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[alsa-devel] [PATCH v2 4/6] ARM: DaVinci: ASoC: Add dummy codec support
by Naresh Medisetty 16 Apr '09

16 Apr '09

From: Naresh.Medisetty <naresh(a)ti.com> McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed. This patch provides stub codec that can be used in these configurations. On DM646x EVM the McASP1 is connected to the S/PDIF out. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/spdif_transciever.c | 69 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/spdif_transciever.h | 17 ++++++++ sound/soc/davinci/Kconfig | 1 + 5 files changed, 92 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transciever.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 121d63f..154c64c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_SPDIF select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C @@ -93,6 +94,8 @@ config SND_SOC_SSM2602 config SND_SOC_TLV320AIC23 tristate +config SND_SOC_SPDIF + tristate config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d8e15a4..10d5729 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -10,6 +10,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-spdif-objs := spdif_transciever.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o @@ -43,6 +44,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c new file mode 100644 index 0000000..45f56ae --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC DaVinci DIT driver + * + * TI DaVinci audio controller can operate in DIT (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> +#include <sound/pcm.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE + + +struct snd_soc_dai dit_stub_dai = { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int davinci_dit_probe(struct platform_device *pdev) +{ + return snd_soc_register_dai(&dit_stub_dai); +} + +static int davinci_dit_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&dit_stub_dai); + return 0; +} + +static struct platform_driver davinci_dit_driver = { + .probe = davinci_dit_probe, + .remove = davinci_dit_remove, + .driver = { + .name = "davinci-dit", + .owner = THIS_MODULE, + }, +}; + +static int __init dit_modinit(void) +{ + return platform_driver_register(&davinci_dit_driver); +} + +static void __exit dit_exit(void) +{ + platform_driver_unregister(&davinci_dit_driver); +} + +module_init(dit_modinit); +module_exit(dit_exit); + diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h new file mode 100644 index 0000000..296f2eb --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.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 CODEC_STUBS_H +#define CODEC_STUBS_H + +extern struct snd_soc_dai dit_stub_dai; + +#endif /* CODEC_STUBS_H */ diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 0ae84d4..03631a7 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -26,6 +26,7 @@ config SND_DM6467_SOC_EVM depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM select SND_DAVINCI_SOC_I2S_MCASP select SND_SOC_TLV320AIC3X + select SND_SOC_SPDIF help Say Y if you want to add support for SoC audio on TI DaVinci DM6467 EVM platform. -- 1.5.6

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[alsa-devel] [PATCH 0/2] ASoC: OMAP: Finalizing the DSP mode support
by Peter Ujfalusi 16 Apr '09

16 Apr '09

Hello, This series should finally got both DSP_A and DSP_B modes working correctly on OMAP platforms. Big thanks goes to Jarkko to get (and test) the DSP_B mode in a good shape. But, the DSP modes should use single-phase mode instead of the dual-phase mode, which is intended to be used with I2S, Left-justified, Right-justified modes. The second patch adds DSP_A mode support to omap-mcbsp. --- Peter Ujfalusi (2): ASoC: OMAP: Use single-phase for DSP mode ASoC: OMAP: Add DSP_A mode support for mcbsp sound/soc/omap/omap-mcbsp.c | 38 ++++++++++++++++++++++++++------------ 1 files changed, 26 insertions(+), 12 deletions(-)

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[alsa-devel] 1.0.19 and MA78GM-US2H and also realek 1.0.19r5.11
by Jerry Geis 16 Apr '09

16 Apr '09

Hi all, I am trying to get audio over HDMI (back panel audio works) on the MA78GM-US2H. ALC 889A is the chipset. I tried 1.0.19 and from realtek 1.0.19r5.11 both of them I dont get audio over HDMI. I am using the catalyst 9.3 X driver. My alsa-info is here. http://www.alsa-project.org/db/?f=1793a79cb89ea2ad101439eaa656e5bbde165029 How can I get this working? Thanks, Jerry

1 0

[alsa-devel] ASoC fixes for 2.6.30-rc for s3c24xx
by Ben Dooks 16 Apr '09

16 Apr '09

These are fixes for the S3C24XX ASOC build, and in my view should be pushed to linus as soon as possible as I am updating the defconfigs that will cause these to be built. They are available at: git://aeryn.fluff.org.uk/bjdooks/linux.git v2630-rc2-asoc this is based on kernel 2.6.30-rc2.

