June 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: dwc: Add missing __iomem annotations
by Mark Brown 28 Jun '12

28 Jun '12

Otherwise sparse gets very upset with us. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e667e2b..1bd042b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -71,12 +71,12 @@ struct dw_i2s_dev { int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; -static inline void i2s_write_reg(void *io_base, int reg, u32 val) +static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val) { writel(val, io_base + reg); } -static inline u32 i2s_read_reg(void *io_base, int reg) +static inline u32 i2s_read_reg(void __iomem *io_base, int reg) { return readl(io_base + reg); } -- 1.7.10

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Mark Brown 28 Jun '12

28 Jun '12

On Thu, Jun 28, 2012 at 05:26:58PM +0530, Laxman Dewangan wrote: > Other driver which can use this API are davinci, OMAP, samsung but > because they are not using the snd_dmaengine_pcm driver's api as of > now so not be very much useful for them. There's no reason why we can't factor out more common code - this is essentially what Lars-Peter was saying. > Should I send one patch per driver change or squash all 4 file > changes in one patch? Whatever will be easy or recommended? It doesn't really matter that much, but please take on board Lars-Peter's comments.

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Mark Brown 28 Jun '12

28 Jun '12

On Thu, Jun 28, 2012 at 12:31:26PM +0530, Laxman Dewangan wrote: > The pcm mmap for the substream dma buffer is done using the > dma_mmap_writecombine(). > Some of the chip sound driver like Spear, Tegra, Davinci, > ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this > mapping locally. > Add api in the snd dmaengine driver so that different client of > snd dmaengine can use this rather than implementing locally. OK, but I'd expect this to come along with one or more patches converting at least some of the users to the new API...

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[alsa-devel] kcontrol get method
by raj raja 28 Jun '12

28 Jun '12

Hi, Generally get and set methods of Kcontrol called via mixer controls. I want to update variable value based on my kcontrol set method. I could not call get method, like as normal functions. So any way to call particular kcontrol get method in driver itself Thanks Raj

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[alsa-devel] ALS100: no interrupts in Linux but works fine in DOS
by Ondrej Zary 27 Jun '12

27 Jun '12

Hello, I have a Freetek PRO32PNP card - an ISA card with ALS100 chip and onboard Crystal wavetable MIDI. The card works fine as SB16 in DOS - without any initialization utility, PnP BIOS configures the card to 0x220, irq5, dma1, dma5. However it does not work in Linux. Driver loads fine (snd-als100, even tried old OSS sb) but when I try to play anything, the player hangs without producing any sound. /proc/interrupts show that no interrupts are generated. Mixer works fine. The only suspicious thing is "unknown tag 0xff" and "invalid checksum" in dmesg (this is from patched kernel that shows size of the unknown tag): [ 0.159650] isapnp: Scanning for PnP cards... [ 0.246683] pnp 01:01.00: dependent set 0 (preferred) io min 0x220 max 0x220 align 0 size 16 flags 0x2 [ 0.246889] pnp 01:01.00: dependent set 0 (preferred) irq 5 flags 0x1 [ 0.247072] pnp 01:01.00: dependent set 0 (preferred) dma 5 (bitmask 0x20) flags 0x72 [ 0.247253] pnp 01:01.00: dependent set 0 (preferred) dma 1 (bitmask 0x2) flags 0x68 [ 0.247872] pnp 01:01.00: dependent set 1 (acceptable) io min 0x220 max 0x280 align 32 size 16 flags 0x0 [ 0.248078] pnp 01:01.00: dependent set 1 (acceptable) irq 5 7 9 10 flags 0x1 [ 0.248265] pnp 01:01.00: dependent set 1 (acceptable) dma 5 6 7 (bitmask 0xe0) flags 0x72 [ 0.248450] pnp 01:01.00: dependent set 1 (acceptable) dma 0 1 3 (bitmask 0xb) flags 0x68 [ 0.248963] pnp 01:01.00: dependent set 2 (functional) io min 0x100 max 0x3f0 align 16 size 16 flags 0x0 [ 0.249213] pnp 01:01.00: dependent set 2 (functional) irq 5 7 9 10 11 12 15 flags 0xf [ 0.249404] pnp 01:01.00: dependent set 2 (functional) dma 0 1 3 5 6 7 (bitmask 0xeb) flags 0x79 [ 0.249996] pnp 01:01.01: independent io min 0x388 max 0x388 align 0 size 8 flags 0x2 [ 0.250535] pnp 01:01.02: independent io min 0x200 max 0x200 align 0 size 2 flags 0x2 [ 0.251171] pnp 01:01.03: dependent set 0 (preferred) io min 0x330 max 0x330 align 0 size 2 flags 0x2 [ 0.251365] pnp 01:01.03: dependent set 0 (preferred) irq 9 flags 0x1 [ 0.251984] pnp 01:01.03: dependent set 1 (acceptable) io min 0x100 max 0x3f0 align 16 size 2 flags 0x0 [ 0.252244] pnp 01:01.03: dependent set 1 (acceptable) irq 5 7 9 10 11 12 15 flags 0xf [ 0.256256] pnp 01:01.03: unknown tag 0xff (card 1) size=18041, ignored [ 4.527093] 01:01: invalid checksum 0xad [ 4.527257] Switching to clocksource tsc [ 4.528594] 01:01: card 'ALS100 Media Audio Controller' [ 4.528699] isapnp: 1 Plug & Play card detected total I've even tried to disable Linux pnp completely (noisapnp pnpbios=off) and load snd-sb16 module manually (modprobe snd-sb16 isapnp=0) with exactly the same result (everything seems to be OK but there's no sound and interrupts). Any ideas? -- Ondrej Zary

