Chanwoo Choi 12 Jul '10

12 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2Sv5 to support sound. As there is currently no driver for the I2Sv5, the code to use I2Sv5 is deleted on this patch. After I2Sv5 driver is posted in a few weeks, I will post incremental patch which some codes for using I2Sv5. The owner of I2Sv5 said that he will post I2Sv5 driver after a few weeks. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 298 +++++++++++++++++++++++++++++++++++++ 3 files changed, 308 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..0c5d8a5 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,11 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..706087f --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,298 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.com> + * + * 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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_codec *codec) +{ + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai voice_dai = { + .name = "Voice", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + /* After I2Sv5 driver is posted, cpu_dai will be enabled. */ + /* .cpu_dai = &s5p_i2s_dai[0], */ + .codec_dai = &wm8994_dai[0], + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai = &voice_dai, + .codec_dai = &wm8994_dai[1], + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .platform = &s3c24xx_soc_platform, + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static struct snd_soc_device aquila_snd_devdata = { + .card = &aquila, + .codec_dev = &soc_codec_dev_wm8994, +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&voice_dai); + if (ret) + return ret; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + platform_set_drvdata(aquila_snd_device, &aquila_snd_devdata); + aquila_snd_devdata.dev = &aquila_snd_device->dev; + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

4 12

[alsa-devel] Help required regarding Capture & palyback code for ALSA
by Reena Chauhan 12 Jul '10

12 Jul '10

Hii I m new to ALSA programming .. i have follow the Guide "Introduction to Sound Programming with ALSA" to make code for capture & playback.when i have compiled the code for capture i dont know from where to give the input. in the Guide it is written that "If you have a microphone connected to your sound card, use a mixer program to set the recording source and level. Alternatively, you can run a CD player program and set the recording source to CD. Try running Listing 4 and redirecting the output to a file" i dont know how to use the mixer program .. but I dont know how to do it to capture some input given by me & playback it .. thanx & Regrads -- Reena Chauhan Scientist-C Application & Baseband Platform, SBSG ,DEAL,Raipur Road Dehradun-248001

1 0

Re: [alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Raymond Yau 10 Jul '10

10 Jul '10

>> It might be. It is important not to mix VT1708 and VT1708S. For VT1708S, the following DACs are used in code: /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: spec->multiout.dac_nids[i] = 0x25; break; } >> So 0x12 is not used for any of the surround outputs. I think your testing config is not compiled with smart 5.1 requirement Refer to smart51_get() , this function exepct red jack and blue jack are configued to Output But you still plugged in Mic into the pink jack according to your photo. Your speakers have to plugged in both pink and blue jacks

2 1

[alsa-devel] Kirkwood SoC - AVB support questions ..?
by errordeveloper＠gmail.com 10 Jul '10

10 Jul '10

Hello dear ALSA developer list members, I have a question regarding hardware features of Marvell Kirkwood chip. I is known that the NIC core on the chip includes support for Ethernet AV Bridging, which is still a draft standard ..but i suppose that Marvell released this chip with quite functional AVB implementation. grepping the sk98 driver code actually shows numerous lines with AVB as well as PTP, so i suppose they have implemented IOCTL in the sk98 driver ..i'm still doubtfull about Linux kernel PTP implementation ..is there now in some form or not? Did any of you have a look at AVB ? how do you think the driver for it could fit into Linux Netdev and ALSA? How are the CobraNet and EtherSound drivers designed, has anyone used those cards ?? (one was from DigiGram and another from Audio Science IIRC) Regards, -- Ilya Dmitrichenko

2 2

[alsa-devel] Envy24HT (VT1721) 192kHz digital out capable?
by Val Krutov 09 Jul '10

09 Jul '10

Could anybody get 192kHz out of VT1721 on spdif output working correctly? If yes please name your card. I have two cards on VT1721 chip, Audiotrak HD2 Advance DE and Chaintech AV-710. Both can work properly only up to 96kHz. The 192kHz samples play at 96kHz according to my receiver display sample rate input but alsamixer reports 192kHz which is incorrect. I believe my receiver' indication rather than alsamixer. Here is my register settings output for Chaintech AV-710 at playng 192kHz sample and everything there looks right to me but my receiver reports it is 96kHz. I just modified the card's ID and others eeprom settings as otherwise spdif can work only at 44.1 and 48 kHz. ICEnsemble ICE1724 at 0x2c40, irq 26 EEPROM: Subvendor : 0x12142419 Size : 28 bytes Version : 1 System Config : 0x2 ACLink : 0x82 I2S : 0x78 S/PDIF : 0xc1 GPIO direction : 0xff GPIO mask : 0xff GPIO state : 0xff Extra #18 : 0x1 Extra #19 : 0x0 Extra #20 : 0x0 Extra #21 : 0x0 Extra #22 : 0x0 Extra #23 : 0x0 Extra #24 : 0x0 Extra #25 : 0x0 Extra #26 : 0x0 Extra #27 : 0x0 Registers: PSDOUT03 : 0x00000000 CCS00 : 0x00 CCS01 : 0xa0 CCS02 : 0x00 CCS03 : 0x00 CCS04 : 0x02 CCS05 : 0x82 CCS06 : 0x78 CCS07 : 0xc1 CCS08 : 0x00 CCS09 : 0x00 CCS0a : 0x00 CCS0b : 0x00 CCS0c : 0x00 CCS0d : 0x0a CCS0e : 0x00 CCS0f : 0x00 CCS10 : 0xa0 CCS11 : 0x1b CCS12 : 0x00 CCS13 : 0x80 CCS14 : 0x00 CCS15 : 0xaa CCS16 : 0xff CCS17 : 0x00 CCS18 : 0xff CCS19 : 0x00 CCS1a : 0x00 CCS1b : 0x00 CCS1c : 0x00 CCS1d : 0x00 CCS1e : 0x2a CCS1f : 0x00 MT00 : 0x00 MT01 : 0x0e MT02 : 0x08 MT03 : 0x08 MT04 : 0x00 MT05 : 0x00 MT06 : 0x00 MT07 : 0x00 MT08 : 0x00 MT09 : 0x00 MT0a : 0x00 MT0b : 0x00 MT0c : 0x00 MT0d : 0x00 MT0e : 0x00 MT0f : 0x00 MT10 : 0x00 MT11 : 0x00 MT12 : 0x00 MT13 : 0x00 MT14 : 0x00 MT15 : 0x00 MT16 : 0x00 MT17 : 0x00 MT18 : 0x80 MT19 : 0x00 MT1a : 0x00 MT1b : 0x00 MT1c : 0x00 MT1d : 0x00 MT1e : 0x00 MT1f : 0x00 MT20 : 0x00 MT21 : 0x00 MT22 : 0x00 MT23 : 0x00 MT24 : 0x00 MT25 : 0x00 MT26 : 0x00 MT27 : 0x00 MT28 : 0x00 MT29 : 0x00 MT2a : 0x00 MT2b : 0x00 MT2c : 0x00 MT2d : 0x00 MT2e : 0x00 MT2f : 0x00 MT30 : 0x00 MT31 : 0x00 MT32 : 0x00 MT33 : 0x00 MT34 : 0x00 MT35 : 0x00 MT36 : 0x00 MT37 : 0x00 MT38 : 0x00 MT39 : 0x00 MT3a : 0x00 MT3b : 0x00 MT3c : 0x00 MT3d : 0x60 MT3e : 0x00 MT3f : 0x00 MT40 : 0x80 MT41 : 0x48 MT42 : 0x98 MT43 : 0x32 MT44 : 0xdf MT45 : 0xed MT46 : 0xff MT47 : 0x3f MT48 : 0x00 MT49 : 0x00 MT4a : 0x00 MT4b : 0x00 MT4c : 0x00 MT4d : 0x00 MT4e : 0x00 MT4f : 0x00 MT50 : 0x00 MT51 : 0x00 MT52 : 0x00 MT53 : 0x00 MT54 : 0x00 MT55 : 0x00 MT56 : 0x00 MT57 : 0x00 MT58 : 0x00 MT59 : 0x00 MT5a : 0x00 MT5b : 0x00 MT5c : 0x00 MT5d : 0x00 MT5e : 0x00 MT5f : 0x00 MT60 : 0x00 MT61 : 0x00 MT62 : 0x00 MT63 : 0x00 MT64 : 0x00 MT65 : 0x00 MT66 : 0x00 MT67 : 0x00 MT68 : 0x00 MT69 : 0x00 MT6a : 0x00 MT6b : 0x00 MT6c : 0x00 MT6d : 0x00 MT6e : 0x00 MT6f : 0x00 MT70 : 0x00 MT71 : 0x00 MT72 : 0x00 MT73 : 0x00 MT74 : 0x00 MT75 : 0x00 MT76 : 0x00 MT77 : 0x00 Valera

1 0

Re: [alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Viliam Kubis 09 Jul '10

09 Jul '10

It might be. It is important not to mix VT1708 and VT1708S. For VT1708S, the following DACs are used in code: /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: spec->multiout.dac_nids[i] = 0x25; break; } So 0x12 is not used for any of the surround outputs. On Fri, 09 Jul 2010 03:32:13 +0200, Raymond Yau <superquad.vortex2(a)gmail.com> wrote: >> 2010/7/8 Clemens Ladisch <clemens(a)ladisch.de> >> >>> Viliam Kubis wrote: >>> > Hello, for quite some time the "Smart 5.1" function in alsa is not >>> working >>> > properly when motherboard has only 3 audio jacks. >>> > >>> > speaker-test -c 4 -t wav -Dsurround40 >>> > >>> > Channels count (4) not available for playbacks: Invalid argument >>> >>> Please try the patch below. >>> >>> (This function uses 0x12 and 0x13 as the NIDs for the center/LFE and >>> side DACs. Shouldn't these be 0x24 and 0x25, as per the datasheet?) >>> >> >> Node 0x12 seem to be for digital out since it connected to node 0x20 >> >> I guess we still lack the info about the retasking of pink and blue >> jack to >> the surround output >> >> >> Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital >> Control: name="IEC958 Playback Con Mask", index=0, device=0 >> Control: name="IEC958 Playback Pro Mask", index=0, device=0 >> Control: name="IEC958 Playback Default", index=0, device=0 >> Control: name="IEC958 Playback Switch", index=0, device=0 >> Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 >> Device: name="VT1708S Digital", type="SPDIF", device=1 >> Converter: stream=5, channel=0 >> Digital: Enabled >> Digital category: 0x0 >> PCM: >> rates [0x5e0]: 44100 48000 88200 96000 192000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> >> >> Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital >> Pincap 0x00000010: OUT >> Pin Default 0x074311f0: [Jack] SPDIF Out at Ext Rear Panel >> Conn = ATAPI, Color = Black >> DefAssociation = 0xf, Sequence = 0x0 >> Misc = NO_PRESENCE >> Pin-ctls: 0x40: OUT >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Connection: 1 >> 0x12 >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

