November 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC update for 2.6.33
by Mark Brown 23 Nov '09

23 Nov '09

The following changes since commit dcdec639ad74f20427df82284f98b68f6ac7c0ae: Mark Brown (1): Merge branch 'ads117x' of git://git.kernel.org/.../broonie/sound-2.6 into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Mark Brown (2): ASoC: Fix suspend with active audio streams Merge branch 'for-2.6.32' into for-2.6.33 sound/soc/soc-dapm.c | 20 +++++++++++++++++--- 1 files changed, 17 insertions(+), 3 deletions(-)

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[alsa-devel] [PATCH 6/8] Audio driver of Always Innovating OMAP3-based Touch Book
by Gregoire Gentil 23 Nov '09

23 Nov '09

Signed-off-by: Gregoire Gentil <gregoire(a)gentil.com> --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3touchbook.c | 211 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 221 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3touchbook.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 653a362..c590252 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -83,6 +83,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. +config SND_OMAP_SOC_OMAP3_TOUCHBOOK + tristate "SoC Audio support for OMAP3 Touch Book" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_TOUCHBOOK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Touch Book. + config SND_OMAP_SOC_OMAP3_BEAGLE tristate "SoC Audio support for OMAP3 Beagle" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d6947..f360b82 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -15,6 +15,7 @@ snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-omap3touchbook-objs := omap3touchbook.o snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o @@ -26,4 +27,5 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_TOUCHBOOK) += snd-soc-omap3touchbook.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/omap3touchbook.c b/sound/soc/omap/omap3touchbook.c new file mode 100644 index 0000000..584f07d --- /dev/null +++ b/sound/soc/omap/omap3touchbook.c @@ -0,0 +1,211 @@ +/* + * omap3touchbook.c -- SoC audio for OMAP3 Touch Book + * + * Author: Gregoire Gentil <gregoire(a)gentil.com> + * + * Derived from sound/soc/omap/omap3beagle.c + * + * 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/clk.h> +#include <linux/irq.h> +#include <linux/interrupt.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 <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define TB_HEADPHONE_GPIO 56 +#define TB_HEADPHONE_IRQ OMAP_GPIO_IRQ(TB_HEADPHONE_GPIO) + +static int omap3touchbook_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 fmt; + int ret; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + default: + return -EINVAL; + } + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3touchbook_ops = { + .hw_params = omap3touchbook_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3touchbook_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .ops = &omap3touchbook_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3touchbook = { + .name = "omap3touchbook", + .platform = &omap_soc_platform, + .dai_link = &omap3touchbook_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3touchbook_snd_devdata = { + .card = &snd_soc_omap3touchbook, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3touchbook_snd_device; + +static void jack_work_func(struct work_struct *wq) +{ + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_dapm_sync(codec); +} +DECLARE_WORK(jack_work, jack_work_func); + +static irqreturn_t touchbook_headphone_event(int irq, void *snd) +{ + int status = gpio_get_value(TB_HEADPHONE_GPIO); + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + if (status) { + snd_soc_dapm_disable_pin(codec, "HFL"); + snd_soc_dapm_disable_pin(codec, "HFR"); + } else { + snd_soc_dapm_enable_pin(codec, "HFL"); + snd_soc_dapm_enable_pin(codec, "HFR"); + } + + schedule_work(&jack_work); + return IRQ_HANDLED; +} + +static int __init omap3touchbook_soc_init(void) +{ + int ret; + + if (!machine_is_touchbook()) { + pr_debug("Not OMAP3 Touch Book!\n"); + return -ENODEV; + } + pr_info("OMAP3 Touch Book SoC init\n"); + + omap3touchbook_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3touchbook_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3touchbook_snd_device, + &omap3touchbook_snd_devdata); + omap3touchbook_snd_devdata.dev = &omap3touchbook_snd_device->dev; + *(unsigned int *)omap3touchbook_dai.cpu_dai->private_data = 1; + + /* Touch Book -- headphone jack sensor */ + omap_set_gpio_debounce(TB_HEADPHONE_GPIO, 1); + omap_set_gpio_debounce_time(TB_HEADPHONE_GPIO, 0xff); + + ret = request_irq(TB_HEADPHONE_IRQ, touchbook_headphone_event, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + "touchbook_headphone", + omap3touchbook_snd_device); + if (ret < 0) + goto err1; + + ret = platform_device_add(omap3touchbook_snd_device); + if (ret) + goto err1; + + /* Detect headphone status */ + touchbook_headphone_event(0, omap3touchbook_snd_device); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3touchbook_snd_device); + + return ret; +} + +static void __exit omap3touchbook_soc_exit(void) +{ + platform_device_unregister(omap3touchbook_snd_device); +} + +module_init(omap3touchbook_soc_init); +module_exit(omap3touchbook_soc_exit); + +MODULE_AUTHOR("Gregoire Gentil <gregoire(a)gentil.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Touch Book"); +MODULE_LICENSE("GPL"); -- 1.6.0.4

