September 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] AIC23 codec ADC problem
by Troy Kisky 30 Sep '08

30 Sep '08

Hi All I'm using the TLV320AIC23 to record sound from a mic. It sounds fine when I play it back, but when I look at the data, the value in one channel is always exactly double the value in the other. Since the mic gives the same input to both ADC's I was expecting to have the same value in the left & right channels. Has anyone seen this before? Any guesses to what I might be doing wrong? Can someone using an AIC23 check their data for me? I'm using a davinci DM644x platform and spi interface to the codec in dsp_a mode. sample data 00ec40 fffeffff 00040002 00040002 00000000 00ec50 fffeffff fffeffff 00000000 fffafffd 00ec60 fffeffff 00020001 00040002 00080004 00ec70 000a0005 0016000b 0014000a 001a000d 00ec80 001c000e 001e000f 00200010 00260013 00ec90 001e000f 001e000f 001e000f 001c000e 00eca0 001c000e 0018000c 0014000a 0014000a 00ecb0 0014000a 000e0007 00120009 0014000a ^ ^ ^ ^ Always even After swapping the left & right channels at the codec 011af0 fffcfffe fffeffff 00060003 00060003 011b00 000a0005 000c0006 000e0007 000c0006 011b10 00080004 00080004 00060003 00060003 011b20 00080004 00060003 00060003 00040002 011b30 00020001 00020001 00000000 00020001 011b40 fffeffff 00000000 00040002 00060003 011b50 000c0006 000e0007 00100008 0014000a 011b60 00120009 000e0007 00100008 00100008 011b70 000e0007 00080004 00080004 00060003 011b80 00060003 00060003 00060003 000a0005 No change, so the ADC's both do convert to the same value. But there appears to be a 1 bit delay between the two channels. My McBsp is reading 1 32 bit element, so I don't think it can shift only half of the element. After switching to dsp_b mode (an extra bit delay before data starts) 00b130 fff8fff8 fff5fff5 fffafffa fffcfffc 00b140 fffefffe 00010001 00000000 00010001 00b150 00020002 00010001 00010001 00010001 00b160 00000000 ffffffff 00020002 00000000 00b170 00000000 00000000 fffefffe fffefffe 00b180 fffdfffd fffdfffd 00000000 ffffffff 00b190 00020002 00010001 00020002 00030003 00b1a0 00030003 00030003 00050005 00030003 00b1b0 00050005 00030003 00010001 00000000 When I play this back in dsp_b mode I hear random soft popping from the speaker, but it is loud. Playing it in dsp_a mode is quiet with no popping. So, I conclude that recording with the codec is buggy in dsp_a mode and playback is buggy in dsp_b mode. I hope someone will tell me I'm wrong. Thanks Troy

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[alsa-devel] maintaining fixed latency between output and input?
by Aaron J. Grier 30 Sep '08

30 Sep '08

what's the most straight-forward way to maintain fixed latency between output and input without sitting in loop and continuously checking snd_pcm_avail_update() for the output and input descriptors? I was originally going to use callbacks, but I've recently read that is not a good idea. poll() seems to be the reccomended method. I could use separate threads for output and input, but then I have to perform synchronization between them, marshal external requests between the two threads, and make a lot more work for myself. I'm visualizing something like this: for(;;) { poll() if (output_is_ready) { /* write output buffer */ /* do output-side housekeeping */ } if (input_is_ready) { /* read input buffer */ /* do input-side housekeeping */ } /* communicate with other threads */ } combining the output and input snd_pcm_t into a single descriptor list for poll() seems straight-forward, but can I pass this superset list to snd_pcm_poll_descriptors_revents() without it getting confused? -- Aaron J. Grier | "Not your ordinary poofy goof." | agrier(a)poofygoof.com

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[alsa-devel] [PATCH 0/5] [RFC] ALSA ASoC Support for TLVaic23b audio codec
by Arun KS 29 Sep '08

29 Sep '08

Hi all, The following patches adds ASOC driver for TLVaic23b audio codec. Tested playback and capture on omap 5912osk board with some hacks in the omap platform and machine driver. Thanks, Arun -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] Secure delivery
by ink＠jurassic.park.msu.ru 29 Sep '08