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[alsa-devel] [PATCH] AOA: Convert onyx and tas codecs to new-style i2c drivers
by Jean Delvare 16 Apr '09

16 Apr '09

Hi all, The legacy i2c model is going away soon, the remaining drivers that still use it need to be converted very quickly. There are 3 sound drivers remaining: sound/aoa/codecs/onyx.c sound/aoa/codecs/tas.c sound/ppc/keywest.c I've given a try to the former two, patch below. I could only build-test it, so I would appreciate if anyone with supported hardware could test the patch. I also welcome reviews and comments, of course. I am not familiar with PowerPC so I might as well have done it wrong. Basically the idea is to move the I2C device instantiation from the codec drivers (onyx, tas) to the I2C adapter driver (i2c-powermac.) This follows the Linux device driver model, requires slightly less code, runs faster and and lets the required chip drivers be loaded by udev or similar automatically. The last driver to convert, keywest, is a mystery to me. I have a hard time understanding how it interacts with tumbler and daca. I have the feeling that it is essentially a glue module to workaround the legacy i2c model limitations, so probably it could go away entirely now, but I'm not sure how to do that in practice. Maybe tumbler and daca would be made separate i2c drivers, I'm not sure. One thing in particular which I find strange is that tumbler has references to the TAS3004 device, much like the tas codec driver. It is unclear to me whether tas is a replacement for tumbler, or if both drivers work together, or if they are for separate hardware. I would appreciate clarifications about this. Thanks. * * * * * The legacy i2c binding model is going away soon, so convert the AOA codec drivers to the new model or they'll break. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> --- drivers/i2c/busses/i2c-powermac.c | 76 +++++++++++++++++++++++++++++++ sound/aoa/codecs/onyx.c | 77 +++++++------------------------- sound/aoa/codecs/tas.c | 89 +++++++++---------------------------- 3 files changed, 116 insertions(+), 126 deletions(-) --- linux-2.6.30-rc1.orig/drivers/i2c/busses/i2c-powermac.c 2009-04-08 08:52:48.000000000 +0200 +++ linux-2.6.30-rc1/drivers/i2c/busses/i2c-powermac.c 2009-04-08 13:48:31.000000000 +0200 @@ -204,7 +204,7 @@ static int __devexit i2c_powermac_remove static int __devinit i2c_powermac_probe(struct platform_device *dev) { struct pmac_i2c_bus *bus = dev->dev.platform_data; - struct device_node *parent = NULL; + struct device_node *parent = NULL, *busnode, *devnode; struct i2c_adapter *adapter; char name[32]; const char *basename; @@ -212,6 +212,7 @@ static int __devinit i2c_powermac_probe( if (bus == NULL) return -EINVAL; + busnode = pmac_i2c_get_bus_node(bus); /* Ok, now we need to make up a name for the interface that will * match what we used to do in the past, that is basically the @@ -289,6 +290,79 @@ static int __devinit i2c_powermac_probe( } } + devnode = NULL; + while ((devnode = of_get_next_child(busnode, devnode)) != NULL) { + struct i2c_board_info info; + const u32 *addr; + + /* Instantiate I2C sound device if present */ + if (of_device_is_compatible(devnode, "pcm3052")) { + printk(KERN_DEBUG "i2c-powermac: found pcm3052\n"); + addr = of_get_property(devnode, "reg", NULL); + if (!addr) + continue; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = (*addr) >> 1; + strlcpy(info.type, "aoa_codec_onyx", I2C_NAME_SIZE); + info.platform_data = devnode; + i2c_new_device(adapter, &info); + continue; + } + + if (of_device_is_compatible(devnode, "tas3004")) { + printk(KERN_DEBUG "i2c-powermac: found tas3004\n"); + addr = of_get_property(devnode, "reg", NULL); + if (!addr) + continue; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = ((*addr) >> 1) & 0x7f; + strlcpy(info.type, "aoa_codec_tas", I2C_NAME_SIZE); + info.platform_data = devnode; + i2c_new_device(adapter, &info); + continue; + } + + /* older machines have no 'codec' node with a 'compatible' + * property that says 'tas3004', they just have a 'deq' + * node without any such property... */ + if (strcmp(devnode->name, "deq") == 0) { + printk(KERN_DEBUG "i2c-powermac: found 'deq' node\n"); + addr = of_get_property(devnode, "i2c-address", NULL); + if (!addr) + continue; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = ((*addr) >> 1) & 0x7f; + /* now, if the address doesn't match any of the two + * that a tas3004 can have, we cannot handle this. + * I doubt it ever happens but hey. */ + if (info.addr != 