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 23 Nov '09

23 Nov '09

The following changes since commit 41b51dd47ea7c406a8d49b97804e8acec9dadaed: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Enric Balletbò i Serra (1): ASoC: Add support for IGEP v2 Graeme Gregory (1): ASoC: ADS117x ADC driver Mark Brown (2): ASoC: Update ads117x to current APIs Merge branch 'ads117x' of git://git.kernel.org/.../broonie/sound-2.6 into for-2.6.33 Troy Kisky (4): ASoC: DaVinci: i2s, reduce underruns by combining into 1 element ASoC: DaVinci: pcm, rename variables in prep for ping/pong ASoC: DaVinci: pcm, fix underrun by using sram ASoC: DaVinci: use edma_pause, edma_resume arch/arm/mach-davinci/include/mach/asp.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ads117x.c | 123 +++++++ sound/soc/codecs/ads117x.h | 13 + sound/soc/davinci/davinci-i2s.c | 79 ++++- sound/soc/davinci/davinci-pcm.c | 551 ++++++++++++++++++++++++++---- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/omap/Kconfig | 7 + sound/soc/omap/Makefile | 2 + sound/soc/omap/igep0020.c | 148 ++++++++ 11 files changed, 859 insertions(+), 79 deletions(-) create mode 100644 sound/soc/codecs/ads117x.c create mode 100644 sound/soc/codecs/ads117x.h create mode 100644 sound/soc/omap/igep0020.c

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[alsa-devel] [PATCH] ASoC: Add power down of widget powered up for suspend
by Joonyoung Shim 23 Nov '09

23 Nov '09

If the widget was powered up when the device enters to the suspend, it remains on because the power of the widget doesn't change, but we need to power down the widget on suspend. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/soc-dapm.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d2af872..099d3b6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -978,8 +978,11 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; power = w->power_check(w); - if (power) + if (power) { sys_power = 1; + if (event == SND_SOC_DAPM_STREAM_SUSPEND) + power = 0; + } if (w->power == power) continue; -- 1.6.3.3

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[alsa-devel] [PATCH 6/8] AI TB: Audio driver of Always Innovating OMAP3-based Touch Book
by Gregoire Gentil 23 Nov '09