29 Sep '08

Bad Gateway: The message has been attached. ++++ Attachment: No Virus found ++++ F-Secure AntiVirus - www.f-secure.com

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[alsa-devel] [PATCH 0/6] ASoC Blackfin and API updates
by Mark Brown 29 Sep '08

29 Sep '08

The following changes since commit ad7f41b60efde140df0f929e8d408ebe2e59cb0e: Jean Delvare (1): ALSA: Drop Vladimir Barinov's e-mail address adding the Blackfin updates Brian posted over the weekend plus some more usage of snd_soc_dapm_nc_pin() are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Cliff Cai (4): ASoC codec: AD73311 audio codec driver ASoC Blackfin: add I2S DAI support for AD73311 ASoC Blackfin: add asoc ad73311 driver supporting in Blackfin boards ASoC Blackfin: fix bug - Audio Latency on AD1981 with MMAP enabled Mark Brown (2): ASoC: Use snd_soc_dapm_nc_pin() in Zaurus machine drivers ASoC: Use snd_soc_dapm_nc_pin() in N810 machine driver sound/soc/blackfin/Kconfig | 16 +++ sound/soc/blackfin/Makefile | 3 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 42 +++++- sound/soc/blackfin/bf5xx-ac97.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 240 +++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-i2s.c | 47 +++++-- sound/soc/blackfin/bf5xx-sport.h | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad73311.c | 107 ++++++++++++++++ sound/soc/codecs/ad73311.h | 90 +++++++++++++ sound/soc/omap/n810.c | 6 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/spitz.c | 14 +- sound/soc/pxa/tosa.c | 4 +- 16 files changed, 547 insertions(+), 39 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad73311.c create mode 100644 sound/soc/codecs/ad73311.c create mode 100644 sound/soc/codecs/ad73311.h

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[alsa-devel] [PATCH 4/5] [RFC] ALSA ASOC adds destination and source port for dma
by Arun KS 29 Sep '08

29 Sep '08

Adds destination and source ports for dma which is required for OMAP1 Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/omap-pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 690bfea..9e670ef 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,6 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); + if (!cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any @@ -147,12 +148,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; + dma_params.dst_port = OMAP_DMA_PORT_MPUI; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; + dma_params.src_port = OMAP_DMA_PORT_MPUI; } /* * Set DMA transfer frame size equal to ALSA period size and frame -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] ALSA: cs46xx: Add PCI IDs for TerraTec and Hercules cards
by Vedran Miletić 29 Sep '08

29 Sep '08

This patch adds PCI IDs for: * TerraTec DMX XFire 1024 * Hercules Gamesurround Fortissimo II * Hercules Gamesurround Fortissimo III 7.1 All those cards were supported as generic CS46xx device, so they will work as before. I'm pretty sure that first two cards work, as they have same hardware design as reference card. Not sure about Fortissimo III, but this won't break it if it worked. Tested on TerraTec DMX XFire 1024. -- Vedran Miletić

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Re: [alsa-devel] [PATCH 1/5] [RFC] ALSA ASoC driver for TLVaic23b audio codec
by Arun KS 29 Sep '08