1 0

Re: [alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Viliam Kubis 09 Jul '10

09 Jul '10

Hello, I have patched the vt1708s_auto_fill_dac_nids function instead. I have left values 0x11 and 0x12 for surround and cen/lfe, althrough it might be 0x11 and 0x24 for vt1708S as originally specified in vt1708s_auto_fill_dac_nids switch statement. The whole function looks like this: static int vt1708S_auto_fill_dac_nids(struct via_spec *spec, const struct auto_pin_cfg *cfg) { int i; hda_nid_t nid; spec->multiout.num_dacs = cfg->line_outs; spec->multiout.dac_nids = spec->private_dac_nids; for (i = 0; i < 4; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: printk(KERN_INFO "***PATCH CREATING SURROUND DAC_NID"); spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: spec->multiout.dac_nids[i] = 0x25; break; } } } if (cfg->line_outs == 1) { printk(KERN_INFO "***PATCH 08/07/10"); spec->multiout.num_dacs = 3; spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x12; } return 0; } The patch is in effect since the following is visible in dmesg after the driver is loaded: ***PATCH 08/07/10 ***PATCH CREATING Surround CONTROL ***PATCH HDA max_channels = 6 ***PATCH playback stream channels_max = 6 Smart 5.1 Patch installed NEW ***PATCH Setting 64 mode [out=64] for pin index 0 nid 26 ***PATCH Setting 64 mode [out=64] for pin index 1 nid 30 ***PATCH Setting 64 mode [out=64] for pin index 2 nid 27 printk(KERN_INFO "***PATCH CREATING SURROUND DAC_NID"); This line does not print since cfg->line_out_pins[AUTO_SEQ_SURROUND] is not filled at that time, but we fix it in the new if block. Surround still does not work (still the same, the rear speakers are soundless). On Thu, 08 Jul 2010 08:33:34 +0200, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> _auto_fill_dac_nids > -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

1 0

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 09 Jul '10

09 Jul '10

The following changes since commit db059c0f6edec0b1deba665e5853f4ed829003b7: ASoC: Automatically manage ALC coefficients for WM8960 (2010-07-06 08:46:10 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Mark Brown (2): ASoC: Include WM8994 GPIO and interrupt registers in codec_reg ASoC: Implement WM8994 OPCLK support Maurus Cuelenaere (1): ASoC: Invert speaker enabling behaviour in SmartQ sound driver Raffaele Recalcati (3): ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S) ASoC: DaVinci: Added selection of clk input pin for McBSP ASoC: DaVinci: More accurate continuous serial clock for McBSP (I2S) arch/arm/mach-davinci/include/mach/asp.h | 48 +++++++++ sound/soc/codecs/wm8994.c | 43 +++++++++ sound/soc/codecs/wm8994.h | 3 + sound/soc/davinci/davinci-i2s.c | 153 +++++++++++++++++++++++++++--- sound/soc/davinci/davinci-i2s.h | 5 + sound/soc/s3c24xx/smartq_wm8987.c | 1 - 6 files changed, 238 insertions(+), 15 deletions(-)

2 1

[alsa-devel] Question about SuperH HDMI sound
by Kuninori Morimoto 09 Jul '10

09 Jul '10

Dear Mark Cc: Guennadi In these days, Guennadi's "SuperH HDMI support patches" which support HDMI video output were posted to ML. And I created HDMI sound support for it in my local environment. It works and I would like to post this patch. Thus, I would like to ask you about HDMI "sound" support file position. HDMI "video" support file is under "drivers/video/sh_mobile_hdmi.c" Should I add sound support in it ? or under "sound/soc/codecs" ? Best regards -- Kuninori Morimoto

2 3

[alsa-devel] Smart 5.1 function not working on VIA VT1708S with *three* jacks on mainboard
by Viliam Kubis 09 Jul '10

09 Jul '10

Hello, for quite some time the "Smart 5.1" function in alsa is not working properly when motherboard has only 3 audio jacks. Me and Raymond on alsa-bugtracker have been trying to solve the issue for a long time but were unsuccessful. Please take a look at this alsa bug ticket: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4895 It contains A LOT of info on the problem + steps that we tried, which included patching the code. Synthom : NO "Surround" , "Center" and "LFE" Control and switch with VT1708S on a motherboard with 3 audio jacks Also NO surround sound. speaker-test -c 4 -t wav -Dsurround40 Channels count (4) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument Please advise if you can. Thank you Best Regards Viliam Kubis -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

3 3

[alsa-devel] wm8776_dai vs. cs4270_dai
by Timur Tabi 09 Jul '10

09 Jul '10

Mark, In wm8776.c, the DAI structure is defined as an array of two elements, one for playback and one for capture: struct snd_soc_dai_driver wm8776_dai[] = { { .name = "WM8776 Playback", .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = WM8776_RATES, .formats = WM8776_FORMATS, }, .ops = &wm8776_dac_ops, }, { .name = "WM8776 Capture", .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = WM8776_RATES, .formats = WM8776_FORMATS, }, .ops = &wm8776_adc_ops, }, }; In my CS4270 driver, I have these combined into one structure: struct snd_soc_dai_driver cs4270_dai = { .name = "cs4270", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 4000, .rate_max = 216000, .formats = CS4270_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 4000, .rate_max = 216000, .formats = CS4270_FORMATS, }, .ops = &cs4270_dai_ops, }; Should the CS4270 driver be changed to use two structures?

2 1

[alsa-devel] Problems running "amixer"
by Adish Kuvelker 08 Jul '10

08 Jul '10

Hello, I have cross-compiled alsa-utils (alsa-utils-1.0.13) and alsa-lib (alsa-lib-1.0.13). But when I ran the "amixer" I found some errors. I am pasting the same for your reference. # ./amixer ALSA lib control.c:910:(snd_ctl_open_noupdate) Invalid CTL default amixer: Mixer attach default error: No such file or directory The snddevices script creates a node "/dev/snd" and the device number of these nodes matches with the ones registered in "/proc/asound". Can anyone point me to the reason why the above mentioned error occurs? If any more information is required over the way my sound nodes are registered with the system I shall post it. Regards; Aadish

2 5

[alsa-devel] Still unhappy with removal of DXS controls
by Soeren D. Schulze 08 Jul '10

08 Jul '10

[Sorry if this is a duplicate -- I was too stupid to subscribe the first time and my mail hasn't been moderated yet, apparently.] Me again... Referring to the issue I described on 2010-03-28 [http://www.spinics.net/lists/alsa-devel/msg32731.html], the situation hasn't changed since then. In short, I *really* need to lower the DXS volumes in order to play sound properly. If I don't do it, I get massive distortion. It's not a matter of the output stage, but of the input stage. Lowering PCM or Master doesn't help anything, but using softvol in mplayer does (I need to get down to 20%, so I effectively hear 14-bit audio, nice...). Back in March, Clemens told me to use hwmixvolume. It's nice, I can now access what I could previously access in alsamixer. But when I close the audio stream and start a new one, the volume is reset to 100% -- blowing my ears with loud and distorted sound. So my solution for now is: stick with Linux 2.6.30... Please, even if the solution with the DXS controls in alsamixer didn't comply with the ALSA policies, *it did work*! The current one *does not*. Maybe make the maximum (or default?) DXS volume available as a mixer control or as a module option or something -- whatever you consider the best solution. But the current state is just not acceptable. Thanks Sören

3 6

[alsa-devel] [PATCH] ASoC: SAMSUNG: Add Aquila sound driver
by 최찬우 08 Jul '10

08 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2Sv5 to support sound. As there is currently no driver for the I2Sv5, the code to use I2Sv5 is deleted on this patch. After I2Sv5 driver is posted in a few weeks, I will post incremental patch which some codes for using I2Sv5. The owner of I2Sv5 said that he will post I2Sv5 driver after a few weeks. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 306 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..0c5d8a5 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,11 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..5caeedd --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,296 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * 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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +extern struct snd_soc_platform s3c24xx_soc_platform; +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_kcontrol_new aquila_controls[] = { +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, +}; + +static int aquila_wm8994_init(struct snd_soc_codec *codec) +{ + int ret; + unsigned int reg; + + /* add aquila specific kcontorls */ + ret = snd_soc_add_controls(codec, aquila_controls, + ARRAY_SIZE(aquila_controls)); + + if (ret < 0) + return ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai voice_dai = { + .name = "Voice", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + /* After I2Sv5 driver is posted, cpu_dai will is enabled. */ + /* .cpu_dai = &s5p_i2s_dai[0], */ + .codec_dai = &wm8994_dai[0], + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai = &voice_dai, + .codec_dai = &wm8994_dai[1], + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .platform = &s3c24xx_soc_platform, + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static struct snd_soc_device aquila_snd_devdata = { + .card = &aquila, + .codec_dev = &soc_codec_dev_wm8994, +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&voice_dai); + if (ret) + return ret; + + aquila_snd_device = platform_device_alloc("soc-audio", 0); + if (!aquila_snd_device) + return -ENOMEM; + + platform_set_drvdata(aquila_snd_device, &aquila_snd_devdata); + aquila_snd_devdata.dev = &aquila_snd_device->dev; + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(C110)"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

2 2

[alsa-devel] [PATCH] alsa-utils: Allow building in subdirectory.
by Thierry Reding 07 Jul '10

07 Jul '10

This patch allows building alsa-utils in a separate subdirectory. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- alsactl/Makefile.am | 2 ++ alsamixer/Makefile.am | 2 +- 2 files changed, 3 insertions(+), 1 deletions(-) diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index 359f73a..945a54c 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -1,5 +1,7 @@ SUBDIRS = init +INCLUDES = -I$(top_srcdir)/include + sbin_PROGRAMS=alsactl man_MANS=alsactl.1 if USE_XMLTO diff --git a/alsamixer/Makefile.am b/alsamixer/Makefile.am index 1de47c6..d01b1e4 100644 --- a/alsamixer/Makefile.am +++ b/alsamixer/Makefile.am @@ -1,4 +1,5 @@ AM_CFLAGS = @CURSES_CFLAGS@ -DCURSESINC="@CURSESINC@" +INCLUDES = -I$(top_srcdir)/include LDADD = @CURSESLIB@ bin_PROGRAMS = alsamixer @@ -18,7 +19,6 @@ alsamixer_SOURCES = card_select.c card_select.h \ widget.c widget.h man_MANS = alsamixer.1 EXTRA_DIST = alsamixer.1 -alsamixer_INCLUDES = -I$(top_srcdir)/include #LDFLAGS = -static #CFLAGS += -g -Wall -- 1.7.1

1 0

[PATCH - Fix Fast Track Ultra series 1/1] Fix 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix Homann 07 Jul '10

07 Jul '10

Signed-off-by: Felix Homann <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..4c1dc12 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2080), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.1 --------------090102060507020204090408--

1 0

[PATCH - Fix Fast Track Ultra series 1/1] Fix 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix Homann 07 Jul '10

07 Jul '10

Signed-off-by: Felix Homann <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..4c1dc12 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2080), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.1 --------------020109060502090203040503--

1 0

Re: [alsa-devel] [PATCH 1/3] ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)
by Mark Brown 06 Jul '10

06 Jul '10

On Tue, Jul 06, 2010 at 05:00:24PM +0530, Sudhakar Rajashekhara wrote: > > Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> > Acked-by: Sudhakar Rajashekhara <sudhakar.raj(a)ti.com> Applied all, thanks.