23 Nov '09

Signed-off-by: Gregoire Gentil <gregoire(a)gentil.com> --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3touchbook.c | 211 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 221 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3touchbook.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 653a362..c590252 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -83,6 +83,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. +config SND_OMAP_SOC_OMAP3_TOUCHBOOK + tristate "SoC Audio support for OMAP3 Touch Book" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_TOUCHBOOK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Touch Book. + config SND_OMAP_SOC_OMAP3_BEAGLE tristate "SoC Audio support for OMAP3 Beagle" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d6947..f360b82 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -15,6 +15,7 @@ snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-omap3touchbook-objs := omap3touchbook.o snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o @@ -26,4 +27,5 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_TOUCHBOOK) += snd-soc-omap3touchbook.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/omap3touchbook.c b/sound/soc/omap/omap3touchbook.c new file mode 100644 index 0000000..584f07d --- /dev/null +++ b/sound/soc/omap/omap3touchbook.c @@ -0,0 +1,211 @@ +/* + * omap3touchbook.c -- SoC audio for OMAP3 Touch Book + * + * Author: Gregoire Gentil <gregoire(a)gentil.com> + * + * Derived from sound/soc/omap/omap3beagle.c + * + * 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/clk.h> +#include <linux/irq.h> +#include <linux/interrupt.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 <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define TB_HEADPHONE_GPIO 56 +#define TB_HEADPHONE_IRQ OMAP_GPIO_IRQ(TB_HEADPHONE_GPIO) + +static int omap3touchbook_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 fmt; + int ret; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + default: + return -EINVAL; + } + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3touchbook_ops = { + .hw_params = omap3touchbook_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3touchbook_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .ops = &omap3touchbook_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3touchbook = { + .name = "omap3touchbook", + .platform = &omap_soc_platform, + .dai_link = &omap3touchbook_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3touchbook_snd_devdata = { + .card = &snd_soc_omap3touchbook, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3touchbook_snd_device; + +static void jack_work_func(struct work_struct *wq) +{ + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_dapm_sync(codec); +} +DECLARE_WORK(jack_work, jack_work_func); + +static irqreturn_t touchbook_headphone_event(int irq, void *snd) +{ + int status = gpio_get_value(TB_HEADPHONE_GPIO); + struct snd_soc_device *socdev = platform_get_drvdata( + omap3touchbook_snd_device); + struct snd_soc_codec *codec = socdev->card->codec; + + if (status) { + snd_soc_dapm_disable_pin(codec, "HFL"); + snd_soc_dapm_disable_pin(codec, "HFR"); + } else { + snd_soc_dapm_enable_pin(codec, "HFL"); + snd_soc_dapm_enable_pin(codec, "HFR"); + } + + schedule_work(&jack_work); + return IRQ_HANDLED; +} + +static int __init omap3touchbook_soc_init(void) +{ + int ret; + + if (!machine_is_touchbook()) { + pr_debug("Not OMAP3 Touch Book!\n"); + return -ENODEV; + } + pr_info("OMAP3 Touch Book SoC init\n"); + + omap3touchbook_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3touchbook_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3touchbook_snd_device, + &omap3touchbook_snd_devdata); + omap3touchbook_snd_devdata.dev = &omap3touchbook_snd_device->dev; + *(unsigned int *)omap3touchbook_dai.cpu_dai->private_data = 1; + + /* Touch Book -- headphone jack sensor */ + omap_set_gpio_debounce(TB_HEADPHONE_GPIO, 1); + omap_set_gpio_debounce_time(TB_HEADPHONE_GPIO, 0xff); + + ret = request_irq(TB_HEADPHONE_IRQ, touchbook_headphone_event, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + "touchbook_headphone", + omap3touchbook_snd_device); + if (ret < 0) + goto err1; + + ret = platform_device_add(omap3touchbook_snd_device); + if (ret) + goto err1; + + /* Detect headphone status */ + touchbook_headphone_event(0, omap3touchbook_snd_device); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3touchbook_snd_device); + + return ret; +} + +static void __exit omap3touchbook_soc_exit(void) +{ + platform_device_unregister(omap3touchbook_snd_device); +} + +module_init(omap3touchbook_soc_init); +module_exit(omap3touchbook_soc_exit); + +MODULE_AUTHOR("Gregoire Gentil <gregoire(a)gentil.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Touch Book"); +MODULE_LICENSE("GPL"); -- 1.6.0.4

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[alsa-devel] Xonar Essence ST (PCI) on 2.6.32-rc8-00011-ga8a8a66: no analog audio, SPDIF fine
by Tony Vroon 23 Nov '09