0x34 && info.addr != 0x35) + continue; + strlcpy(info.type, "aoa_codec_tas", I2C_NAME_SIZE); + info.platform_data = devnode; + i2c_new_device(adapter, &info); + continue; + } + + /* if that didn't work, try desperate mode for older + * machines that have stuff missing from the device tree */ + if (of_device_is_compatible(busnode, "k2-i2c")) { + unsigned short probe_onyx[] = { + 0x46, 0x47, I2C_CLIENT_END + }; + + printk(KERN_DEBUG "i2c-powermac: found k2-i2c, " + "checking if onyx chip is on it\n"); + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "aoa_codec_onyx", I2C_NAME_SIZE); + info.platform_data = devnode; + i2c_new_probed_device(adapter, &info, probe_onyx); + } + } + return rc; } --- linux-2.6.30-rc1.orig/sound/aoa/codecs/onyx.c 2009-03-26 15:08:13.000000000 +0100 +++ linux-2.6.30-rc1/sound/aoa/codecs/onyx.c 2009-04-08 13:50:11.000000000 +0200 @@ -47,7 +47,7 @@ MODULE_DESCRIPTION("pcm3052 (onyx) codec struct onyx { /* cache registers 65 to 80, they are write-only! */ u8 cache[16]; - struct i2c_client i2c; + struct i2c_client *i2c; struct aoa_codec codec; u32 initialised:1, spdif_locked:1, @@ -72,7 +72,7 @@ static int onyx_read_register(struct ony *value = onyx->cache[reg-FIRSTREGISTER]; return 0; } - v = i2c_smbus_read_byte_data(&onyx->i2c, reg); + v = i2c_smbus_read_byte_data(onyx->i2c, reg); if (v < 0) return -1; *value = (u8)v; @@ -84,7 +84,7 @@ static int onyx_write_register(struct on { int result; - result = i2c_smbus_write_byte_data(&onyx->i2c, reg, value); + result = i2c_smbus_write_byte_data(onyx->i2c, reg, value); if (!result) onyx->cache[reg-FIRSTREGISTER] = value; return result; @@ -998,10 +998,10 @@ static void onyx_exit_codec(struct aoa_c static struct i2c_driver onyx_driver; -static int onyx_create(struct i2c_adapter *adapter, - struct device_node *node, - int addr) +static int onyx_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) { + struct device_node *node = client->dev.platform_data; struct onyx *onyx; u8 dummy; @@ -1011,20 +1011,12 @@ static int onyx_create(struct i2c_adapte return -ENOMEM; mutex_init(&onyx->mutex); - onyx->i2c.driver = &onyx_driver; - onyx->i2c.adapter = adapter; - onyx->i2c.addr = addr & 0x7f; - strlcpy(onyx->i2c.name, "onyx audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&onyx->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } + onyx->i2c = client; + i2c_set_clientdata(client, onyx); /* we try to read from register ONYX_REG_CONTROL * to check if the codec is present */ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) { - i2c_detach_client(&onyx->i2c); printk(KERN_ERR PFX "failed to read control register\n"); goto fail; } @@ -1036,7 +1028,6 @@ static int onyx_create(struct i2c_adapte onyx->codec.node = of_node_get(node); if (aoa_codec_register(&onyx->codec)) { - i2c_detach_client(&onyx->i2c); goto fail; } printk(KERN_DEBUG PFX "created and attached onyx instance\n"); @@ -1046,47 +1037,10 @@ static int onyx_create(struct i2c_adapte return -EINVAL; } -static int onyx_i2c_attach(struct i2c_adapter *adapter) +static int onyx_i2c_remove(struct i2c_client *client) { - struct device_node *busnode, *dev = NULL; - struct pmac_i2c_bus *bus; - - bus = pmac_i2c_adapter_to_bus(adapter); - if (bus == NULL) - return -ENODEV; - busnode = pmac_i2c_get_bus_node(bus); + struct onyx *onyx = i2c_get_clientdata(client); - while ((dev = of_get_next_child(busnode, dev)) != NULL) { - if (of_device_is_compatible(dev, "pcm3052")) { - const u32 *addr; - printk(KERN_DEBUG PFX "found pcm3052\n"); - addr = of_get_property(dev, "reg", NULL); - if (!addr) - return -ENODEV; - return onyx_create(adapter, dev, (*addr)>>1); - } - } - - /* if that didn't work, try desperate mode for older - * machines that have stuff missing from the device tree */ - - if (!of_device_is_compatible(busnode, "k2-i2c")) - return -ENODEV; - - printk(KERN_DEBUG PFX "found k2-i2c, checking if onyx chip is on it\n"); - /* probe both possible addresses for the onyx chip */ - if (onyx_create(adapter, NULL, 0x46) == 0) - return 0; - return onyx_create(adapter, NULL, 0x47); -} - -static int onyx_i2c_detach(struct i2c_client *client) -{ - struct onyx *onyx = container_of(client, struct onyx, i2c); - int err; - - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&onyx->codec); of_node_put(onyx->codec.node); if (onyx->codec_info) @@ -1095,13 +1049,20 @@ static int onyx_i2c_detach(struct i2c_cl return 0; } +static const struct i2c_device_id onyx_i2c_id[] = { + { "aoa_codec_onyx", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, onyx_i2c_id); + static