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

Re: [alsa-devel] iec958 regression with alsa 1.0.22 and kernel 2.6.33 on Intel DG45FC motherboard
by Wu Fengguang 06 May '10

06 May '10

Hi David, On Thu, May 06, 2010 at 07:05:16AM +0800, David Härdeman wrote: > A recent upgrade to kernel 2.6.33.x and alsa 1.0.22 broke spdif (iec958) Upgraded from which kernel? > output on my Intel DG45FC motherboard (IDT 92HD73E1X5 codec, > snd-hda-intel driver). > > After playing around a bit with module params, it turns out that loading > snd-hda-intel with model=no-jd fixes this problem (previously the module > worked without any model option at all, note also that model=intel does > not work which is kinda weird). You mean model=intel never worked? > Any suggestions on a proper fix? The current situation is not really > user-friendly... Thanks, Fengguang

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[alsa-devel] SPDIF out with HDA/Conexant 20561?
by pl bossart 06 May '10

06 May '10

Has anyone had any luck enabling S/PDIF output with the Conexant 20561/ctx5051? On my Lenovo W500 I see a digital output device, I can play on it but I can't hear any sound on my receiver. There was a patch by Takashi earlier this year to correct a wrong nodeID (03b515c45e2a93331d2ac9dc1c16440f7fa313ec, ALSA: hda - Fix SPDIF output widget for Cxt5051 codec), so I figure some amount of testing has been done already? I wonder if this comes from a non-initialized pin configuration as reported in /proc/asound/card0/codec#0? Node 0x1c [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x12 [Audio Output] wcaps 0x211: 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="Conexant Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Thanks - Pierre

2 4

Re: [alsa-devel] ALSA 1.0.23 Kernel Panic in snd_hda_codec_idt:stac92xx_add_jack
by Robert Krakora 06 May '10

06 May '10

All: I am seeing a kernel panic with ALSA Drivers 1.0.23 in snd_hda_codec_idt:stac92xx_add_jack on my Dell Studio 1737 running stock CentOS 5.4. Before I file a bug report I just wanted to make sure that you did not already have a patch for this problem. Please see the attached .jpg for the trace. I took a picture as this is a hard failure with a backtrace (keyboard blinks). Best Regards, -- Rob Krakora Senior Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

2 2

[alsa-devel] [GIT PULL} ASoC updates for 2.6.35
by Liam Girdwood 06 May '10

06 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/lrg/asoc-2.6.git for-2.6.35 Jarkko Nikula (2): ASoC: omap: Add basic audio support for Nokia RX-51/N900 ASoC: tlv320aic3x: Add platform data and reset gpio handling Liam Girdwood (3): ASoC: pandora - update DAPM pins ASoC: zoom2 - update DAPM pins ASoC: tlv320dac33 - disable regulators at i2c remove() Peter Ujfalusi (9): ASoC: TWL4030: AIF/APLL fix in DAPM domain ASoC: TWL4030: Remove OUTL/R outputs ASoC: tlv320dac33: Optimize power up, and restore ASoC: tlv320dac33: Revised module loading, and DAC33 ID read ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structure ASoC: tlv320dac33: Support for turning off the codec ASoC: tpa6130a2: TLV mapping for tpa6140a2 ASoC: tpa6130a2: Support for limiting gain ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC power include/sound/tlv320aic3x.h | 17 +++ include/sound/tpa6130a2-plat.h | 1 + sound/soc/codecs/tlv320aic3x.c | 25 ++++ sound/soc/codecs/tlv320dac33.c | 223 ++++++++++++++++-------------- sound/soc/codecs/tpa6130a2.c | 99 +++++++++++++- sound/soc/codecs/twl4030.c | 86 ++++++++---- sound/soc/omap/Kconfig | 10 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 2 - sound/soc/omap/rx51.c | 294 ++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/zoom2.c | 3 - 11 files changed, 619 insertions(+), 143 deletions(-) create mode 100644 include/sound/tlv320aic3x.h create mode 100644 sound/soc/omap/rx51.c

2 1

[alsa-devel] [PATCHv2] ASoC: tlv320aic3x: Add platform data and reset gpio handling
by Jarkko Nikula 06 May '10