23 Nov '09

Good evening, After the success of the Xonar D2X in my system, I decided to upgrade to an Essence ST (of the PCI variety, no D6 daughterboard). Unfortunately it seems that it produces no audio for me over the headphone port (the reason I bought this card) or the front-panel headphone port. I currently lack the RCA cabling to hook up the analog "speakers" port, but I promise to try this as soon as practical. The SPDIF output is functional and can be switched on and off using the mixer controls. I'm wondering if the Essence ST driver was written based on an a ST+D6 engineering sample as mine is the ST (no D6) retail box. Is it possible that the hardware is different? lspci output follows, in the hope that it is of use: 01:06.0 Multimedia audio controller [0401]: C-Media Electronics Inc CMI8788 [Oxygen HD Audio] [13f6:8788] Subsystem: ASUSTeK Computer Inc. Virtuoso 100 (Xonar ST) [1043:835d] Flags: bus master, medium devsel, latency 64, IRQ 17 I/O ports at c800 [size=256] Capabilities: [c0] Power Management version 2 Kernel driver in use: AV200 Also, from the ALSA side: # cat /proc/asound/ST/cmi8788 CMI8788 00: 80 e7 1e 86 d0 ed eb cf 00 00 88 bf ff 7f ff 3f 10: d8 ce 98 7e 7c ef c4 c1 00 00 8a bf 37 11 4f 04 20: 60 f7 93 bf af 5a 00 00 d7 4e 00 00 00 00 00 00 30: 30 81 91 92 20 92 ba 74 00 00 00 00 00 00 00 00 40: 10 00 00 00 10 40 00 00 00 00 00 00 00 00 00 00 50: 72 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 60: 19 01 00 09 19 01 00 09 00 00 00 00 00 00 00 00 70: 02 20 00 01 24 08 00 00 00 00 00 00 00 00 00 00 80: 00 40 4d 43 00 00 00 00 00 00 00 00 00 00 00 00 90: 98 11 bf 00 c0 01 00 00 00 00 00 00 00 00 00 00 a0: 00 00 00 00 01 00 93 01 8f 01 00 00 00 00 00 00 b0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 c0: 20 e0 00 00 e4 00 00 00 00 00 00 00 00 00 00 00 d0: 10 00 03 00 00 00 00 00 00 00 00 00 1f 9f 96 00 e0: 01 00 00 00 07 00 0e 00 00 02 05 00 00 b2 02 00 f0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 AC97 00: 0000 0000 0000 0000 0000 8000 0000 8808 10: 8808 8808 8808 9f1f 8808 0000 8000 0000 20: 0000 0000 0000 030c 0bc2 3c30 0000 0000 30: 0000 0000 0000 8080 8080 2000 0000 0000 40: 0000 0000 0000 0000 0000 0000 0000 0000 50: 0000 0000 0000 0000 0000 0000 0004 0000 60: 8080 188f 8043 0004 0000 0000 0000 0000 70: 0300 0011 c000 3935 0000 0000 434d 4969 At any rate, thank you for the oxygen/virtuoso driver. Hopefully this can be solved. The card does seem to be driven fine as the PCM sinks receive their audio without complaint and any multimedia player I use starts output. Regards, Tony V.

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[PATCH] Davinci sound soc driver patch for Neuros OSD2 board support.
by Andrey Porodko 23 Nov '09

23 Nov '09

Signed-off-by: Andrey Porodko <panda(a)chelcom.ru> --- sound/soc/davinci/davinci-evm.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f6..ffdd30e 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -55,6 +55,9 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; + else if (machine_is_neuros_osd2()) + sysclk = 27000000; + else if (machine_is_davinci_da830_evm() || machine_is_davinci_da850_evm()) sysclk = 24576000; @@ -246,6 +249,9 @@ static int __init evm_init(void) if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; + } else if (machine_is_neuros_osd2()) { + evm_snd_dev_data = &evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; -- 1.5.6.5 --------------050305050402010500000508--

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[PATCH] Davinci sound soc driver patch for Neuros OSD2 board support.
by Andrey Porodko 23 Nov '09

23 Nov '09

Signed-off-by: Andrey Porodko <panda(a)chelcom.ru> --- sound/soc/davinci/davinci-evm.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f6..ffdd30e 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -55,6 +55,9 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; + else if (machine_is_neuros_osd2()) + sysclk = 27000000; + else if (machine_is_davinci_da830_evm() || machine_is_davinci_da850_evm()) sysclk = 24576000; @@ -246,6 +249,9 @@ static int __init evm_init(void) if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; + } else if (machine_is_neuros_osd2()) { + evm_snd_dev_data = &evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; -- 1.5.6.5 --------------000804080502060002040509--

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[alsa-devel] [PATCH 1/2] opti-miro: make miro.h header available outside the alsa directory
by Krzysztof Helt 23 Nov '09

