May 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add platform data and reset gpio handling
by Jarkko Nikula 05 May '10

05 May '10

Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- include/sound/tlv320aic3x.h | 17 +++++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 26 ++++++++++++++++++++++++++ 2 files changed, 43 insertions(+), 0 deletions(-) create mode 100644 include/sound/tlv320aic3x.h diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h new file mode 100644 index 0000000..e493a14 --- /dev/null +++ b/include/sound/tlv320aic3x.h @@ -0,0 +1,17 @@ +/* + * Platform data for Texas Instruments TLV320AIC3x codec + * + * Author: Jarkko Nikula <jhnikula(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#ifndef __TLV320AIC3x_H__ +#define __TLV320AIC3x_H__ + +struct aic3x_pdata { + int gpio_reset; +}; + +#endif \ No newline at end of file diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b2c2794..cdcfb6b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -38,6 +38,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/gpio.h> #include <linux/regulator/consumer.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -48,6 +49,7 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <sound/tlv320aic3x.h> #include "tlv320aic3x.h" @@ -65,6 +67,7 @@ struct aic3x_priv { struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES]; unsigned int sysclk; int master; + int gpio_reset; }; /* @@ -1279,6 +1282,10 @@ static int aic3x_unregister(struct aic3x_priv *aic3x) snd_soc_unregister_dai(&aic3x_dai); snd_soc_unregister_codec(&aic3x->codec); + if (aic3x->gpio_reset >= 0) { + gpio_set_value(aic3x->gpio_reset, 0); + gpio_free(aic3x->gpio_reset); + } regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); @@ -1303,6 +1310,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, { struct snd_soc_codec *codec; struct aic3x_priv *aic3x; + struct aic3x_pdata *pdata = i2c->dev.platform_data; int ret, i; aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); @@ -1319,6 +1327,15 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, aic3x); + aic3x->gpio_reset = -1; + if (pdata && pdata->gpio_reset >= 0) { + ret = gpio_request(pdata->gpio_reset, "tlv320aic3x reset"); + if (ret != 0) + goto err_gpio; + aic3x->gpio_reset = pdata->gpio_reset; + gpio_direction_output(aic3x->gpio_reset, 0); + } + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; @@ -1336,11 +1353,20 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, goto err_enable; } + if (aic3x->gpio_reset >= 0) { + udelay(1); + gpio_set_value(aic3x->gpio_reset, 1); + } + return aic3x_register(codec); err_enable: regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); err_get: + regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); + if (aic3x->gpio_reset >= 0) + gpio_free(aic3x->gpio_reset); +err_gpio: kfree(aic3x); return ret; } -- 1.7.0

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[alsa-devel] [PATCH] ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite P500-PSPGSC-01800T
by Daniel Chen 05 May '10

05 May '10

Hi, Attached is another patch against sound-2.6 master HEAD (based on the previous patch from yesterday) that makes the OR's machine have audible sound by default. Please apply. Best, -Dan

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[alsa-devel] [PATCH] ALSA: hda: Use olpc-xo-1_5 quirk for Toshiba Satellite Pro T130-15F
by Daniel Chen 05 May '10

05 May '10

Hi, Attached is a patch against sound-2.6 master HEAD that makes the OR's machine have audible sound by default. Please apply. Best, -Dan

2 1

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 05 May '10

05 May '10

The following changes since commit cb7b76961f73e4ae934d44f7b2e7ba974442f2fe: Takashi Iwai (1): Merge branch 'for-2.6.35' of git://git.kernel.org/.../lrg/asoc-2.6 into topic/asoc 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 Mark Brown (2): ASoC: Remove redundant WM8960 SYSCLKSEL clkdiv option ASoC: Add WM9090 amplifier driver include/sound/wm9090.h | 28 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8960.c | 4 - sound/soc/codecs/wm8960.h | 1 - sound/soc/codecs/wm9090.c | 773 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9090.h | 715 +++++++++++++++++++++++++++++++++++++++++ 7 files changed, 1522 insertions(+), 5 deletions(-) create mode 100644 include/sound/wm9090.h create mode 100644 sound/soc/codecs/wm9090.c create mode 100644 sound/soc/codecs/wm9090.h