06 May '10

Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- v2: - Removed added regulator_bulk_disable brain fart from aic3x_i2c_probe - Added comment to gpio_reset number in aic3x_pdata --- include/sound/tlv320aic3x.h | 17 +++++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++++++++ 2 files changed, 42 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..b1a5f34 --- /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; /* < 0 if not used */ +}; + +#endif \ No newline at end of file diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b2c2794..435adf1 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 +1353,19 @@ 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: + 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] ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC power
by Peter Ujfalusi 06 May '10

06 May '10

Do not change the codec defaults for the following registers: 0x40, 0x41: Line output gains, do not use amplification 0x42: LOM/LOP Voltage hold, and selection 0x44: LOM inversion control It has been found, that the values configured to these registers can cause amplification, which can make the output of DAC33 distorted. The codec reset values are considered safe in all environmnts. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- If there is a need for user control over these, than additional kcontrols can be invented for the settings. But sorting out the meaning, and naming of such controls could take considerable time. At the moment, it is safer to use the codec defaults for these instead of the original values. sound/soc/codecs/tlv320dac33.c | 11 ----------- 1 files changed, 0 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 68b7ccb..ad5e263 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -303,21 +303,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec) /* C : (defaults) */ dac33_write(codec, DAC33_DAC_CTRL_C, 0x00); - /* 64-65 : L&R DAC power control - Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/ - dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); - dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); - /* 73 : volume soft stepping control, clock source = internal osc (?) */ dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN); - /* 66 : LOP/LOM Modes */ - dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff); - - /* 68 : LOM inverted from LOP */ - dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2)); - dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB); /* Restore only selected registers (gains mostly) */ -- 1.7.1

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[alsa-devel] [PATCHv2] ASoC: omap: Add basic audio support for Nokia RX-51/N900
by Jarkko Nikula 06 May '10

06 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 | 294 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 306 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..47d831e --- /dev/null +++ b/sound/soc/omap/rx51.c @@ -0,0 +1,294 @@ +/* + * 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" + +/* + * 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; + + 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: + + return err; +} + +static void __exit rx51_soc_exit(void) +{ + platform_device_unregister(rx51_snd_device); +} + +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

5 4

[alsa-devel] [PATCH] es1688: allocate snd_es1688 structure as a part of snd_card structure
by Krzysztof Helt 06 May '10