1 0

[alsa-devel] [PATCH 2/3] ASoC: Add SmartQ sound driver
by Maurus Cuelenaere 06 Jul '10

06 Jul '10

This adds sound support for the SmartQ board. The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750 driver is used for driving the WM8987, as they are register compatible. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/soc/s3c24xx/Kconfig | 6 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smartq_wm8987.c | 296 +++++++++++++++++++++++++++++++++++++ 3 files changed, 304 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smartq_wm8987.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2a7cc22..4f46ba0 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -125,3 +125,9 @@ config SND_SOC_SMDK_WM9713 select SND_S3C_SOC_AC97 help Sat Y if you want to add support for SoC audio on the SMDK. + +config SND_S3C64XX_SOC_SMARTQ + tristate "SoC I2S Audio support for SmartQ board" + depends on SND_S3C24XX_SOC && MACH_SMARTQ + select SND_S3C64XX_SOC_I2S + select SND_SOC_WM8750 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 81d8dc5..50172c3 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -29,6 +29,7 @@ snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o +snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -41,3 +42,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o +obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c new file mode 100644 index 0000000..c90ef96 --- /dev/null +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -0,0 +1,296 @@ +/* sound/soc/s3c24xx/smartq_wm8987.c + * + * Copyright 2010 Maurus Cuelenaere <mcuelenaere(a)gmail.com> + * + * Based on smdk6410_wm8987.c + * Copyright 2007 Wolfson Microelectronics PLC. - linux(a)wolfsonmicro.com + * Graeme Gregory - graeme.gregory(a)wolfsonmicro.com + * + * 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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#include "../codecs/wm8750.h" + +/* + * WM8987 is register compatible with WM8750, so using that as base driver. + */ + +static struct snd_soc_card snd_soc_smartq; + +static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret; + + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c_i2sv2_get_clock(cpu_dai)); + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + clk = 11289600; + 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; + + /* set cpu DAI configuration */ + 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; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SmartQ WM8987 HiFi DAI operations. + */ +static struct snd_soc_ops smartq_hifi_ops = { + .hw_params = smartq_hifi_hw_params, +}; + +static struct snd_soc_jack smartq_jack; + +static struct snd_soc_jack_pin smartq_jack_pins[] = { + /* Disable speaker when headphone is plugged in */ + { + .pin = "Internal Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = true, + }, +}; + +static struct snd_soc_jack_gpio smartq_jack_gpios[] = { + { + .gpio = S3C64XX_GPL(12), + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 200, + }, +}; + +static const struct snd_kcontrol_new wm8987_smartq_controls[] = { + SOC_DAPM_PIN_SWITCH("Internal Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), +}; + +static int smartq_speaker_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8987_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Internal Speaker", smartq_speaker_event), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LOUT2"}, + {"Headphone Jack", NULL, "ROUT2"}, + + {"Internal Speaker", NULL, "LOUT2"}, + {"Internal Speaker", NULL, "ROUT2"}, + + {"Mic Bias", NULL, "Internal Mic"}, + {"LINPUT2", NULL, "Mic Bias"}, +}; + +static int smartq_wm8987_init(struct snd_soc_codec *codec) +{ + int err = 0; + + /* Add SmartQ specific widgets */ + snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets, + ARRAY_SIZE(wm8987_dapm_widgets)); + + /* add SmartQ specific controls */ + err = snd_soc_add_controls(codec, wm8987_smartq_controls, + ARRAY_SIZE(wm8987_smartq_controls)); + + if (err < 0) + return err; + + /* setup SmartQ specific audio path */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "LINPUT1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "ROUT1"); + + /* set endpoints to default off mode */ + snd_soc_dapm_enable_pin(codec, "Internal Speaker"); + snd_soc_dapm_enable_pin(codec, "Internal Mic"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + err = snd_soc_dapm_sync(codec); + if (err) + return err; + + /* Headphone jack detection */ + err = snd_soc_jack_new(&snd_soc_smartq, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack); + if (err) + return err; + + err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), + smartq_jack_pins); + if (err) + return err; + + err = snd_soc_jack_add_gpios(&smartq_jack, + ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + return err; +} + +static struct snd_soc_dai_link smartq_dai[] = { + { + .name = "wm8987", + .stream_name = "SmartQ Hi-Fi", + .cpu_dai = &s3c64xx_i2s_dai[0], + .codec_dai = &wm8750_dai, + .init = smartq_wm8987_init, + .ops = &smartq_hifi_ops, + }, +}; + +static struct snd_soc_card snd_soc_smartq = { + .name = "SmartQ", + .platform = &s3c24xx_soc_platform, + .dai_link = smartq_dai, + .num_links = ARRAY_SIZE(smartq_dai), +}; + +static struct snd_soc_device smartq_snd_devdata = { + .card = &snd_soc_smartq, + .codec_dev = &soc_codec_dev_wm8750, +}; + +static struct platform_device *smartq_snd_device; + +static int __init smartq_init(void) +{ + int ret; + + if (!machine_is_smartq7() && !machine_is_smartq5()) { + pr_info("Only SmartQ is supported by this ASoC driver\n"); + return -ENODEV; + } + + smartq_snd_device = platform_device_alloc("soc-audio", -1); + if (!smartq_snd_device) + return -ENOMEM; + + platform_set_drvdata(smartq_snd_device, &smartq_snd_devdata); + smartq_snd_devdata.dev = &smartq_snd_device->dev; + + ret = platform_device_add(smartq_snd_device); + if (ret) { + platform_device_put(smartq_snd_device); + return ret; + } + + /* Initialise GPIOs used by amplifiers */ + ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown"); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n"); + goto err_unregister_device; + } + + /* Disable amplifiers */ + ret = gpio_direction_output(S3C64XX_GPK(12), 1); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n"); + goto err_free_gpio_amp_shut; + } + + return 0; + +err_free_gpio_amp_shut: + gpio_free(S3C64XX_GPK(12)); +err_unregister_device: + platform_device_unregister(smartq_snd_device); + + return ret; +} + +static void __exit smartq_exit(void) +{ + snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + platform_device_unregister(smartq_snd_device); +} + +module_init(smartq_init); +module_exit(smartq_exit); + +/* Module information */ +MODULE_AUTHOR("Maurus Cuelenaere <mcuelenaere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC SmartQ WM8987"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

3 9

[alsa-devel] [PATCH 1/3] ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)
by Raffaele Recalcati 06 Jul '10

06 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added two clocking options for dm365 McBSP peripheral when used with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock from external pin and generates frame sync). A slave clock management can be important when the external codec needs the system clock and the bit clock synchronized (tested with uda1345). This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- sound/soc/davinci/davinci-i2s.c | 110 +++++++++++++++++++++++++++++++++++--- sound/soc/davinci/davinci-i2s.h | 5 ++ 2 files changed, 106 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..c8f038c 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -26,6 +26,7 @@ #include <mach/asp.h> #include "davinci-pcm.h" +#include "davinci-i2s.h" /* @@ -68,16 +69,21 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE BIT(31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE BIT(31) #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM BIT(29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,6 +150,9 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + unsigned int fmt; + int clk_div; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -254,10 +263,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = cpu_dai->private_data; unsigned int pcr; unsigned int srgr; + /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + dev->fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -372,6 +383,18 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + + if (div_id != DAVINCI_MCBSP_CLKGDV) + return -ENODEV; + + dev->clk_div = div; + return 0; +} + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,8 +403,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; @@ -396,12 +419,47 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + freq = clk_get_rate(dev->clk); + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFS: + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = dev->clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Clock and frame sync given from external sources */ + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + break; + default: + return -EINVAL; + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,12 +484,41 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + break; + default: + return -EINVAL; + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + break; + default: + return -EINVAL; + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -442,6 +529,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,6 +591,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h index 241648c..0b1e77b 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/davinci/davinci-i2s.h @@ -12,6 +12,11 @@ #ifndef _DAVINCI_I2S_H #define _DAVINCI_I2S_H +/* McBSP dividers */ +enum davinci_mcbsp_div { + DAVINCI_MCBSP_CLKGDV, /* Sample rate generator divider */ +}; + extern struct snd_soc_dai davinci_i2s_dai; #endif -- 1.7.0.4

3 2

[alsa-devel] [PATCH 3/3] ASoC: DaVinci: More accurate continuous serial clock for McBSP (I2S)
by Raffaele Recalcati 06 Jul '10

06 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> i2s_accurate_sck switch can be used to have a better approximate sampling frequency. The clock is an externally visible bit clock and it is named i2s continuous serial clock (I2S_SCK). The trade off is between more accurate clock (fast clock) and less accurate clock (slow clock). The waveform will be not symmetric. Probably it is possible to get a better algorithm for calculating the divider, trying to keep a slower clock as possible. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 33 ++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s.c | 24 +++++++++++++++++---- 2 files changed, 52 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 0847d21..b12c69e 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -73,6 +73,39 @@ struct snd_platform_data { */ int clk_input_pin; + /* + * This flag works when both clock and FS are outputs for the cpu + * and makes clock more accurate (FS is not symmetrical and the + * clock is very fast. + * The clock becoming faster is named + * i2s continuous serial clock (I2S_SCK) and it is an externally + * visible bit clock. + * + * first line : WordSelect + * second line : ContinuousSerialClock + * third line: SerialData + * + * SYMMETRICAL APPROACH: + * _______________________ LEFT + * _| RIGHT |______________________| + * _ _ _ _ _ _ _ _ + * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_ + * _ _ _ _ _ _ _ _ + * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_ + * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ + * + * ACCURATE CLOCK APPROACH: + * ______________ LEFT + * _| RIGHT |_______________________________| + * _ _ _ _ _ _ _ _ _ + * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| | + * _ _ _ _ dummy cycles + * _/ \_ ... _/ \_/ \_ ... _/ \__________________ + * \_/ \_/ \_/ \_/ + * + */ + bool i2s_accurate_sck; + /* McASP specific fields */ int tdm_slots; u8 op_mode; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ba5644b..b251bc9 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -155,6 +155,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; int clk_input_pin; + bool i2s_accurate_sck; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -447,11 +448,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SRGR_CLKSM; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); - /* symmetric waveforms */ - clk_div = freq / (mcbsp_word_length * 16) / - params->rate_num * params->rate_den; - srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * - 16 - 1); + if (dev->i2s_accurate_sck) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } clk_div &= 0xFF; srgr |= clk_div; break; @@ -662,6 +675,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; dev->clk_input_pin = pdata->clk_input_pin; + dev->i2s_accurate_sck = pdata->i2s_accurate_sck; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { -- 1.7.0.4