29 Sep '08

Mark Brown wrote: > On Mon, Sep 29, 2008 at 03:15:04PM +0530, Arun KS wrote: >> ASOC Audio driver for TLVaic23b audio codec > > This looks good overall, thanks! There's quite a few comments below but > they're mostly minor things at the coding style issues. > > The only issue I've got overall is that the name of the codec seems to > vary through the driver - sometimes it's referred to as the tlv320aic23, > sometimes as aic23 and sometimes as tlvaic23. It'd be good if the > naming scheme were more consistent. My inclination would be to go with > tlv320aic23 since it will reduce the risk of collisions and the general > style is to use the full part number (then again, I know the tlv320aic3x > driver didn't do that). > >> --- a/sound/soc/codecs/Kconfig >> +++ b/sound/soc/codecs/Kconfig >> @@ -55,3 +55,7 @@ config SND_SOC_TWL4030 >> tristate >> depends on SND_SOC && TWL4030_CORE >> > > This patch is against the OMAP tree - please generate all submitted > patches against either the ASoC dev tree or the topic/asoc branch of: > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > Even if other patches in the series depend on the OMAP tree there should > be no reason not to merge an I2C codec driver like this with no external > dependencies immediately. > >> +config SND_SOC_TLV320AIC23 >> + tristate >> + depends on I2C >> + > > Please also add this to the SND_SOC_ALL_CODECS Kconfig option. > >> +static int aic23_write(struct snd_soc_codec *codec, unsigned int reg, >> + unsigned int value) >> +{ >> + >> + u8 data; >> + >> + /* TLV320AIC23 has 7 bit address and 9 bits of data >> + * so we need to switch one data bit into reg and rest >> + * of data into val >> + */ >> + >> + if ((reg < 0 || reg > 9) && (reg != 15)) { >> + printk(KERN_WARNING "Invalid register R%d\n", reg); >> + return -1; >> + } > > It'd be nice if this said which device was being written to - adding in > the driver name to the printk(), for example. It'd make it easier for > someone seeing the message in their logs to work out where it came from. > >> + data = (reg << 1) | (value >> 8 & 0x01); >> + >> + aic23_write_reg_cache(codec, reg, value); >> + >> + if (codec->hw_write(codec->control_data, (u8) data, >> + (u8) (value & 0xff)) == 0) >> + return 0; >> + >> + printk(KERN_ERR "I2C: cannot write %03x to register R%d\n", value, reg); > > Again, saying where the error came from would be nice. > >> + SOC_SINGLE("Mic Mute", ANALOG_AUDIO_CONTROL_ADDR, 1, 0x01, 0), > > This should be "Mic Switch" to help UIs display the control correctly. > >> +/* aic23 related */ >> +static const struct aic23_samplerate_reg_info >> + rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = { >> + {4000, 0x06, 1}, /* 4000 */ > > You could just use [] for the size of the array and let the compiler > figure it out - you actually use ARRAY_SIZE() rather than the #define > for the array size later anyway. > >> +static int aic23_set_bias_level(struct snd_soc_codec *codec, >> + enum snd_soc_bias_level level) >> +{ >> + u16 reg; >> + >> + switch (level) { >> + case SND_SOC_BIAS_ON: >> + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); >> + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, >> + reg & (~DEVICE_POWER_OFF)); >> + /* Activate interface */ >> + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); >> + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg | ACT_ON); >> + break; >> + case SND_SOC_BIAS_PREPARE: >> + break; >> + case SND_SOC_BIAS_STANDBY: >> + reg = aic23_read_reg_cache(codec, POWER_DOWN_CONTROL_ADDR); >> + aic23_write(codec, POWER_DOWN_CONTROL_ADDR, >> + reg | DEVICE_POWER_OFF); >> + /* Deactivate interface */ >> + reg = aic23_read_reg_cache(codec, DIGITAL_INTERFACE_ACT_ADDR); >> + aic23_write(codec, DIGITAL_INTERFACE_ACT_ADDR, reg & (~ACT_ON)); > > Hrm. This probably deserves a comment about what DEVICE_POWER_OFF does > that the normal DAPM control doesn't. Given that there are no bypass > paths there should be no risk of interfering with DAPM but it raises a > bit of an eyebrow. There is bypass for Line and sidetone for Mic. I will add that also and will make DEVICE_POWER_OFF also as DAPM control. Will send the patches against ASoC dev tree. > >> +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) >> + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; >> + codec->hw_read = NULL; >> + ret = i2c_add_driver(&aic23b_i2c_driver); >> + if (ret != 0) >> + printk(KERN_ERR "can't add i2c driver"); >> +#else >> + /* Add other interfaces here */ >> +#endif > > If the device doesn't support anything except I2C then please remove all > the conditionals on I2C support from the driver (they don't add > anything). If the device does support other control methods then please > change this to be something like: > > #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) > ... > #endif > /* Add other interfaces here */ > > That way if someone adds support for the other control interfaces > they'll be encouraged to do it in a way that allows kernels to be built > supporting multiple systems simultaneously. > >> +#define AIC23_CACHEREGNUM 16 >> +/* Codec TLV320AIC23 */ >> +#define LEFT_LINE_VOLUME_ADDR 0x00 >> +#define RIGHT_LINE_VOLUME_ADDR 0x01 >> +#define LEFT_CHANNEL_VOLUME_ADDR 0x02 >> +#define RIGHT_CHANNEL_VOLUME_ADDR 0x03 >> +#define ANALOG_AUDIO_CONTROL_ADDR 0x04 >> +#define DIGITAL_AUDIO_CONTROL_ADDR 0x05 >> +#define POWER_DOWN_CONTROL_ADDR 0x06 >> +#define DIGITAL_AUDIO_FORMAT_ADDR 0x07 >> +#define SAMPLE_RATE_CONTROL_ADDR 0x08 >> +#define DIGITAL_INTERFACE_ACT_ADDR 0x09 >> +#define RESET_CONTROL_ADDR 0x0F > > These (and most of the rest of the defines in the header) should all be > namespaced - there's a very good chance of them colliding with other > things and this header is going to be included in the machine drivers. > >> +/* Define to set the AIC23 as the master w.r.t McBSP */ >> +#define AIC23_MASTER > >> +#ifndef DEFAULT_BITPERSAMPLE >> +#define DEFAULT_BITPERSAMPLE 16 >> +#endif > >> +#define DEFAULT_SAMPLE_RATE 44100 >> +#define CODEC_CLOCK 12000000 >> +#define AUDIO_MCBSP OMAP_MCBSP1 > > None of these should be in this driver - they should all be part of the > machine driver. > >> +struct aic23_samplerate_reg_info { >> + u32 sample_rate; >> + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ >> + u8 divider /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ >> +}; > > This could just be pushed into the driver. > >