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[alsa-devel] [PATCH RFC] ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582
by Anisse Astier 05 May '10

05 May '10

Add a quirk for all-in-one computer Dell Inspiron One 19 Touch to have proper HP and Mic support. Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi, I have a Dell 19T all-in-one computer, which has audio problems by default: - speakers works OK - no jack detection, no audio in HP - internal or external Mic not working If I specify model=ideapad, the problems go away. But Mic boost appears as an "Analog Mic Boost Capture Enum" between 0 and 40dB instead of a normal bar control in alsamixer. I also tested model=dell-vostro, laptop, and dell-laptop, and they give worse results. There's no BIOS update available from vendor ATM. I'm using kernel tree sound-2.6 on branch master. I'm attaching alsa-info.sh output with and without model=ideapad in addition to the patch. Regards, Anisse sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 509a104..8446273 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2967,6 +2967,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), {} -- 1.7.0.6

2 1

[alsa-devel] sound/core/timer.c: inconsistent lock state reported for tu->qlock
by Stefan Richter 05 May '10

05 May '10

Hi, I got the following log from a rarely used PC that I just updated from 2.6.34-rc4 to 2.6.34-rc5. I don't know though under which kernel I last used the soundcard of it; could have been 2.6.33, could have been a .34-rc. It is an x86 Mac mini with 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Kernel is vanilla except for various firewire patches, which should not matter to this issue. ================================= [ INFO: inconsistent lock state ] 2.6.34-rc5 #5 --------------------------------- inconsistent {HARDIRQ-ON-W} -> {IN-HARDIRQ-W} usage. dolphin/4003 [HC1[1]:SC0[0]:HE0:SE1] takes: (&(&tu->qlock)->rlock){?.+...}, at: [<f84ec472>] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] {HARDIRQ-ON-W} state was registered at: [<c1048de9>] __lock_acquire+0x654/0x1482 [<c1049c73>] lock_acquire+0x5c/0x73 [<c125ac3e>] _raw_spin_lock+0x25/0x34 [<f84ec370>] snd_timer_user_ccallback+0x55/0x95 [snd_timer] [<f84ecc4b>] snd_timer_notify1+0x53/0xca [snd_timer] [<f84ecdbe>] _snd_timer_stop+0xfc/0x106 [snd_timer] [<f84eceab>] snd_timer_stop+0x13/0x42 [snd_timer] [<f84ee2dc>] snd_timer_user_ioctl+0xa27/0xaf5 [snd_timer] [<c1092c2f>] vfs_ioctl+0x22/0x87 [<c10931aa>] do_vfs_ioctl+0x472/0x4ac [<c1093210>] sys_ioctl+0x2c/0x45 [<c1002850>] sysenter_do_call+0x12/0x36 irq event stamp: 2042234 hardirqs last enabled at (2042233): [<c1002883>] sysenter_exit+0xf/0x18 hardirqs last disabled at (2042234): [<c1002d67>] common_interrupt+0x27/0x34 softirqs last enabled at (2042226): [<c102cfc3>] __do_softirq+0x102/0x10a softirqs last disabled at (2042221): [<c102cff6>] do_softirq+0x2b/0x43 other info that might help us debug this: 2 locks held by dolphin/4003: #0: (snd_pcm_link_rwlock){.-....}, at: [<f8eb0a35>] snd_pcm_period_elapsed+0x2a/0xaf [snd_pcm] #1: (&(&substream->self_group.lock)->rlock){-.....}, at: [<f8eb0a3f>] snd_pcm_period_elapsed+0x34/0xaf [snd_pcm] stack backtrace: Pid: 4003, comm: dolphin Tainted: G W 2.6.34-rc5 #5 Call Trace: [<c125882d>] ? printk+0xf/0x12 [<c104761b>] print_usage_bug+0x14e/0x15a [<c10478c8>] mark_lock+0x2a1/0x4a7 [<c1048066>] ? check_usage_forwards+0x0/0xb0 [<c1048d69>] __lock_acquire+0x5d4/0x1482 [<c1049c08>] ? __lock_acquire+0x1473/0x1482 [<c1049c73>] lock_acquire+0x5c/0x73 [<f84ec472>] ? snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] [<c125ac3e>] _raw_spin_lock+0x25/0x34 [<f84ec472>] ? snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] [<f84ec472>] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] [<f84ec44a>] ? snd_timer_user_tinterrupt+0x0/0x132 [snd_timer] [<f84ed2e1>] snd_timer_interrupt+0x213/0x264 [snd_timer] [<c125ac45>] ? _raw_spin_lock+0x2c/0x34 [<f8eb0a7d>] snd_pcm_period_elapsed+0x72/0xaf [snd_pcm] [<f8f733bb>] azx_interrupt+0x8d/0x109 [snd_hda_intel] [<c1057fb0>] handle_IRQ_event+0x1d/0xa2 [<c105980b>] handle_fasteoi_irq+0x79/0xb2 [<c10046ff>] handle_irq+0x3b/0x48 [<c1003f3d>] do_IRQ+0x45/0x9f [<c1002d6e>] common_interrupt+0x2e/0x34 [<c1250000>] ? piix_init_one+0x1d2/0x628 -- Stefan Richter -=====-==-=- -=-- ==-== http://arcgraph.de/sr/