1 0

[alsa-devel] [PATCH 2/3] ASoC: DaVinci: Added selection of clk input pin for McBSP
by Raffaele Recalcati 06 Jul '10

06 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> When McBSP peripheral gets the clock from an external pin, there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR and MCBSP_CLKS. evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different hardware connection and I use MCBSP_CLKS, so I have added this possibility. This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm) Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 15 +++++++++++++++ sound/soc/davinci/davinci-i2s.c | 29 ++++++++++++++++++++++++----- 2 files changed, 39 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..0847d21 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,16 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * If McBSP peripheral gets the clock from an external pin, + * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR + * and MCBSP_CLKS. + * Depending on different hardware connections it is possible + * to use this setting to change the behaviour of McBSP + * driver. The dm365_clk_input_pin enum is available for dm365 + */ + int clk_input_pin; + /* McASP specific fields */ int tdm_slots; u8 op_mode; @@ -78,6 +88,11 @@ enum { MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; +enum dm365_clk_input_pin { + MCBSP_CLKR = 0, /* DM365 */ + MCBSP_CLKS, +}; + #define INACTIVE_MODE 0 #define TX_MODE 1 #define RX_MODE 2 diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c8f038c..ba5644b 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -122,6 +122,7 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { }; struct davinci_mcbsp_dev { + struct device *dev; struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -153,6 +154,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; + int clk_input_pin; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -279,11 +281,26 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: - /* McBSP CLKR pin is the input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + /* + * Selection of the clock input pin that is the + * input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + switch (dev->clk_input_pin) { + case MCBSP_CLKS: + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case MCBSP_CLKR: + pcr |= DAVINCI_MCBSP_PCR_SCLKME; + break; + default: + dev_err(dev->dev, "bad clk_input_pin\n"); + return -EINVAL; + } + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -644,6 +661,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { @@ -676,6 +694,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; + dev->dev = &pdev->dev; davinci_i2s_dai.private_data = dev; davinci_i2s_dai.capture.dma_data = dev->dma_params; -- 1.7.0.4

1 0

[PATCH - Fix Fast Track Ultra series: 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix 06 Jul '10

06 Jul '10

Signed-off-by: Felix <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..af9c2a1 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2081), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.0.4 --------------090100050500090907030802--

1 0

[PATCH - Fix Fast Track Ultra series: 44.1/88.1 kHz 1/Fix playback/capture issues and related chrashes at 44100 Hz and 88200 Hz for M-Audio Fast Track Ultra series.
by Felix 06 Jul '10

06 Jul '10

Signed-off-by: Felix <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index 1a701f1..bb9f938 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index f8797f6..af9c2a1 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1827,10 +1827,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + USB_DEVICE(0x0763, 0x2081), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1855,7 +1895,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ .ifnum = QUIRK_ANY_INTERFACE, @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x01, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- 1.7.0.4 --------------040009020804090202030406--

1 0

[alsa-devel] Resent HPI 4.03.36 updated summaries
by linux＠audioscience.com 06 Jul '10

06 Jul '10

Same patches, new summaries.

2 7

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 06 Jul '10

06 Jul '10

The following changes since commit ce93a3702832121d517ad348817929f22fcce47c: ASoC: Add SmartQ sound driver (2010-07-04 18:04:12 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Mark Brown (4): ASoC: Fix sorting of Makefile and Kconfig ASoC: Remove current WM8960 deemphasis control ASoC: Automatically manage DAC deemphasis rate for WM8960 ASoC: Automatically manage ALC coefficients for WM8960 sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/codecs/wm8960.c | 99 +++++++++++++++++++++++++++++++++++++++++---- 3 files changed, 93 insertions(+), 10 deletions(-)

2 1

[alsa-devel] [PATCH] Updated alsa-kernel asihpi sources so patches are no longer required.
by linux＠audioscience.com 06 Jul '10

06 Jul '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- .gitignore | 2 -- pci/asihpi/Makefile | 3 --- pci/asihpi/hpidspcd.c | 2 ++ pci/asihpi/hpidspcd.patch | 21 --------------------- pci/asihpi/hpios.c | 2 ++ pci/asihpi/hpios.patch | 35 ----------------------------------- 6 files changed, 4 insertions(+), 61 deletions(-) create mode 100644 pci/asihpi/hpidspcd.c delete mode 100644 pci/asihpi/hpidspcd.patch create mode 100644 pci/asihpi/hpios.c delete mode 100644 pci/asihpi/hpios.patch diff --git a/.gitignore b/.gitignore index 204b952..db642b2 100644 --- a/.gitignore +++ b/.gitignore @@ -75,8 +75,6 @@ pci/ac97/ac97_bus.c pci/ac97/ac97_codec.c pci/ad1889.c pci/ali5451/ali5451.c -pci/asihpi/hpidspcd.c -pci/asihpi/hpios.c pci/atiixp.c pci/atiixp_modem.c pci/au88x0/au88x0.c diff --git a/pci/asihpi/Makefile b/pci/asihpi/Makefile index 95ee449..50ab325 100644 --- a/pci/asihpi/Makefile +++ b/pci/asihpi/Makefile @@ -5,7 +5,6 @@ endif include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf -clean-files := hpidspcd.c hpios.c EXTRA_CFLAGS = -I$(SND_TOPDIR)/alsa-kernel/pci/asihpi @@ -13,5 +12,3 @@ include $(SND_TOPDIR)/alsa-kernel/pci/asihpi/Makefile include $(SND_TOPDIR)/Rules.make -hpidspcd.c: hpidspcd.patch $(SND_TOPDIR)/alsa-kernel/pci/asihpi/hpidspcd.c -hpios.c: hpios.patch $(SND_TOPDIR)/alsa-kernel/pci/asihpi/hpios.c diff --git a/pci/asihpi/hpidspcd.c b/pci/asihpi/hpidspcd.c new file mode 100644 index 0000000..9b7d80a --- /dev/null +++ b/pci/asihpi/hpidspcd.c @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/pci/asihpi/hpidspcd.c" diff --git a/pci/asihpi/hpidspcd.patch b/pci/asihpi/hpidspcd.patch deleted file mode 100644 index 7831639..0000000 --- a/pci/asihpi/hpidspcd.patch +++ /dev/null @@ -1,21 +0,0 @@ ---- ../../alsa-kernel/pci/asihpi/hpidspcd.c 2010-04-21 18:36:16.613453111 +0200 -+++ hpidspcd.c 2010-04-21 18:41:26.104970371 +0200 -@@ -1,3 +1,4 @@ -+#include "adriver.h" - /***********************************************************************/ - /*! - -@@ -73,8 +74,13 @@ - sprintf(fw_name, "asihpi/dsp%04x.bin", adapter); - HPI_DEBUG_LOG(INFO, "requesting firmware for %s\n", fw_name); - -+#if (LINUX_VERSION_CODE < KERNEL_VERSION(2 , 5 , 0)) -+ err = request_firmware(&ps_firmware, fw_name, -+ ps_dsp_code->ps_dev->slot_name); -+#else - err = request_firmware(&ps_firmware, fw_name, - &ps_dsp_code->ps_dev->dev); -+#endif - if (err != 0) { - HPI_DEBUG_LOG(ERROR, "%d, request_firmware failed for %s\n", - err, fw_name); diff --git a/pci/asihpi/hpios.c b/pci/asihpi/hpios.c new file mode 100644 index 0000000..3f151a3 --- /dev/null +++ b/pci/asihpi/hpios.c @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/pci/asihpi/hpios.c" diff --git a/pci/asihpi/hpios.patch b/pci/asihpi/hpios.patch deleted file mode 100644 index 1af0f7f..0000000 --- a/pci/asihpi/hpios.patch +++ /dev/null @@ -1,35 +0,0 @@ ---- ../../alsa-kernel/pci/asihpi/hpios.c 2010-04-21 18:36:16.617452068 +0200 -+++ hpios.c 2010-04-21 18:40:32.700700812 +0200 -@@ -1,3 +1,4 @@ -+#include "adriver.h" - /****************************************************************************** - - AudioScience HPI driver -@@ -26,6 +27,7 @@ - #include <linux/delay.h> - #include <linux/sched.h> - -+#if LINUX_VERSION_CODE >= KERNEL_VERSION(2 , 6 , 14) - void hpios_delay_micro_seconds(u32 num_micro_sec) - { - if ((usecs_to_jiffies(num_micro_sec) > 1) && !in_interrupt()) { -@@ -38,6 +40,19 @@ - mdelay(num_micro_sec / 1000); - - } -+#else -+void hpios_delay_micro_seconds(u32 num_micro_sec) -+{ -+ if ((num_micro_sec / 1000 >= 1000000 / HZ) && !in_interrupt()) { -+ set_current_state(TASK_UNINTERRUPTIBLE); -+ schedule_timeout((HZ * num_micro_sec + (HZ - 1)) / 1000000); -+ } else if (num_micro_sec <= 2000) -+ udelay(num_micro_sec); -+ else -+ mdelay(num_micro_sec / 1000); -+ -+} -+#endif - - void hpios_locked_mem_init(void) - { -- 1.7.0.4

3 3

Re: [alsa-devel] Help with dummy.c (where/how to write?)
by Smilen Dimitrov 06 Jul '10

06 Jul '10

Hi Eliot, Thanks so much for your response: >> >> I am trying to work on dummy.c: >> >> http://lxr.linux.no/#linux+v2.6.32/sound/drivers/dummy.c >> >> as an exercise, and I'd like to modify it so it behaves as an >> 'oscillator'; [snip] >> > > You are better to look at the loopback driver in alsa-driver repo: > > http://git.alsa-project.org/?p=alsa-driver.git;a=blob_plain;f=drivers/aloop… > > You would put your generation code in clear_capture_buf() > Fantastic info - that is exactly what I needed :) (Don't know why your response hasn't shown in the alsa-devel archives yet, hopefully my reply will bump it up there) Cheers!