06 May '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Allocate the snd_es1688 during the snd_card allocation. This allows to remove the card pointer from the snd_es1688 structure. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the first part of changes to merge the es968 and the es1688 drivers. include/sound/es1688.h | 10 ++++---- sound/isa/es1688/es1688.c | 23 ++++++++++++--------- sound/isa/es1688/es1688_lib.c | 43 +++++++++++++--------------------------- sound/isa/gus/gusextreme.c | 26 +++++++++++++----------- 4 files changed, 46 insertions(+), 56 deletions(-) diff --git a/include/sound/es1688.h b/include/sound/es1688.h index 10fcf14..4c29572 100644 --- a/include/sound/es1688.h +++ b/include/sound/es1688.h @@ -44,7 +44,6 @@ struct snd_es1688 { unsigned char pad; unsigned int dma_size; - struct snd_card *card; struct snd_pcm *pcm; struct snd_pcm_substream *playback_substream; struct snd_pcm_substream *capture_substream; @@ -108,14 +107,15 @@ struct snd_es1688 { void snd_es1688_mixer_write(struct snd_es1688 *chip, unsigned char reg, unsigned char data); int snd_es1688_create(struct snd_card *card, + struct snd_es1688 *chip, unsigned long port, unsigned long mpu_port, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 ** rchip); -int snd_es1688_pcm(struct snd_es1688 *chip, int device, struct snd_pcm ** rpcm); -int snd_es1688_mixer(struct snd_es1688 *chip); + unsigned short hardware); +int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, int device, + struct snd_pcm **rpcm); +int snd_es1688_mixer(struct snd_card *card, struct snd_es1688 *chip); #endif /* __SOUND_ES1688_H */ diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 07df201..2816794 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -79,8 +79,8 @@ static int __devinit snd_es1688_match(struct device *dev, unsigned int n) return enable[n]; } -static int __devinit snd_es1688_legacy_create(struct snd_card *card, - struct device *dev, unsigned int n, struct snd_es1688 **rchip) +static int __devinit snd_es1688_legacy_create(struct snd_card *card, + struct snd_es1688 *chip, struct device *dev, unsigned int n) { static long possible_ports[] = {0x220, 0x240, 0x260}; static int possible_irqs[] = {5, 9, 10, 7, -1}; @@ -104,14 +104,14 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card, } if (port[n] != SNDRV_AUTO_PORT) - return snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + return snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_AUTO); i = 0; do { port[n] = possible_ports[i]; - error = snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + error = snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_AUTO); } while (error < 0 && ++i < ARRAY_SIZE(possible_ports)); return error; @@ -125,19 +125,22 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) struct snd_pcm *pcm; int error; - error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + error = snd_card_create(index[n], id[n], THIS_MODULE, + sizeof(struct snd_es1688), &card); if (error < 0) return error; - error = snd_es1688_legacy_create(card, dev, n, &chip); + chip = card->private_data; + + error = snd_es1688_legacy_create(card, chip, dev, n); if (error < 0) goto out; - error = snd_es1688_pcm(chip, 0, &pcm); + error = snd_es1688_pcm(card, chip, 0, &pcm); if (error < 0) goto out; - error = snd_es1688_mixer(chip); + error = snd_es1688_mixer(card, chip); if (error < 0) goto out; diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index c76bb00..80c11fd 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -638,23 +638,20 @@ static const char *snd_es1688_chip_id(struct snd_es1688 *chip) } int snd_es1688_create(struct snd_card *card, + struct snd_es1688 *chip, unsigned long port, unsigned long mpu_port, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 **rchip) + unsigned short hardware) { static struct snd_device_ops ops = { .dev_free = snd_es1688_dev_free, }; - struct snd_es1688 *chip; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (chip == NULL) return -ENOMEM; chip->irq = -1; @@ -662,25 +659,21 @@ int snd_es1688_create(struct snd_card *card, if ((chip->res_port = request_region(port + 4, 12, "ES1688")) == NULL) { snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4); - snd_es1688_free(chip); return -EBUSY; } if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) { snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq); - snd_es1688_free(chip); return -EBUSY; } chip->irq = irq; if (request_dma(dma8, "ES1688")) { snd_printk(KERN_ERR "es1688: can't grab DMA8 %d\n", dma8); - snd_es1688_free(chip); return -EBUSY; } chip->dma8 = dma8; spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); - chip->card = card; chip->port = port; mpu_port &= ~0x000f; if (mpu_port < 0x300 || mpu_port > 0x330) @@ -689,23 +682,16 @@ int snd_es1688_create(struct snd_card *card, chip->mpu_irq = mpu_irq; chip->hardware = hardware; - if ((err = snd_es1688_probe(chip)) < 0) { - snd_es1688_free(chip); + err = snd_es1688_probe(chip); + if (err < 0) return err; - } - if ((err = snd_es1688_init(chip, 1)) < 0) { - snd_es1688_free(chip); - return err; - } - /* Register device */ - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_es1688_free(chip); + err = snd_es1688_init(chip, 1); + if (err < 0) return