23 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Move the miro.h header to the include/sound directory. It can be used in the Miro PCM20 radio driver (v4l). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is a part of my effort to restore the PCM20's radio driver. include/sound/aci.h | 73 ++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/miro.h | 73 ---------------------------------------------- 3 files changed, 74 insertions(+), 74 deletions(-) create mode 100644 include/sound/aci.h delete mode 100644 sound/isa/opti9xx/miro.h diff --git a/include/sound/aci.h b/include/sound/aci.h new file mode 100644 index 0000000..bb796d0 --- /dev/null +++ b/include/sound/aci.h @@ -0,0 +1,73 @@ +#ifndef _ACI_H_ +#define _ACI_H_ + +#define ACI_REG_COMMAND 0 /* write register offset */ +#define ACI_REG_STATUS 1 /* read register offset */ +#define ACI_REG_BUSY 2 /* busy register offset */ +#define ACI_REG_RDS 2 /* PCM20: RDS register offset */ +#define ACI_MINTIME 500 /* ACI time out limit */ + +#define ACI_SET_MUTE 0x0d +#define ACI_SET_POWERAMP 0x0f +#define ACI_SET_TUNERMUTE 0xa3 +#define ACI_SET_TUNERMONO 0xa4 +#define ACI_SET_IDE 0xd0 +#define ACI_SET_WSS 0xd1 +#define ACI_SET_SOLOMODE 0xd2 +#define ACI_SET_PREAMP 0x03 +#define ACI_GET_PREAMP 0x21 +#define ACI_WRITE_TUNE 0xa7 +#define ACI_READ_TUNERSTEREO 0xa8 +#define ACI_READ_TUNERSTATION 0xa9 +#define ACI_READ_VERSION 0xf1 +#define ACI_READ_IDCODE 0xf2 +#define ACI_INIT 0xff +#define ACI_STATUS 0xf0 +#define ACI_S_GENERAL 0x00 +#define ACI_ERROR_OP 0xdf + +/* ACI Mixer */ + +/* These are the values for the right channel GET registers. + Add an offset of 0x01 for the left channel register. + (left=right+0x01) */ + +#define ACI_GET_MASTER 0x03 +#define ACI_GET_MIC 0x05 +#define ACI_GET_LINE 0x07 +#define ACI_GET_CD 0x09 +#define ACI_GET_SYNTH 0x0b +#define ACI_GET_PCM 0x0d +#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */ +#define ACI_GET_LINE2 0x12 + +#define ACI_GET_EQ1 0x22 /* from Bass ... */ +#define ACI_GET_EQ2 0x24 +#define ACI_GET_EQ3 0x26 +#define ACI_GET_EQ4 0x28 +#define ACI_GET_EQ5 0x2a +#define ACI_GET_EQ6 0x2c +#define ACI_GET_EQ7 0x2e /* ... to Treble */ + +/* And these are the values for the right channel SET registers. + For left channel access you have to add an offset of 0x08. + MASTER is an exception, which needs an offset of 0x01 */ + +#define ACI_SET_MASTER 0x00 +#define ACI_SET_MIC 0x30 +#define ACI_SET_LINE 0x31 +#define ACI_SET_CD 0x34 +#define ACI_SET_SYNTH 0x33 +#define ACI_SET_PCM 0x32 +#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */ +#define ACI_SET_LINE2 0x36 + +#define ACI_SET_EQ1 0x40 /* from Bass ... */ +#define ACI_SET_EQ2 0x41 +#define ACI_SET_EQ3 0x42 +#define ACI_SET_EQ4 0x43 +#define ACI_SET_EQ5 0x44 +#define ACI_SET_EQ6 0x45 +#define ACI_SET_EQ7 0x46 /* ... to Treble */ + +#endif /* _ACI_H_ */ diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index db4a4fb..932a067 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -40,7 +40,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -#include "miro.h" +#include <sound/aci.h> MODULE_AUTHOR("Martin Langer <martin-langer(a)gmx.de>"); MODULE_LICENSE("GPL"); diff --git a/sound/isa/opti9xx/miro.h b/sound/isa/opti9xx/miro.h deleted file mode 100644 index 6e1385b..0000000 --- a/sound/isa/opti9xx/miro.h +++ /dev/null @@ -1,73 +0,0 @@ -#ifndef _MIRO_H_ -#define _MIRO_H_ - -#define ACI_REG_COMMAND 0 /* write register offset */ -#define ACI_REG_STATUS 1 /* read register offset */ -#define ACI_REG_BUSY 2 /* busy register offset */ -#define ACI_REG_RDS 2 /* PCM20: RDS register offset */ -#define ACI_MINTIME 500 /* ACI time out limit */ - -#define ACI_SET_MUTE 0x0d -#define ACI_SET_POWERAMP 0x0f -#define ACI_SET_TUNERMUTE 0xa3 -#define ACI_SET_TUNERMONO 0xa4 -#define ACI_SET_IDE 0xd0 -#define ACI_SET_WSS 0xd1 -#define ACI_SET_SOLOMODE 0xd2 -#define ACI_SET_PREAMP 0x03 -#define ACI_GET_PREAMP 0x21 -#define ACI_WRITE_TUNE 0xa7 -#define ACI_READ_TUNERSTEREO 0xa8 -#define ACI_READ_TUNERSTATION 0xa9 -#define ACI_READ_VERSION 0xf1 -#define ACI_READ_IDCODE 0xf2 -#define ACI_INIT 0xff -#define ACI_STATUS 0xf0 -#define ACI_S_GENERAL 0x00 -#define ACI_ERROR_OP 0xdf - -/* ACI Mixer */ - -/* These are the values for the right channel GET registers. - Add an offset of 0x01 for the left channel register. - (left=right+0x01) */ - -#define ACI_GET_MASTER 0x03 -#define ACI_GET_MIC 0x05 -#define ACI_GET_LINE 0x07 -#define ACI_GET_CD 0x09 -#define ACI_GET_SYNTH 0x0b -#define ACI_GET_PCM 0x0d -#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */ -#define ACI_GET_LINE2 0x12 - -#define ACI_GET_EQ1 0x22 /* from Bass ... */ -#define ACI_GET_EQ2 0x24 -#define ACI_GET_EQ3 0x26 -#define ACI_GET_EQ4 0x28 -#define ACI_GET_EQ5 0x2a -#define ACI_GET_EQ6 0x2c -#define ACI_GET_EQ7 0x2e /* ... to Treble */ - -/* And these are the values for the right channel SET registers. - For left channel access you have to add an offset of 0x08. - MASTER is an exception, which needs an offset of 0x01 */ - -#define ACI_SET_MASTER 0x00 -#define ACI_SET_MIC 0x30 -#define ACI_SET_LINE 0x31 -#define ACI_SET_CD 0x34 -#define ACI_SET_SYNTH 0x33 -#define ACI_SET_PCM 0x32 -#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */ -#define ACI_SET_LINE2 0x36 - -#define ACI_SET_EQ1 0x40 /* from Bass ... */ -#define ACI_SET_EQ2 0x41 -#define ACI_SET_EQ3 0x42 -#define ACI_SET_EQ4 0x43 -#define ACI_SET_EQ5 0x44 -#define ACI_SET_EQ6 0x45 -#define ACI_SET_EQ7 0x46 /* ... to Treble */ - -#endif /* _MIRO_H_ */ -- 1.6.4 ---------------------------------------------------------------------- Internetowi nie placa! 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[alsa-devel] [PATCH] AACI cleanup
by Russell King - ARM Linux 23 Nov '09