1 0

[alsa-devel] [RFC 00/16] ASoC: multi-component - CODECs
by Liam Girdwood 06 Jul '10

06 Jul '10

This is the second part of the multi-component series and focuses on the changes required to support ASoC multi-component on CODEC drivers. I've split the CODEC driver patches into smaller chunks and tried to group (where possible) similar CODECs together. All the patches will be rebased together at upstream time so we don't break bisect. Most CODEC drivers were extremely straight forward with changes limited to their probe() and remove() sections. Some CODECs did require a little extra work and this is marked in their commit messages. Thanks Liam

5 24

[alsa-devel] [RFC 0/5] ASoC multi-component support : core
by Liam Girdwood 06 Jul '10

06 Jul '10

ASoC is now coming up to it's 5th birthday and is beginning to show it's age a little when it comes to supporting the next generation of smart phones and multimedia devices with it's current single CODEC and single DMA platform design. Historically, ASoC is designed around an embedded sound card with a single CODEC and a single platform DMA driver (with multiple DAIs). This design works great for most of the current audio hardware in production, however we are now seeing hardware designs for the next generation of devices that can have many CODECs, DMAs, DSPs and a host of other audio peripherals connected to the audio subsystem. So we now need ASoC to support new audio hardware architectures with multiple audio CODECs, DMA engines, DSPs, DAIs etc. This patch series allows an ASoC sound card to support multiple components (CODECs, DAI and DMA) and is the first part in a series of work (other parts coming later in the year from Mark and I) designed to enhance ASoC for modern mobile audio hardware. This patch series provides the following important features:- 1) Allow sound cards to have multiple CODECs and platform DMA controllers. 2) Allow CODEC drivers to support more than one CODEC device. 3) Allow all CODEC, platform DMA and DAI drivers to have platform data. This is achieved by: - 1) Separating the component struct driver data from the component struct device data. 2) Making all components kernel devices. (A side effect of this change is that we can now store all our component private data in struct device private data (like everyone else) and there is also now less pointer indirection for components.) The change mainly affects only the probe() and remove() sections of component drivers and the component enumeration section of the core. The core also now handles setting up the CODEC data structs too (meaning CODEC drivers dont have to do this). All component PCM, DAPM and Kcontrol operations remain unchanged. I've so far tested the new multi-component enumeration on the TI OMAP and Marvell PXA platforms with AC97, MFD and I2C based CODECs. I do however need ASoC platform maintainers to test on their systems (as I only have a subset of the hardware). Some ASOC platforms do need a little extra checking/fixups too :- o Samsung platform: required the most work to untangle the components. o i.MX platform: required work to de-couple components. o Injenic J4740 platform: I do not have a toolchain to even build this arch so it will also need someone to build it too. If you find your platform does not work then the most likely reason is that I'm missing a platform_device registration for your CODEC, DAI, etc. device in your arch board.c or device.c files. Please let me know or send a patch :) All the multi-component code is in my multi-component branch here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git This RFC post contains the changes to ASoC core only. I'll post the multi-component CODEC and platform changes as separate RFCs in order not to flood the list (as there are 38 patches in all). This patch series will eventually be rebased into a single patch for upstream so we dont break bisect. The subsequent patches for CODECs and platforms in this series can be found in the git branch above. The intention is to upstream this for 2.6.36 Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

5 14

[alsa-devel] Help with dummy.c (where/how to write?)
by Smilen Dimitrov 05 Jul '10

05 Jul '10

Hi all, I am trying to work on dummy.c: http://lxr.linux.no/#linux+v2.6.32/sound/drivers/dummy.c as an exercise, and I'd like to modify it so it behaves as an 'oscillator'; that is, when I choose 'dummy' as an input in say Audacity, and press Record - I'd like to be able to write something to the audio (pcm) buffer of the driver directly from the driver itself (say a single sample), and be able to observe those samples recorded in Audacity. So far, by adding some debug statements, the only thing I can figure, is that when 'Record' in Audacity is pressed, 'dummy_hrtimer_callback' is called continuously - but I cannot tell what functions to use to modify / write to output buffers (not even sure what the output buffers there are, I guess dpcm->substream->private_data). Could anyone point to me if there is a similar example existing already, or if not. provide me with some pointers on how to do this? Thanks in advance, Cheers!

1 0

[alsa-devel] [PATCH] ALSA: pcm_lib: avoid timing jitter in snd_pcm_read/write()
by David Dillow 05 Jul '10

05 Jul '10

When using poll() to wait for the next period -- or avail_min samples -- one gets a consistent delay for each system call that is usually just a little short of the selected period time. However, When using snd_pcm_read/write(), one gets a jittery delay that alternates between less than a millisecond and approximately two period times. This is caused by snd_pcm_lib_{read,write}1() transferring any available samples to the user's buffer and adjusting the application pointer prior to sleeping to the end of the current period. When the next period interrupt occurs, there is then less than avail_min samples remaining to be transferred in the period, so we end up sleeping until a second period occurs. This is solved by using runtime->twake as the number of samples needed for a wakeup in addition to selecting the proper wait queue to wake in snd_pcm_update_state(). This requires twake to be non-zero when used by snd_pcm_lib_{read,write}1() even if avail_min is zero. Signed-off-by: Dave Dillow <dave(a)thedillows.org> --- Hans, you may want this one as well for the SIS7019, as it gives more margin to deal with captured data on alternating calls into the stack. diff --git a/include/sound/pcm.h b/include/sound/pcm.h index dd76cde..83c6fa6 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -313,7 +313,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - unsigned int twake: 1; /* do transfer (!poll) wakeup */ + snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e9d98be..bcf95d3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -287,8 +287,11 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) - wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); + if (runtime->twake) { + if (avail >= runtime->twake) + wake_up(&runtime->tsleep); + } else if (avail >= runtime->control->avail_min) + wake_up(&runtime->sleep); return 0; } @@ -1707,7 +1710,7 @@ EXPORT_SYMBOL(snd_pcm_period_elapsed); * The available space is stored on availp. When err = 0 and avail = 0 * on the capture stream, it indicates the stream is in DRAINING state. */ -static int wait_for_avail_min(struct snd_pcm_substream *substream, +static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t *availp) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1757,7 +1760,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, avail = snd_pcm_playback_avail(runtime); else avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->control->avail_min) + if (avail >= runtime->twake) break; } _endloop: @@ -1820,7 +1823,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1833,7 +1836,9 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; } @@ -2042,7 +2047,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2060,7 +2065,9 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; if (!avail)

5 12

[alsa-devel] [PATCH] ASoC: DaVinci: More accurate calculation for clock divider for McBSP (I2S)
by Raffaele Recalcati 05 Jul '10

05 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> i2s_accurate_sck switch can be used to have a better approximate sampling frequency. The trade off is between more accurate clock (fast clock) and less accurate clock (slow clock). The waveform will be not symmetric. Probably it is possible to get a better algorithm for calculating the divider, trying to keep a slower clock as possible. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 33 ++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s.c | 28 ++++++++++++++++++++----- 2 files changed, 55 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 0847d21..5149abe 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -73,6 +73,39 @@ struct snd_platform_data { */ int clk_input_pin; + /* + * This flag works when both clock and FS are outputs for the cpu + * and makes clock more accurate (FS is not symmetrical and the + * clock is very fast. + * The clock becoming faster is named + * i2s continuous serial clocl (I2S_SCK) and it is an externally + * visible bit clock. + * + * first line : WordSelect + * second line : ContinuousSerialClock + * third line: SerialData + * + * SYMMETRICAL APPROACH: + * _______________________ LEFT + * _| RIGHT |______________________| + * _ _ _ _ _ _ _ _ + * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_ + * _ _ _ _ _ _ _ _ + * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_ + * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ + * + * ACCURATE CLOCK APPROACH: + * ______________ LEFT + * _| RIGHT |_______________________________| + * _ _ _ _ _ _ _ _ _ + * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| | + * _ _ _ _ dummy cycles + * _/ \_ ... _/ \_/ \_ ... _/ \__________________ + * \_/ \_/ \_/ \_/ + * + */ + bool i2s_accurate_sck; + /* McASP specific fields */ int tdm_slots; u8 op_mode; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 17f594f..e6dcd81 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -122,6 +122,7 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { }; struct davinci_mcbsp_dev { + struct device *dev; struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -154,6 +155,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; int clk_input_pin; + bool i2s_accurate_sck; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -296,7 +298,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, pcr |= DAVINCI_MCBSP_PCR_SCLKME; break; default: - dev_err(&pdev->dev, "bad clk_input_pin\n"); + dev_err(dev->dev, "bad clk_input_pin\n"); return -EINVAL; } @@ -447,11 +449,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SRGR_CLKSM; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); - /* symmetric waveforms */ - clk_div = freq / (mcbsp_word_length * 16) / - params->rate_num * params->rate_den; - srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * - 16 - 1); + if (dev->i2s_accurate_sck) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } clk_div &= 0xFF; srgr |= clk_div; break; @@ -662,6 +676,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; dev->clk_input_pin = pdata->clk_input_pin; + dev->i2s_accurate_sck = pdata->i2s_accurate_sck; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { @@ -694,6 +709,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; + dev->dev = &pdev->dev; davinci_i2s_dai.private_data = dev; davinci_i2s_dai.capture.dma_data = dev->dma_params; -- 1.7.0.4

1 0

Re: [alsa-devel] ALSA: hda - Don't check capture source mixer if no ADC is available
by Dan Carpenter 05 Jul '10

05 Jul '10

Commit fbe618f2168 "ALSA: hda - Don't check capture source mixer if no ADC is available" added support for when there is no ADC available and spec->adc_nids is NULL. If spec->adc_nids is NULL, can it cause a problem later when we call snd_hda_add_nid()? sound/pci/hda/patch_realtek.c +2643 alc_build_controls(96) error: we previously assumed 'spec->adc_nids' could be null. 2635 if (spec->cap_mixer) { 2636 const char *kname = kctl ? kctl->id.name : NULL; 2637 for (knew = spec->cap_mixer; knew->name; knew++) { 2638 if (kname && strcmp(knew->name, kname) == 0) 2639 continue; 2640 kctl = snd_hda_find_mixer_ctl(codec, knew->name); 2641 for (i = 0; kctl && i < kctl->count; i++) { 2642 err = snd_hda_add_nid(codec, kctl, i, 2643 spec->adc_nids[i]); ^^^^^^^^^^^^^^^^^ 2644 if (err < 0) 2645 return err; 2646 } 2647 } 2648 } regards, dan carpenter

2 1

[alsa-devel] linux vs c99 fixed width types
by Eliot Blennerhassett 05 Jul '10

05 Jul '10

Hi Takashi, & list it is quite likely that we will be moving our own codebase to use the C99 fixed width types rather than our own e.g. uint32_t vs HW32 etc. Currently HW32 is translated into u32 before I submit patches to you. Should I do the same for the C99 types, or let them pass through unaltered? regards -- Eliot Blennerhassett AudioScience Inc.