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[alsa-devel] [PATCH 2/5] [RFC] ALSA ASOC Support for 5912 osk
by Arun KS 29 Sep '08

29 Sep '08

Adding ASOC machine driver for osk Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 3 +- sound/soc/omap/osk.c | 230 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 240 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/osk.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index d7b8939..f2e6d01 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -22,3 +22,11 @@ config SND_OMAP_SOC_OVERO help Say Y if you want to add support for SoC audio on the Gumstix Overo. +config SND_OMAP_SOC_OSK + tristate "SoC Audio support for 5912 OSK" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on 5912 OSK. + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index b96b97b..ccb3ddd 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -8,7 +8,8 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o snd-soc-overo-objs := overo.o +snd-soc-osk-objs := osk.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o - +obj-$(CONFIG_SND_OMAP_SOC_OSK) += snd-soc-osk.o diff --git a/sound/soc/omap/osk.c b/sound/soc/omap/osk.c new file mode 100644 index 0000000..f4dfe66 --- /dev/null +++ b/sound/soc/omap/osk.c @@ -0,0 +1,230 @@ +/* + * osk.c -- SoC audio for 5912 OSK + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS <arunks(a)mistralsolutions.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/clk.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 <linux/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +static struct clk *aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(aic23_mclk); +} + +static int osk_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_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +static const struct snd_soc_dapm_widget aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int osk_aic23_init(struct snd_soc_codec *codec) +{ + + /* Add OSK specific widgets */ + snd_soc_dapm_new_controls(codec, aic23_dapm_widgets, + ARRAY_SIZE(aic23_dapm_widgets)); + + /* Set up OSK specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &aic23_dai, + .init = osk_aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_osk = { + .name = "5912OSK", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .machine = &snd_soc_machine_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_aic23, +}; + +static struct platform_device *osk_snd_device; + +#define CODEC_CLOCK 12000000 + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + return -ECANCELED; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(aic23_mclk), CODEC_CLOCK, + clk_get_usecount(aic23_mclk)); + + return 0; +err1: + clk_put(aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC 5912 OSK"); +MODULE_LICENSE("GPL"); -- 1.5.3.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] ICH9 Sound
by Ronald Horn 29 Sep '08

29 Sep '08

Hello, I'm using a HP Pavilion dv7-1045eg. The Problem is I have no sound, because there is no driver witch works. Here are some reports from my system: / lspci -nv/ 00:1b.0 0403: 8086:293e (rev 03) Subsystem: 103c:30f4 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at df300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel / cat /proc/asound/cards/ 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xdf300000 irq 22 /head -n 1 /proc/asound/card0/codec*/ ==> /proc/asound/card0/codec#0 <== Codec: IDT 92HD71B7X ==> /proc/asound/card0/codec#2 <== Codec: Generic 10de ID 6 I'm using openSuse 11.0 with Kernel 2.6.25.16-0.1-pae. Maybe your project could develop some drivers. Greetings Ronald

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