December 2011 - Alsa-devel - mailman.alsa-project.org

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

22 Dec '11

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 snd_card_xxx() is obsolete...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

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

22 Dec '11

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

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

21 Dec '11

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

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

21 Dec '11

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

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

21 Dec '11

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

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

21 Dec '11

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

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

21 Dec '11

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

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

21 Dec '11

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

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

20 Dec '11

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

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

20 Dec '11

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

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