3 2

[alsa-devel] Update asihpi to hpi version 4.03.36
by linux＠audioscience.com 05 Jul '10

05 Jul '10

2 7

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 05 Jul '10

05 Jul '10

The following changes since commit e827e32efc313e1a061c81007e96fff439ccb802: Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2010-06-24 11:11:41 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Eric Bénard (1): ASoC: clean i.MX Kconfig Kuninori Morimoto (1): ASoC: ak4642: Add Digital Playback Volume control Mark Brown (1): Merge remote branch 'takashi/topic/asoc' into for-2.6.36 Vladimir Zapolskiy (4): ASoC: uda134x: replace a macro with a value in platform struct. ASoC: uda134x: fix bias level setup on initialization ASoC: uda134x: add DATA011 register found in codecs family ASoC: uda134x: correct bias level setup for codecs family include/sound/uda134x.h | 12 ++++++++ sound/soc/codecs/ak4642.c | 24 ++++++++++++++-- sound/soc/codecs/uda134x.c | 64 +++++++++++++++++++++++++++++-------------- sound/soc/codecs/uda134x.h | 5 ++- sound/soc/imx/Kconfig | 14 ++++----- 5 files changed, 85 insertions(+), 34 deletions(-)

2 2

[alsa-devel] PC Beeps not working with HDA Intel module
by Shérab 05 Jul '10

05 Jul '10

Dear all, The problem is on a computer with an HDA Intel sound card, Intel G45 DEVCTG chip and appears will all version s of the kernel. Running e.g. beep 440 produces a beep whose audible frequency is very low, definitely not 440 Hz. However, it is worth notiing that grub is able to produce a beep of the right frequency at boot time. Can this problem be fixed ? Should you need any additional bit of information or testing, please ask. Many thanks in advance for your help, Shérab.

2 1

[alsa-devel] Full duplex code sample?
by robert 05 Jul '10

05 Jul '10

Dear All, Is there an example code on how to program a full duplex audio transmission? I did search the internet but it is full of Paul Davis advising to use Jack instead of ALSA. I think I would rather use ALSA for my purposes. Cheers, Robert

2 1

[alsa-devel] [RFC][PATCH] snd-usb: improving support for CM6206
by Adrian Pardini 05 Jul '10

05 Jul '10

Hi all, I'm working in my spare time to get better support for this chip. Right now it works but some features present in the datasheet are missing. This patch gives the option to select the source for the "headphone" jack. I have some concerns with the way I accomplished this and I'd like if someone could kick me in the right direction. In particular with the caching of some register values. Added another field to usb_mixer_interface, maybe it's not the best place. Also, according to the datasheet, in order to read back the register I must issue a request as done in snd_usb_cm106_write_int_reg() and the result will come in the HID input report. For this it doesn't matter but I'm planning on making some controls for spdif and I'll need it. Where should I tap to get that information?. To be done (I need help): * Mapping of streams: Front is ok. Rear_L jack is assigned to "Center", Rear_R jack gets "Speaker Woofer". "Center/Bass" jack is managed with the mixer "Speaker". Traced the circuit and the connections are fine, how do I switch them from ALSA? (nb: not with a ttable, from the driver) To be done (I think I'll somehow get this done on my own): * Input levels are reported with more resolution than the real. Add another case to build_feature_ctl(). Not very important now. * Engaging of the "mute" controls: now I can't mute left or right channel without muting everything. Have to play more with mixer_maps. Kind regards and thanks a lot for your time. Signed off by: Adrian Pardini <adrian.pardini(a)solar.org.ar> --- diff --git a/alsa-kernel/usb/mixer.h b/alsa-kernel/usb/mixer.h index 26c636c..291ba4b 100644 --- a/alsa-kernel/usb/mixer.h +++ b/alsa-kernel/usb/mixer.h @@ -22,6 +22,7 @@ struct usb_mixer_interface { u8 audigy2nx_leds[3]; u8 xonar_u1_status; + u16 cm6206reg2; }; #define MAX_CHANNELS 10 /* max logical channels */ diff --git a/usb/mixer_quirks.c b/usb/mixer_quirks.c index 794cdbd..8adeef0 100644 --- a/usb/mixer_quirks.c +++ b/usb/mixer_quirks.c @@ -360,6 +360,129 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } +/* + * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely + * documented in the device's data sheet. + */ +int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) +{ + u8 buf[4]; + buf[0] = 0x20; + buf[1] = value & 0xff; + buf[2] = (value >> 8) & 0xff; + buf[3] = reg; + return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, + USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, + 0, 0, &buf, 4, 1000); +} + +static int snd_cm6206_headphone_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = kcontrol->private_value; + return 0; +} + +static int snd_cm6206_headphone_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + struct usb_device *dev = mixer->chip->dev; + + u16 reg2 = mixer->cm6206reg2; + u16 idx = kcontrol->private_value = ucontrol->value.integer.value[0]; + + reg2 = (u16)( (((1<<7) | (idx<<5))<<8) | (reg2 & 0x1800) ); + + snd_usb_cm106_write_int_reg(dev, 2, reg2); + mixer->cm6206reg2 = reg2; + return 1; +} + +static int snd_cm6206_headphone_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = kcontrol->private_value & (1<<11); + ucontrol->value.integer.value[1] = kcontrol->private_value & (1<<12); + return 0; +} + +static int snd_cm6206_headphone_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + struct usb_device *dev = mixer->chip->dev; + + u16 reg2 = mixer->cm6206reg2; + u16 mute; + + mute = ucontrol->value.integer.value[1] ? (1<<12) : 0; + mute |= ucontrol->value.integer.value[0] ? (1<<11) : 0; + kcontrol->private_value = mute; + + reg2 = (u16)(mute | (reg2 & ~0x1800)); + + snd_usb_cm106_write_int_reg(dev, 2, reg2); + mixer->cm6206reg2 = reg2; + return 1; +} + +static int snd_cm6206_headphone_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const names[] = { + "Side", + "Surround", + "Center", + "Front" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, names[uinfo->value.enumerated.item]); + return 0; +} + +#define snd_cm6206_headphone_mute_info snd_ctl_boolean_stereo_info + +static struct snd_kcontrol_new cm6206_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "Headphone Source", + .info = snd_cm6206_headphone_source_info, + .get = snd_cm6206_headphone_source_get, + .put = snd_cm6206_headphone_source_put, + .private_value = 3 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "Headphone", + .info = snd_cm6206_headphone_mute_info, + .get = snd_cm6206_headphone_mute_get, + .put = snd_cm6206_headphone_mute_put, + .private_value = 24 + } +}; + +static int snd_cm6206_controls_create(struct usb_mixer_interface *mixer) +{ + int i, err; + + for (i=0; i < ARRAY_SIZE(cm6206_controls); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&cm6206_controls[i], mixer)); + if (err < 0) + return err; + } + mixer->cm6206reg2 = 0xf800; /* Headphone source=front, no mute. */ + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err; @@ -385,6 +508,11 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) return err; } + if (mixer->chip->usb_id == USB_ID(0x0d8c, 0x0102)) { + err = snd_cm6206_controls_create(mixer); + if (err < 0) + return err; + } return 0; } @@ -415,4 +543,3 @@ void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, break; } } - diff --git a/alsa-kernel/usb/mixer_quirks.h b/alsa-kernel/usb/mixer_quirks.h index bdbfab0..bc5d577 100644 --- a/alsa-kernel/usb/mixer_quirks.h +++ b/alsa-kernel/usb/mixer_quirks.h @@ -9,5 +9,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid); +int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value); + #endif /* SND_USB_MIXER_QUIRKS_H */ diff --git a/usb/quirks.c b/usb/quirks.c index bf97ccd..ab57d58 100644 --- a/usb/quirks.c +++ b/usb/quirks.c @@ -376,22 +376,6 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) return 0; } -/* - * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely - * documented in the device's data sheet. - */ -static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) -{ - u8 buf[4]; - buf[0] = 0x20; - buf[1] = value & 0xff; - buf[2] = (value >> 8) & 0xff; - buf[3] = reg; - return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, - USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, - 0, 0, &buf, 4, 1000); -} - static int snd_usb_cm106_boot_quirk(struct usb_device *dev) { /* -- Adrian. http://elesquinazotango.com.ar http://www.noalcodigodescioli.blogspot.com/

1 0

Re: [alsa-devel] [PATCH 2/6] sound/oss: convert to unlocked_ioctl
by Sam Ravnborg 04 Jul '10

04 Jul '10

> > diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c > index c1070e3..3547450 100644 > --- a/sound/oss/au1550_ac97.c > +++ b/sound/oss/au1550_ac97.c > @@ -824,22 +824,26 @@ mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd, > return codec->mixer_ioctl(codec, cmd, arg); > } > > -static int > -au1550_ioctl_mixdev(struct inode *inode, struct file *file, > - unsigned int cmd, unsigned long arg) > +static long > +au1550_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) > { > struct au1550_state *s = (struct au1550_state *)file->private_data; > struct ac97_codec *codec = s->codec; > + int ret; > + > + lock_kernel(); > + ret = mixdev_ioctl(codec, cmd, arg); > + unlock_kernel(); > > - return mixdev_ioctl(codec, cmd, arg); > + return ret; > } General comment... In several places you declare a local variable of type "int", but the function return a long. I know that mixdev_ioctl() return an int so nothing is lost but it just looks wrong. > diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c > index 3f3c3f7..b5464fb 100644 > --- a/sound/oss/dmasound/dmasound_core.c > +++ b/sound/oss/dmasound/dmasound_core.c > @@ -337,8 +337,8 @@ static int mixer_release(struct inode *inode, struct file *file) > unlock_kernel(); > return 0; > } > -static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd, > - u_long arg) > + > +static long mixer_ioctl(struct file *file, u_int cmd, u_long arg) > { > if (_SIOC_DIR(cmd) & _SIOC_WRITE) > mixer.modify_counter++; > @@ -362,11 +362,22 @@ static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd, > return -EINVAL; > } > > +static long mixer_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) > +{ > + int ret; > + > + lock_kernel(); > + ret = mixer_ioctl(file, cmd, arg); > + unlock_kernel(); > + > + return ret; > +} Here it looks like a potential but. mixer_ioctl() return a long which is stored in an int and then we return a long. > -static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd, > - u_long arg) > +static long sq_ioctl(struct file *file, u_int cmd, u_long arg) > { > int val, result; > u_long fmt; > @@ -1114,18 +1124,29 @@ static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd, > return IOCTL_OUT(arg,val); > > default: > - return mixer_ioctl(inode, file, cmd, arg); > + return mixer_ioctl(file, cmd, arg); > } > return -EINVAL; > } > > +static long sq_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) > +{ > + int ret; > + > + lock_kernel(); > + ret = sq_ioctl(file, cmd, arg); > + unlock_kernel(); > + > + return ret; > +} ditto: long -> int -> long Sam

2 1

[alsa-devel] ESI DuaFire Vendor Information
by Florian Scholz 04 Jul '10

04 Jul '10

Hi I like to implement a driver for the ESI DuaFire Interface and need the information from ESI. Andreas Preuss from ESI wrote me (translated): thanks you for your message. every information that you need for the development of a firewire audio driver are public. the information are accessable for the alsa-project and maybe the ffado project.