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[alsa-devel] [PATCH] ASoC: omap: Add basic audio support for Nokia RX-51/N900
by Jarkko Nikula 05 May '10

05 May '10

This patch adds support for integrated stereo speakers and digital microphone found on Nokia RX-51 hardware. This is a cut down version based on Maemo kernel sources and earlier patchset by Eduardo Valentin et al. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.ht… Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Eduardo Valentin <eduardo.valentin(a)nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- This is for sound-2.6.git. This is independent and boot safe patch but audio codec doesn't probe without a few patches to arch/arm/mach-omap2/board-rx51-peripherals.c. See http://marc.info/?l=linux-omap&m=127290212826770&w=2 Some audio features would need more work and some, even being trivial to add, need more testing are there support for them in mainline. Thus this heavily simplified and little bit cleaned up version. --- sound/soc/omap/Kconfig | 10 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/rx51.c | 305 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 317 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/rx51.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index f11963c..83be4a7 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -18,6 +18,16 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_RX51 + tristate "SoC Audio support for Nokia RX-51" + depends on SND_OMAP_SOC && MACH_NOKIA_RX51 + select OMAP_MCBSP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on Nokia RX-51 + hardware. This is also known as Nokia N900 product. + config SND_OMAP_SOC_AMS_DELTA tristate "SoC Audio support for Amstrad E3 (Delta) videophone" depends on SND_OMAP_SOC && MACH_AMS_DELTA diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0bc00ca..3a75755 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-rx51-objs := rx51.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o @@ -22,6 +23,7 @@ snd-soc-zoom2-objs := zoom2.o snd-soc-igep0020-objs := igep0020.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c new file mode 100644 index 0000000..425362b --- /dev/null +++ b/sound/soc/omap/rx51.c @@ -0,0 +1,305 @@ +/* + * rx51.c -- SoC audio for Nokia RX-51 + * + * Copyright (C) 2008 - 2009 Nokia Corporation + * + * Contact: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * Eduardo Valentin <eduardo.valentin(a)nokia.com> + * Jarkko Nikula <jhnikula(a)gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic3x.h" + +#define RX51_CODEC_RESET_GPIO 60 +/* + * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This + * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c + */ +#define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7) + +static int rx51_spk_func; +static int rx51_dmic_func; + +static void rx51_ext_control(struct snd_soc_codec *codec) +{ + if (rx51_spk_func) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); + if (rx51_dmic_func) + snd_soc_dapm_enable_pin(codec, "DMic"); + else + snd_soc_dapm_disable_pin(codec, "DMic"); + + snd_soc_dapm_sync(codec); +} + +static int rx51_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->card->codec; + + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); + rx51_ext_control(codec); + + return 0; +} + +static int rx51_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set the codec system clock for DAC and ADC */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000, + SND_SOC_CLOCK_IN); +} + +static struct snd_soc_ops rx51_ops = { + .startup = rx51_startup, + .hw_params = rx51_hw_params, +}; + +static int rx51_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_spk_func; + + return 0; +} + +static int