struct i2c_driver onyx_driver = { .driver = { .name = "aoa_codec_onyx", .owner = THIS_MODULE, }, - .attach_adapter = onyx_i2c_attach, - .detach_client = onyx_i2c_detach, + .probe = onyx_i2c_probe, + .remove = onyx_i2c_remove, + .id_table = onyx_i2c_id, }; static int __init onyx_init(void) --- linux-2.6.30-rc1.orig/sound/aoa/codecs/tas.c 2009-03-24 13:44:06.000000000 +0100 +++ linux-2.6.30-rc1/sound/aoa/codecs/tas.c 2009-04-08 13:50:22.000000000 +0200 @@ -82,7 +82,7 @@ MODULE_DESCRIPTION("tas codec driver for struct tas { struct aoa_codec codec; - struct i2c_client i2c; + struct i2c_client *i2c; u32 mute_l:1, mute_r:1 , controls_created:1 , drc_enabled:1, @@ -108,9 +108,9 @@ static struct tas *codec_to_tas(struct a static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data) { if (len == 1) - return i2c_smbus_write_byte_data(&tas->i2c, reg, *data); + return i2c_smbus_write_byte_data(tas->i2c, reg, *data); else - return i2c_smbus_write_i2c_block_data(&tas->i2c, reg, len, data); + return i2c_smbus_write_i2c_block_data(tas->i2c, reg, len, data); } static void tas3004_set_drc(struct tas *tas) @@ -884,10 +884,10 @@ static void tas_exit_codec(struct aoa_co static struct i2c_driver tas_driver; -static int tas_create(struct i2c_adapter *adapter, - struct device_node *node, - int addr) +static int tas_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) { + struct device_node *node = client->dev.platform_data; struct tas *tas; tas = kzalloc(sizeof(struct tas), GFP_KERNEL); @@ -896,17 +896,11 @@ static int tas_create(struct i2c_adapter return -ENOMEM; mutex_init(&tas->mtx); - tas->i2c.driver = &tas_driver; - tas->i2c.adapter = adapter; - tas->i2c.addr = addr; + tas->i2c = client; + i2c_set_clientdata(client, tas); + /* seems that half is a saner default */ tas->drc_range = TAS3004_DRC_MAX / 2; - strlcpy(tas->i2c.name, "tas audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&tas->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN); tas->codec.owner = THIS_MODULE; @@ -915,69 +909,23 @@ static int tas_create(struct i2c_adapter tas->codec.node = of_node_get(node); if (aoa_codec_register(&tas->codec)) { - goto detach; + goto fail; } printk(KERN_DEBUG "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - addr, node->full_name); + (unsigned int)client->addr, node->full_name); return 0; - detach: - i2c_detach_client(&tas->i2c); fail: mutex_destroy(&tas->mtx); kfree(tas); return -EINVAL; } -static int tas_i2c_attach(struct i2c_adapter *adapter) -{ - struct device_node *busnode, *dev = NULL; - struct pmac_i2c_bus *bus; - - bus = pmac_i2c_adapter_to_bus(adapter); - if (bus == NULL) - return -ENODEV; - busnode = pmac_i2c_get_bus_node(bus); - - while ((dev = of_get_next_child(busnode, dev)) != NULL) { - if (of_device_is_compatible(dev, "tas3004")) { - const u32 *addr; - printk(KERN_DEBUG PFX "found tas3004\n"); - addr = of_get_property(dev, "reg", NULL); - if (!addr) - continue; - return tas_create(adapter, dev, ((*addr) >> 1) & 0x7f); - } - /* older machines have no 'codec' node with a 'compatible' - * property that says 'tas3004', they just have a 'deq' - * node without any such property... */ - if (strcmp(dev->name, "deq") == 0) { - const u32 *_addr; - u32 addr; - printk(KERN_DEBUG PFX "found 'deq' node\n"); - _addr = of_get_property(dev, "i2c-address", NULL); - if (!_addr) - continue; - addr = ((*_addr) >> 1) & 0x7f; - /* now, if the address doesn't match any of the two - * that a tas3004 can have, we cannot handle this. - * I doubt it ever happens but hey. */ - if (addr != 0x34 && addr != 0x35) - continue; - return tas_create(adapter, dev, addr); - } - } - return -ENODEV; -} - -static int tas_i2c_detach(struct i2c_client *client) +static int tas_i2c_remove(struct i2c_client *client) { - struct tas *tas = container_of(client, struct tas, i2c); - int err; + struct tas *tas = i2c_get_clientdata(client); u8 tmp = TAS_ACR_ANALOG_PDOWN; - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&tas->codec); of_node_put(tas->codec.node); @@ -989,13 +937,20 @@ static int tas_i2c_detach(struct i2c_cli return 0; } +static const struct i2c_device_id tas_i2c_id[] = { + { "aoa_codec_tas", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas_i2c_id); + static struct i2c_driver tas_driver = { .driver = { .name = "aoa_codec_tas", .owner = THIS_MODULE, }, - .attach_adapter = tas_i2c_attach, - .detach_client = tas_i2c_detach, + .probe = tas_i2c_probe, + .remove = tas_i2c_remove, + .id_table = tas_i2c_id, }; static int __init tas_init(void) -- Jean Delvare