2 3

[alsa-devel] [PATCH 1/3] ASoC: codec: Add WM8987 device id to WM8750 driver
by Maurus Cuelenaere 04 Jul '10

04 Jul '10

The WM8987 codec is register compatible with the WM8750, so just add it to the SPI and I²C device table. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/soc/codecs/wm8750.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 9407e19..e2c05e3 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -884,6 +884,7 @@ static int wm8750_i2c_remove(struct i2c_client *client) static const struct i2c_device_id wm8750_i2c_id[] = { { "wm8750", 0 }, + { "wm8987", 0 }, /* WM8987 is register compatible with WM8750 */ { } }; MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); @@ -925,14 +926,22 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id wm8750_spi_id[] = { + { "wm8750", 0 }, + { "wm8987", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, wm8750_spi_id); + static struct spi_driver wm8750_spi_driver = { .driver = { - .name = "wm8750", + .name = "WM8750 SPI Codec", .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = wm8750_spi_probe, .remove = __devexit_p(wm8750_spi_remove), + .id_table = wm8750_spi_id, }; #endif -- 1.7.0.4

2 3

[alsa-devel] Question about snd_pcm_sw_params_set_start_threshold
by Christian Gmeiner 03 Jul '10

03 Jul '10

Hi all, I am running into some troubles with my linux plugin for the vdr. I want to play audio in sync of video. My current solution is to buffer about 500ms of video on the output device (pts based FIFO). I can play audio with alsa, but sometimes I am not in sync with the video. So I want to set start threshold to 0 to disable automatic playback start. Instead I want to buffer audio and start playback manual, if pts of video is near current scr clock of output device. What are my options to manually start playback? thanks, -- Christian Gmeiner, MSc

1 0

[alsa-devel] [PATCH] sis7019: fix capture issues with multiple periods per buffer
by David Dillow 03 Jul '10

03 Jul '10

When using a timing voice to clock out periods during capture, the driver would slowly loose synchronization and never catch up, eventually reaching a point where it no longer generated interrupts. To avoid this situation, the virtual period clocking was changed to shorten the next timing period when our timing voice falls too far behind the capture voice. In addition, the first virtual period for the timing voice was slightly too short, causing the timing voice to initially be ahead of the capture voice. While tracking down this problem, I noticed that the expected sample offset was being incorrectly initialized, causing an overrun to be incorrectly reported when the timing voice happened to be perfectly synchronized. Reported-by: Hans Schou <linux(a)schou.dk> Signed-off-by: Dave Dillow <dave(a)thedillows.org> --- This has survived overnight testing at 44.1 kHz/16 bit/mono without issue. diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 9cc1b5a..614ff6e 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -264,11 +264,13 @@ static void sis_update_voice(struct voice *voice) * if using small periods. * * If we're less than 9 samples behind, we're on target. + * Otherwise, shorten the next vperiod by the amount we've + * been delayed. */ if (sync > -9) voice->vperiod = voice->sync_period_size + 1; else - voice->vperiod = voice->sync_period_size - 4; + voice->vperiod = voice->sync_period_size + sync + 10; if (voice->vperiod < voice->buffer_size) { sis_update_sso(voice, voice->vperiod); @@ -736,7 +738,7 @@ static void sis_prepare_timing_voice(struct voice *voice, period_size = buffer_size; /* Initially, we want to interrupt just a bit behind the end of - * the period we're clocking out. 10 samples seems to give a good + * the period we're clocking out. 12 samples seems to give a good * delay. * * We want to spread our interrupts throughout the virtual period, @@ -747,7 +749,7 @@ static void sis_prepare_timing_voice(struct voice *voice, * * This is all moot if we don't need to use virtual periods. */ - vperiod = runtime->period_size + 10; + vperiod = runtime->period_size + 12; if (vperiod > period_size) { u16 tail = vperiod % period_size; u16 quarter_period = period_size / 4; @@ -776,7 +778,7 @@ static void sis_prepare_timing_voice(struct voice *voice, */ timing->flags |= VOICE_SYNC_TIMING; timing->sync_base = voice->ctrl_base; - timing->sync_cso = runtime->period_size - 1; + timing->sync_cso = runtime->period_size; timing->sync_period_size = runtime->period_size; timing->sync_buffer_size = runtime->buffer_size; timing->period_size = period_size;

2 4

Re: [alsa-devel] [PATCH] - arecord - remove "front" and related devices from the output of list-pcm option of arecord
by Jaroslav Kysela 03 Jul '10

03 Jul '10

On Thu, 1 Jul 2010, Raymond Yau wrote: > Replace "front" device by "hw" device and remove "rear", "center_lfe" , > "side" , "surroundxx" from the output of "arecord -L" since "front" device > cannot be used for capturing and the front device is for playback only . NAK. Fix alsa-lib/configuration to return proper "IOID". Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 5

[alsa-devel] [PATCH 3/3] ARM: S3C64XX: Add audio support to SmartQ
by Maurus Cuelenaere 03 Jul '10

03 Jul '10

This adds the I²C board information for the WM8987 used in the SmartQ as audio codec and adds the I²C/I²S platform drivers. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- arch/arm/mach-s3c64xx/mach-smartq.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s3c64xx/mach-smartq.c b/arch/arm/mach-s3c64xx/mach-smartq.c index 028d080..5d8673b 100644 --- a/arch/arm/mach-s3c64xx/mach-smartq.c +++ b/arch/arm/mach-s3c64xx/mach-smartq.c @@ -199,6 +199,9 @@ static struct platform_device smartq_lcd_power_device = { .dev.platform_data = &smartq_lcd_power_data, }; +static struct i2c_board_info smartq_i2c_devs[] __initdata = { + { I2C_BOARD_INFO("wm8987", 0x1a), }, +}; static struct platform_device *smartq_devices[] __initdata = { &s3c_device_hsmmc1, /* Init iNAND first, ... */ @@ -213,6 +216,7 @@ static struct platform_device *smartq_devices[] __initdata = { &s3c_device_timer[1], &s3c_device_ts, &s3c_device_usb_hsotg, + &s3c64xx_device_iis0, &smartq_backlight_device, &smartq_lcd_power_device, &smartq_usb_otg_vbus_dev, @@ -354,6 +358,9 @@ void __init smartq_machine_init(void) s3c_sdhci2_set_platdata(&smartq_internal_hsmmc_pdata); s3c24xx_ts_set_platdata(&smartq_touchscreen_pdata); + i2c_register_board_info(0, smartq_i2c_devs, + ARRAY_SIZE(smartq_i2c_devs)); + WARN_ON(smartq_power_off_init()); WARN_ON(smartq_usb_host_init()); WARN_ON(smartq_usb_otg_init()); -- 1.7.0.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: Add SmartQ sound driver
by Maurus Cuelenaere 03 Jul '10

03 Jul '10

This adds sound support for the SmartQ board. The hardware consists of a S3C6410 coupled with a WM8987 over I²S. As there's no driver for the WM8987 currently, this uses the WM8750 driver (which should be register compatible). Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/soc/s3c24xx/Kconfig | 6 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smartq_wm8987.c | 300 +++++++++++++++++++++++++++++++++++++ 3 files changed, 308 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smartq_wm8987.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2a7cc22..4f46ba0 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -125,3 +125,9 @@ config SND_SOC_SMDK_WM9713 select SND_S3C_SOC_AC97 help Sat Y if you want to add support for SoC audio on the SMDK. + +config SND_S3C64XX_SOC_SMARTQ + tristate "SoC I2S Audio support for SmartQ board" + depends on SND_S3C24XX_SOC && MACH_SMARTQ + select SND_S3C64XX_SOC_I2S + select SND_SOC_WM8750 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 81d8dc5..50172c3 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -29,6 +29,7 @@ snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o +snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -41,3 +42,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o +obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c new file mode 100644 index 0000000..ef6183f --- /dev/null +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -0,0 +1,300 @@ +/* sound/soc/s3c24xx/smartq_wm8987.c + * + * Copyright 2010 Maurus Cuelenaere <mcuelenaere(a)gmail.com> + * + * Based on smdk6410_wm8987.c + * Copyright 2007 Wolfson Microelectronics PLC. - linux(a)wolfsonmicro.com + * Graeme Gregory - graeme.gregory(a)wolfsonmicro.com + * + * 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; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#include "../codecs/wm8750.h" + +/* + * According to http://www.wolfsonmicro.com/products/WM8987 the WM8987 is + * register compatible with the WM8750, so I'm currently using that one as + * there's no WM8987 codec driver. + */ + +static struct snd_soc_card snd_soc_smartq; + +static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret; + + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c_i2sv2_get_clock(cpu_dai)); + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + clk = 11289600; + 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; + + /* set cpu DAI configuration */ + 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; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SmartQ WM8987 HiFi DAI operations. + */ +static struct snd_soc_ops smartq_hifi_ops = { + .hw_params = smartq_hifi_hw_params, +}; + +static struct snd_soc_jack smartq_jack; + +static struct snd_soc_jack_pin smartq_jack_pins[] = { + /* Disable speaker when headphone is plugged in */ + { + .pin = "Internal Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = true, + }, +}; + +static struct snd_soc_jack_gpio smartq_jack_gpios[] = { + { + .gpio = S3C64XX_GPL(12), + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 200, + }, +}; + +static const struct snd_kcontrol_new wm8987_smartq_controls[] = { + SOC_DAPM_PIN_SWITCH("Internal Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), +}; + +static int smartq_speaker_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8987_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Internal Speaker", smartq_speaker_event), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LOUT2"}, + {"Headphone Jack", NULL, "ROUT2"}, + + {"Internal Speaker", NULL, "LOUT2"}, + {"Internal Speaker", NULL, "ROUT2"}, + + {"Mic Bias", NULL, "Internal Mic"}, + {"LINPUT2", NULL, "Mic Bias"}, +}; + +static int smartq_wm8987_init(struct snd_soc_codec *codec) +{ + int err = 0; + + /* Add SmartQ specific widgets */ + snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets, + ARRAY_SIZE(wm8987_dapm_widgets)); + + /* add SmartQ specific controls */ + err = snd_soc_add_controls(codec, wm8987_smartq_controls, + ARRAY_SIZE(wm8987_smartq_controls)); + + if (err < 0) + return err; + + /* setup SmartQ specific audio path */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "LINPUT1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "ROUT1"); + + /* set endpoints to default off mode */ + snd_soc_dapm_enable_pin(codec, "Internal Speaker"); + snd_soc_dapm_enable_pin(codec, "Internal Mic"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + err = snd_soc_dapm_sync(codec); + if (err) + return err; + + /* Headphone jack detection */ + err = snd_soc_jack_new(&snd_soc_smartq, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack); + if (err) + return err; + + err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), + smartq_jack_pins); + if (err) + return err; + + err = snd_soc_jack_add_gpios(&smartq_jack, + ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + return err; +} + +static struct snd_soc_dai_link smartq_dai[] = { + { + .name = "wm8987", + .stream_name = "SmartQ Hi-Fi", + .cpu_dai = &s3c64xx_i2s_dai[0], + .codec_dai = &wm8750_dai, + .init = smartq_wm8987_init, + .ops = &smartq_hifi_ops, + }, +}; + +static struct snd_soc_card snd_soc_smartq = { + .name = "SmartQ", + .platform = &s3c24xx_soc_platform, + .dai_link = smartq_dai, + .num_links = ARRAY_SIZE(smartq_dai), +}; + +static struct snd_soc_device smartq_snd_devdata = { + .card = &snd_soc_smartq, + .codec_dev = &soc_codec_dev_wm8750, +}; + +static struct platform_device *smartq_snd_device; + +static int __init smartq_init(void) +{ + int ret; + + if (!machine_is_smartq7() && !machine_is_smartq5()) { + pr_info("Only SmartQ is supported by this ASoC driver\n"); + return -ENODEV; + } + + smartq_snd_device = platform_device_alloc("soc-audio", -1); + if (!smartq_snd_device) + return -ENOMEM; + + platform_set_drvdata(smartq_snd_device, &smartq_snd_devdata); + smartq_snd_devdata.dev = &smartq_snd_device->dev; + + ret = platform_device_add(smartq_snd_device); + if (ret) { + platform_device_put(smartq_snd_device); + return ret; + } + + /* Initialise GPIOs used by amplifiers */ + ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown"); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n"); + goto err_unregister_device; + } + + /* Disable amplifiers */ + ret = gpio_direction_output(S3C64XX_GPK(12), 1); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n"); + goto err_free_gpio_amp_shut; + } + + dev_info(&smartq_snd_device->dev, "SmartQ WM8987 audio driver inited\n"); + + return 0; + +err_free_gpio_amp_shut: + gpio_free(S3C64XX_GPK(12)); +err_unregister_device: + platform_device_unregister(smartq_snd_device); + + return ret; +} + +static void __exit smartq_exit(void) +{ + snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + platform_device_unregister(smartq_snd_device); +} + +module_init(smartq_init); +module_exit(smartq_exit); + +/* Module information */ +MODULE_AUTHOR("Maurus Cuelenaere"); +MODULE_DESCRIPTION("ALSA SoC SmartQ WM8987"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