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[alsa-devel] periods_max initialization in snd_usb_hardware
by Amit Nagal 27 Jun '12

27 Jun '12

Hi , snd_usb_hardware is defined in sound/usb/pcm.c as : static struct snd_pcm_hardware snd_usb_hardware = { ..... .buffer_bytes_max = 1024 * 1024, .period_bytes_min = 64, .period_bytes_max = 512 * 1024, .periods_min = 2, .periods_max = 1024, }; here periods_max is intialized with value = 1024 . but going by buffer_bytes_max and period_bytes_min value defined above , periods_max = ( buffer_bytes_max / period_bytes_min ) = 16384 . why this upper cap of 1024 is maintained for periods_max even though from calculation it comes as 16384 ? Thanx & Regards Amit Nagal

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[alsa-devel] how to update hw_ptr and appl_ptr
by Zhen Fu 27 Jun '12

27 Jun '12

Hi developer, I recently look the function snd_pcm_update_hw_ptr0 want to find how to update status->hw_ptr and control->appl_ptr. If you have to know this, can you tell me? Thank you very much. -- Best Regards, Zhen Fu

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[alsa-devel] [PATCH 1/2] ALSA: HDA: Support single 3-pin jack without VREF on the actual pin
by David Henningsson 27 Jun '12

27 Jun '12

Some ASUS device has a single 3-pin jack that can either be a mic or a headphone, but the pin does not have VREF capabilities. We've been told by Realtek to instead enable VREF on pin 0x18 in that case. BugLink: https://bugs.launchpad.net/bugs/1018262 Tested-by: Chih-Hsyuan Ho <chih.ho(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 49 ++++++++++++++++++++++++++++------------- 1 file changed, 34 insertions(+), 15 deletions(-) Alsa info: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1018262/+attachment/32… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5c81ee9..6174d4a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -303,6 +303,38 @@ static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx) static void call_update_outputs(struct hda_codec *codec); static void alc_inv_dmic_sync(struct hda_codec *codec, bool force); +/* for shared I/O, change the pin-control accordingly */ +static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + hda_nid_t pin = spec->autocfg.inputs[1].pin; + /* NOTE: this assumes that there are only two inputs, the + * first is the real internal mic and the second is HP/mic jack. + */ + + val = snd_hda_get_default_vref(codec, pin); + + /* This pin does not have vref caps - let's instead enable vref on pin 0x18 + instead, as suggested by Realtek */ + if (val == AC_PINCTL_VREF_HIZ) { + const hda_nid_t vref_pin = 0x18; + /* Sanity check pin 0x18 */ + if (get_wcaps_type(get_wcaps(codec, vref_pin)) == AC_WID_PIN && + get_defcfg_connect(snd_hda_codec_get_pincfg(codec, vref_pin)) == AC_JACK_PORT_NONE) { + unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin); + if (vref_val != AC_PINCTL_VREF_HIZ) + snd_hda_set_pin_ctl(codec, vref_pin, PIN_IN | (set_as_mic ? vref_val : 0)); + } + } + + val = set_as_mic ? val | PIN_IN : PIN_HP; + snd_hda_set_pin_ctl(codec, pin, val); + + spec->automute_speaker = !set_as_mic; + call_update_outputs(codec); +} + /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, unsigned int idx, bool force) @@ -329,21 +361,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; spec->cur_mux[adc_idx] = idx; - /* for shared I/O, change the pin-control accordingly */ - if (spec->shared_mic_hp) { - unsigned int val; - hda_nid_t