err; - } - *rchip = chip; - return 0; + /* Register device */ + return snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); } static struct snd_pcm_ops snd_es1688_playback_ops = { @@ -730,12 +716,14 @@ static struct snd_pcm_ops snd_es1688_capture_ops = { .pointer = snd_es1688_capture_pointer, }; -int snd_es1688_pcm(struct snd_es1688 * chip, int device, struct snd_pcm ** rpcm) +int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, + int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "ESx688", device, 1, 1, &pcm)) < 0) + err = snd_pcm_new(card, "ESx688", device, 1, 1, &pcm); + if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es1688_playback_ops); @@ -1009,18 +997,15 @@ static unsigned char snd_es1688_init_table[][2] = { { ES1688_REC_DEV, 0x17 } }; -int snd_es1688_mixer(struct snd_es1688 *chip) +int snd_es1688_mixer(struct snd_card *card, struct snd_es1688 *chip) { - struct snd_card *card; unsigned int idx; int err; unsigned char reg, val; - if (snd_BUG_ON(!chip || !chip->card)) + if (snd_BUG_ON(!chip || !card)) return -EINVAL; - card = chip->card; - strcpy(card->mixername, snd_es1688_chip_id(chip)); for (idx = 0; idx < ARRAY_SIZE(snd_es1688_controls); idx++) { diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 65e4b18..008e8e5 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -95,7 +95,7 @@ static int __devinit snd_gusextreme_match(struct device *dev, unsigned int n) } static int __devinit snd_gusextreme_es1688_create(struct snd_card *card, - struct device *dev, unsigned int n, struct snd_es1688 **rchip) + struct snd_es1688 *chip, struct device *dev, unsigned int n) { static long possible_ports[] = {0x220, 0x240, 0x260}; static int possible_irqs[] = {5, 9, 10, 7, -1}; @@ -119,14 +119,14 @@ static int __devinit snd_gusextreme_es1688_create(struct snd_card *card, } if (port[n] != SNDRV_AUTO_PORT) - return snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_1688, rchip); + return snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_1688); i = 0; do { port[n] = possible_ports[i]; - error = snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_1688, rchip); + error = snd_es1688_create(card, chip, port[n], mpu_port[n], + irq[n], mpu_irq[n], dma8[n], ES1688_HW_1688); } while (error < 0 && ++i < ARRAY_SIZE(possible_ports)); return error; @@ -206,9 +206,8 @@ static int __devinit snd_gusextreme_detect(struct snd_gus_card *gus, return 0; } -static int __devinit snd_gusextreme_mixer(struct snd_es1688 *chip) +static int __devinit snd_gusextreme_mixer(struct snd_card *card) { - struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; int error; @@ -241,17 +240,20 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) struct snd_opl3 *opl3; int error; - error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + error = snd_card_create(index[n], id[n], THIS_MODULE, + sizeof(struct snd_es1688), &card); if (error < 0) return error; + es1688 = card->private_data; + if (mpu_port[n] == SNDRV_AUTO_PORT) mpu_port[n] = 0; if (mpu_irq[n] > 15) mpu_irq[n] = -1; - error = snd_gusextreme_es1688_create(card, dev, n, &es1688); + error = snd_gusextreme_es1688_create(card, es1688, dev, n); if (error < 0) goto out; @@ -280,11 +282,11 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) } gus->codec_flag = 1; - error = snd_es1688_pcm(es1688, 0, NULL); + error = snd_es1688_pcm(card, es1688, 0, NULL); if (error < 0) goto out; - error = snd_es1688_mixer(es1688); + error = snd_es1688_mixer(card, es1688); if (error < 0) goto out; @@ -300,7 +302,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) if (error < 0) goto out; - error = snd_gusextreme_mixer(es1688); + error = snd_gusextreme_mixer(card); if (error < 0) goto out; -- 1.6.4 -------------------------------------------------- Odkryj tajemnice TAROTA! 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[alsa-devel] [PATCH 1/2] powerpc/8610: add probing for individual DMA channels, not just DMA controllers
by Timur Tabi 05 May '10

05 May '10

A future version of the MPC8610 HPCD's ASoC DMA driver will probe on individual DMA channel nodes, so the DMA controller nodes' compatible string must be listed in mpc8610_ids[] for the probe to work. Also remove the "gianfar" compatible from mpc8610_ids[], since there is no gianfar (or any other networking device) on the 8610. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Kumar, the ASoC mainters are willing to pick up this patch, but they want an ACK from you first. Or, you could pick it up, since by itself it's harmless. arch/powerpc/platforms/86xx/mpc8610_hpcd.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c index 5abe137..018cc67 100644 --- a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c +++ b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c @@ -83,7 +83,8 @@ static struct of_device_id __initdata mpc8610_ids[] = { { .compatible = "fsl,mpc8610-immr", }, { .compatible = "fsl,mpc8610-guts", }, { .compatible = "simple-bus", }, - { .compatible = "gianfar", }, + /* So that the DMA channel nodes can be probed individually: */ + { .compatible = "fsl,eloplus-dma", }, {} }; -- 1.6.5

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[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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