2 2

[alsa-devel] [PATCH 2/2] ARM: S3C64XX: Add audio support to SmartQ
by Maurus Cuelenaere 03 Jul '10

03 Jul '10

This adds the I²C board information for the WM8987 used in the SmartQ as audio codec and adds the I²C/I²S platform drivers. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- arch/arm/mach-s3c64xx/mach-smartq.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s3c64xx/mach-smartq.c b/arch/arm/mach-s3c64xx/mach-smartq.c index 028d080..ee77255 100644 --- a/arch/arm/mach-s3c64xx/mach-smartq.c +++ b/arch/arm/mach-s3c64xx/mach-smartq.c @@ -199,6 +199,9 @@ static struct platform_device smartq_lcd_power_device = { .dev.platform_data = &smartq_lcd_power_data, }; +static struct i2c_board_info smartq_i2c_devs[] __initdata = { + { I2C_BOARD_INFO("wm8750", 0x1a), }, +}; static struct platform_device *smartq_devices[] __initdata = { &s3c_device_hsmmc1, /* Init iNAND first, ... */ @@ -213,6 +216,7 @@ static struct platform_device *smartq_devices[] __initdata = { &s3c_device_timer[1], &s3c_device_ts, &s3c_device_usb_hsotg, + &s3c64xx_device_iis0, &smartq_backlight_device, &smartq_lcd_power_device, &smartq_usb_otg_vbus_dev, @@ -354,6 +358,9 @@ void __init smartq_machine_init(void) s3c_sdhci2_set_platdata(&smartq_internal_hsmmc_pdata); s3c24xx_ts_set_platdata(&smartq_touchscreen_pdata); + i2c_register_board_info(0, smartq_i2c_devs, + ARRAY_SIZE(smartq_i2c_devs)); + WARN_ON(smartq_power_off_init()); WARN_ON(smartq_usb_host_init()); WARN_ON(smartq_usb_otg_init()); -- 1.7.0.4

2 1

[alsa-devel] alsamixer-qt4 version 0.3 released
by Sebastian H. 01 Jul '10

01 Jul '10

Hello again The new version 0.3.0 of alsamixer-qt4 is available. Check it out at http://xwmw.org/alsamixer-qt4/ The files are at sourceforge http://sourceforge.net/projects/alsamixer-qt4/files/ Regards, Sebastian H.

1 0

[alsa-devel] [PATCH v3] ASoC: ak4642: Add Digital Playback Volume control
by Kuninori Morimoto 01 Jul '10

01 Jul '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 o remove "default volume" o remove explain from commit log sound/soc/codecs/ak4642.c | 24 +++++++++++++++++++++--- 1 files changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7528a54..b41599d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -36,6 +36,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -111,6 +112,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -204,7 +222,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -214,8 +231,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -548,6 +563,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -- 1.7.0.4

2 1

[alsa-devel] wrong decibel data?
by Nicolo' Chieffo 01 Jul '10

01 Jul '10

Hello, I have a laptop with an intel hda audio chipset (STAC92xx) My problem is that when I put pulseaudio volume to 14% (or less) I can't hear any audio coming from speakers or headphones: in fact the alsamixer volume is set to 0% I already filed a bug to pulseaudio [1] in which the developers say that this is a common problem which is usually caused by the ALSA audio driver which reports a wrong decibel value. [1] http://pulseaudio.org/ticket/580

9 95

[alsa-devel] [PATCH 3/3] ASoC: DaVinci: Added fast clock timing for McBSP (I2S)
by Raffaele Recalcati 01 Jul '10

01 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> i2s_fast_clock switch can be used to have a better approximate frequency. The waveform will be not symmetric. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++++ sound/soc/davinci/davinci-i2s.c | 24 +++++++++++++++++++----- 2 files changed, 26 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 314570d..2d8f8af 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,13 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not symmetrical, but sampling + * frequency is better approximated + */ + bool i2s_fast_clock; + /* To be used when cpu gets clock from external pin */ int clk_input_pin; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index e478be9..0af5bae 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -150,6 +150,7 @@ struct davinci_mcbsp_dev { */ unsigned enable_channel_combine:1; + bool i2s_fast_clock; unsigned int fmt; int clk_div; int clk_input_pin; @@ -443,11 +444,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SRGR_CLKSM; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); - /* symmetric waveforms */ - clk_div = freq / (mcbsp_word_length * 16) / - params->rate_num * params->rate_den; - srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * - 16 - 1); + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } clk_div &= 0xFF; srgr |= clk_div; break; @@ -643,6 +656,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->i2s_fast_clock = pdata->i2s_fast_clock; dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); -- 1.7.0.4

2 2

[alsa-devel] [PATCH 2/3] ASoC: DaVinci: Added selection of clk input pin for McBSP
by Raffaele Recalcati 01 Jul '10

01 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> When McBSP peripheral gets the clock from an external pin, there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR and MCBSP_CLKS. evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different hardware connection and I use MCBSP_CLKS, so I have added this possibility. This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm) Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 8 ++++++++ sound/soc/davinci/davinci-i2s.c | 28 +++++++++++++++++++++++----- 2 files changed, 31 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..314570d 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,9 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* To be used when cpu gets clock from external pin */ + int clk_input_pin; + /* McASP specific fields */ int tdm_slots; u8 op_mode; @@ -78,6 +81,11 @@ enum { MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; +enum { + MCBSP_CLKR = 0, /* DM365 */ + MCBSP_CLKS, +}; + #define INACTIVE_MODE 0 #define TX_MODE 1 #define RX_MODE 2 diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 319e8ca..e478be9 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -152,6 +152,7 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; + int clk_input_pin; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -278,11 +279,26 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: - /* McBSP CLKR pin is the input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + /* + * Selection of the clock input pin that is the + * input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + switch (dev->clk_input_pin) { + case MCBSP_CLKS: + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case MCBSP_CLKR: + pcr |= DAVINCI_MCBSP_PCR_SCLKME; + break; + default: + printk(KERN_ERR "%s:bad clk_input_pin\n", __func__); + return -EINVAL; + } + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -436,6 +452,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, srgr |= clk_div; break; case SND_SOC_DAIFMT_CBM_CFS: + /* Clock given on CLKS */ srgr = DAVINCI_MCBSP_SRGR_FSGM; clk_div = dev->clk_div - 1; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); @@ -626,6 +643,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)
by Raffaele Recalcati 01 Jul '10

01 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added two clocking options for dm365 McBSP peripheral when used with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock from external pin and generates frame sync). A slave clock management can be important when the external codec needs the system clock and the bit clock synchronized (tested with uda1345). This patch has been developed against the: http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- sound/soc/davinci/davinci-i2s.c | 92 +++++++++++++++++++++++++++++++++++---- 1 files changed, 83 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..319e8ca 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -68,16 +68,21 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE (1 << 31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE (1 << 31) #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM (1 << 29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,6 +149,9 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + unsigned int fmt; + int clk_div; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -254,10 +262,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = cpu_dai->private_data; unsigned int pcr; unsigned int srgr; + /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + dev->fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -372,6 +382,16 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + int srgr; + + dev->clk_div = div; + return 0; +} + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,8 +400,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; @@ -396,12 +416,44 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + freq = clk_get_rate(dev->clk); + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + case SND_SOC_DAIFMT_CBM_CFS: + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = dev->clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + break; + default: + /* Clock and frame sync given from external sources */ + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,12 +478,29 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -442,6 +511,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,6 +573,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; -- 1.7.0.4

2 1

[alsa-devel] [RFC 00/15] ASoC: multi-component - Platform Drivers
by Liam Girdwood 01 Jul '10

01 Jul '10

This is the third and final part of the ASoC multi-component series and focuses on the SoC Platform support. I've split the patches by SoC architecture and again all will be rebased later to avoid breaking git bisect. Most platform drivers were quite straight forward with most changes in the probe() and remove() areas. However, the Samsung and i.MX platform did require some extra work to decouple some of the components. Thanks Liam

5 22