pin = spec->autocfg.inputs[1].pin; - /* NOTE: this assumes that there are only two inputs, the - * first is the real internal mic and the second is HP jack. - */ - if (spec->cur_mux[adc_idx]) - val = snd_hda_get_default_vref(codec, pin) | PIN_IN; - else - val = PIN_HP; - snd_hda_set_pin_ctl(codec, pin, val); - spec->automute_speaker = !spec->cur_mux[adc_idx]; - call_update_outputs(codec); - } + if (spec->shared_mic_hp) + update_shared_mic_hp(codec, spec->cur_mux[adc_idx]); if (spec->dyn_adc_switch) { alc_dyn_adc_pcm_resetup(codec, idx); -- 1.7.9.5

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Re: [alsa-devel] emu10k1 driver can't allocate memory
by Takashi Iwai 27 Jun '12

27 Jun '12

At Sun, 20 May 2012 03:04:14 +0200, Mikael Magnusson wrote: > > Hi, (please CC me on replies, not subscribed) > > I recently upgraded to 3.3.6 from 2.6.33.15 (yeah, I know). Since then > I noticed sometimes sounds were sometimes missing, then I noticed > thousands of these in dmesg, > [37280.283484] max memory size is 0x7fffffff (addr = 0x1d5d70071038000)!! I guess you are enabling IOMMU? I guess it translates the address somehow in a way emu10k1 doesn't like. There are really little changes in emu10k1 driver itself, so if the kernel upgrade changes the behavior, it must be in the core part such like IOMMU handling. Try to boot without iommu and see whether it changes the behavior again. Takashi > [37280.283486] emu: failure page = 32 > I also checked asfxload -M output, right after boot it shows all the > synth memory free as you would expect, but after a while of usage, it > is close to 0 (around 50KiB or so). Trying to run a simple program > that writes to /dev/audio fails with > write(3</dev/audio>, > "\32\21\t\3\0\0\4\n\23\34&/6:;94,#\31\17\10\2\0\0\4\v\24\36(07"..., > 4096) = -1 ENOMEM (Cannot allocate memory) > You can also note the output from asfxload -M dropping rapidly while > the program is making sound before it fails. Native alsa apps seem to > fare slightly better as the memory decrease caused by them is actually > returned when they close the device again. > > Loading and unloading soundfonts seem to work as normal, the memory is > returned when the soundfont is unloaded afaict. (I have the patch > applied that makes it possible to load soundfonts (changes the DMA32 > zone to 2GB). I don't think this is the cause of my problem since the > given addresses are way way outside the 4GB range as well.) > > Note that even well before the reported memory is at 0, apps randomly > fail to open the sound output and are silent. Curiously espeak when > successful doesn't cause a decrease in available memory, but when it > fails, it causes about 100KiB drop. > > I can't (don't dare) bisect because this is my production machine, but > I can test patches, or check the major releases between 2.6.33 and > 3.3.6 if you have no idea what causes this. > > I'm not the only one with the problem either, > http://www.google.com/search?nfpr=1&tbs=li:1&q=emu%3A+failure+page+%3D+32+m… > > -- > Mikael Magnusson >

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[alsa-devel] Configuring an audio path within ALSA
by rashmi narayan 27 Jun '12

27 Jun '12

Dear ALSA Folks, I am a newbie to ALSA .I am trying to configure a path within the driver to open up /power up the path within the codec from MIC-IN to LineOUT. How can I do this within ALSA. Please send me some pointers. Thank you, Rashmi

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