23 Nov '09

Below is a patch which adds me to MAINTAINERS for this driver, and fixes the buffer size calculation to use the size which ALSA is expecting. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> -- Please review carefully, since I am presently unable to run-time test changes to this driver due to incompatibilities system ABI. MAINTAINERS | 5 +++++ sound/arm/aaci.c | 7 ++----- 2 files changed, 7 insertions(+), 5 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index c824b4d..dffb0c5 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -512,6 +512,11 @@ W: http://www.arm.linux.org.uk/ S: Maintained F: arch/arm/ +ARM PRIMECELL AACI PL041 DRIVER +M: Russell King <linux(a)arm.linux.org.uk> +S: Maintained +F: sound/arm/aaci.* + ARM PRIMECELL MMCI PL180/1 DRIVER S: Orphan F: drivers/mmc/host/mmci.* diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1f0f821..a03fe80 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -18,10 +18,7 @@ #include <linux/interrupt.h> #include <linux/err.h> #include <linux/amba/bus.h> - -#include <asm/io.h> -#include <asm/irq.h> -#include <asm/sizes.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/initval.h> @@ -534,7 +531,7 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream) struct aaci_runtime *aacirun = runtime->private_data; aacirun->start = (void *)runtime->dma_area; - aacirun->end = aacirun->start + runtime->dma_bytes; + aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream); aacirun->ptr = aacirun->start; aacirun->period = aacirun->bytes = frames_to_bytes(runtime, runtime->period_size);

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