rx51_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_spk_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_spk_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + +static int rx51_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 1); + else + gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 0); + + return 0; +} + +static int rx51_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_dmic_func; + + return 0; +} + +static int rx51_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_dmic_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_dmic_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + +static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), + SND_SOC_DAPM_MIC("DMic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Ext Spk", NULL, "HPLOUT"}, + {"Ext Spk", NULL, "HPROUT"}, + + {"DMic Rate 64", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "DMic"}, +}; + +static const char *spk_function[] = {"Off", "On"}; +static const char *input_function[] = {"ADC", "Digital Mic"}; + +static const struct soc_enum rx51_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), +}; + +static const struct snd_kcontrol_new aic34_rx51_controls[] = { + SOC_ENUM_EXT("Speaker Function", rx51_enum[0], + rx51_get_spk, rx51_set_spk), + SOC_ENUM_EXT("Input Select", rx51_enum[1], + rx51_get_input, rx51_set_input), +}; + +static int rx51_aic34_init(struct snd_soc_codec *codec) +{ + int err; + + /* Set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "MIC3L"); + snd_soc_dapm_nc_pin(codec, "MIC3R"); + snd_soc_dapm_nc_pin(codec, "LINE1R"); + + /* Add RX-51 specific controls */ + err = snd_soc_add_controls(codec, aic34_rx51_controls, + ARRAY_SIZE(aic34_rx51_controls)); + if (err < 0) + return err; + + /* Add RX-51 specific widgets */ + snd_soc_dapm_new_controls(codec, aic34_dapm_widgets, + ARRAY_SIZE(aic34_dapm_widgets)); + + /* Set up RX-51 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link rx51_dai[] = { + { + .name = "TLV320AIC34", + .stream_name = "AIC34", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &aic3x_dai, + .init = rx51_aic34_init, + .ops = &rx51_ops, + }, +}; + +/* Audio private data */ +static struct aic3x_setup_data rx51_aic34_setup = { + .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, + .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, +}; + +/* Audio card */ +static struct snd_soc_card rx51_sound_card = { + .name = "RX-51", + .dai_link = rx51_dai, + .num_links = ARRAY_SIZE(rx51_dai), + .platform = &omap_soc_platform, +}; + +/* Audio subsystem */ +static struct snd_soc_device rx51_snd_devdata = { + .card = &rx51_sound_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &rx51_aic34_setup, +}; + +static struct platform_device *rx51_snd_device; + +static int __init rx51_soc_init(void) +{ + int err; + + if (!machine_is_nokia_rx51()) + return -ENODEV; + + BUG_ON(gpio_request(RX51_CODEC_RESET_GPIO, NULL) < 0); + gpio_direction_output(RX51_CODEC_RESET_GPIO, 0); + + gpio_set_value(RX51_CODEC_RESET_GPIO, 0); + udelay(1); + gpio_set_value(RX51_CODEC_RESET_GPIO, 1); + msleep(1); + + rx51_snd_device = platform_device_alloc("soc-audio", -1); + if (!rx51_snd_device) { + err = -ENOMEM; + goto err1; + } + + platform_set_drvdata(rx51_snd_device, &rx51_snd_devdata); + rx51_snd_devdata.dev = &rx51_snd_device->dev; + *(unsigned int *)rx51_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + + err = platform_device_add(rx51_snd_device); + if (err) + goto err2; + + return 0; +err2: + platform_device_put(rx51_snd_device); +err1: + gpio_free(RX51_CODEC_RESET_GPIO); + + return err; +} + +static void __exit rx51_soc_exit(void) +{ + platform_device_unregister(rx51_snd_device); + gpio_free(RX51_CODEC_RESET_GPIO); +} + +module_init(rx51_soc_init); +module_exit(rx51_soc_exit); + +MODULE_AUTHOR("Nokia Corporation"); +MODULE_DESCRIPTION("ALSA SoC Nokia RX-51"); +MODULE_LICENSE("GPL"); -- 1.7.0

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[alsa-devel] timestamps
by Jonas Holmberg 04 May '10

04 May '10

How can I get the CLOCK_MONOTONIC timestamp for the first sample in the data I get from snd_pcm_readi()? /Jonas

2 1

[alsa-devel] [PATCH] ASoC: tpa6130a2: TLV mapping for tpa6140a2
by Peter Ujfalusi 04 May '10

04 May '10

Both tpa6130a2, and tpa6140a2 is supported by the same driver, but the gain dB scaling is different on the amplifiers. Provide different mixer control for the chips with correct TLV mapping. User space will see: "TPA6130A2 Headphone Playback Volume" in case of 6130 "TPA6140A2 Headphone Playback Volume" in case of 6140 The way machine drivers are using this amplifier remained the same. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 31 ++++++++++++++++++++++++++++--- 1 files changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 958d49c..0cf3e38 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -53,6 +53,7 @@ struct tpa6130a2_data { struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; int power_gpio; unsigned char power_state; + enum tpa_model id; }; static int tpa6130a2_i2c_read(int reg) @@ -263,6 +264,20 @@ static const struct snd_kcontrol_new tpa6130a2_controls[] = { tpa6130_tlv), }; +static const unsigned int tpa6140_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 8, TLV_DB_SCALE_ITEM(-5900, 400, 0), + 9, 16, TLV_DB_SCALE_ITEM(-2500, 200, 0), + 17, 31, TLV_DB_SCALE_ITEM(-1000, 100, 0), +}; + +static const struct snd_kcontrol_new tpa6140a2_controls[] = { + SOC_SINGLE_EXT_TLV("TPA6140A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0, + tpa6130a2_get_reg, tpa6130a2_set_reg, + tpa6140_tlv), +}; + /* * Enable or disable channel (left or right) * The bit number for mute and amplifier are the same per channel: @@ -368,13 +383,22 @@ static const struct snd_soc_dapm_route audio_map[] = { int tpa6130a2_add_controls(struct snd_soc_codec *codec) { + struct tpa6130a2_data *data; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets, ARRAY_SIZE(tpa6130a2_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - return snd_soc_add_controls(codec, tpa6130a2_controls, - ARRAY_SIZE(tpa6130a2_controls)); + if (data->id == TPA6140A2) + return snd_soc_add_controls(codec, tpa6140a2_controls, + ARRAY_SIZE(tpa6140a2_controls)); + else + return snd_soc_add_controls(codec, tpa6130a2_controls, + ARRAY_SIZE(tpa6130a2_controls)); } EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); @@ -407,6 +431,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, pdata = client->dev.platform_data; data->power_gpio = pdata->power_gpio; + data->id = pdata->id; mutex_init(&data->mutex); @@ -425,7 +450,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, gpio_direction_output(data->power_gpio, 0); } - switch (pdata->id) { + switch (data->id) { case TPA6130A2: for (i = 0; i < ARRAY_SIZE(data->supplies); i++) data->supplies[i].supply = tpa6130a2_supply_names[i]; -- 1.7.1

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[alsa-devel] Playing multiple sounds at the same time
by Andreas Falkenhahn 04 May '10

04 May '10

Hi, I'm new to ALSA programming so please bear with me if the following question is all too obvious. I did some research on my question before but I didn't come across a clear answer. What I want to do is really simple: Playing multiple sounds in different formats at the same time. E.g. Sound 1: 12345 hz, 8-bit signed, mono Sound 2: 44100 hz, 16-bit unsigned, stereo Sound 3: 8000 hz, 8-bit unsigned, stereo Now I want to play sounds 1 to 3 at the same time. My question is now: Can ALSA do automatic mixing for me here or do I have to do the mixing on my own? In other words, can I simply feed all these three sound streams to ALSA or do I have to mix them into a fourth audio stream first and feeding this new fourth stream to ALSA then? Thanks, Andreas

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