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[alsa-devel] hda Regression: jack doesn't mute speaker
by Ortwin Glück 16 Apr '09

16 Apr '09

Hi, Since 2.6.30-rc1 when I plug in headphones in the side jack the built-in speakers are not muted any more. Worked fine until 2.6.29.1. Also the "Front" and "Surround" outputs are now both labeled "Speaker" (see attached images of kmix). lspci: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) uname -a: Linux mithril 2.6.30-rc2 #2 SMP PREEMPT Wed Apr 15 22:51:36 CEST 2009 i686 Intel(R) Core(TM)2 CPU T7600 @ 2.33GHz GenuineIntel GNU/Linux dmesg: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 6 hda_codec: STAC922x, Apple subsys_id=106b1a00 input: HDA Intel Line In at Ext Left Jack as /class/input/input8 input: HDA Intel Mic at Ext Left Jack as /class/input/input9 hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. uname -a: Linux mithril 2.6.29.1 #3 SMP PREEMPT Wed Apr 15 23:03:20 CEST 2009 i686 Intel(R) Core(TM)2 CPU T7600 @ 2.33GHz GenuineIntel GNU/Linux dmesg: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 hda_codec: STAC922x, Apple subsys_id=106b1a00 Apr 15 23:20:22 mithril input: HDA Intel Line In at Ext Left Jack as /class/input/input8 Apr 15 23:20:22 mithril input: HDA Intel Mic at Ext Left Jack as /class/input/input9 Apr 15 23:20:22 mithril input: HDA Intel HP Out at Ext Left Jack as /class/input/input10 hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj.

2 4

Re: [alsa-devel] 15 April unstable does not handle wave-in
by Takashi Iwai 16 Apr '09

16 Apr '09

At Wed, 15 Apr 2009 12:13:51 -0400 (GMT-04:00), Susan Cragin wrote: > > >> >> I just downloaded and installed the unstable snapshot 15-April 9:55. > >> >> Both my sound cards are shown by asoundconf, but neither works. The wave-in wave-out is still not recognized. > >> > > >> >I suppose you are referring to CA0110? Could you elaborate? > >> > > >> >The ca0110 codec support itself has no big change. The last one was > >> >in Feb. 18. The rest is either HD-audio core or PCM core. But these > >> >have little to do with the probing... > >> > > >> > > >> >Takashi > >> > >> Please excuse my technical ignorance. Here is everything I know. > >> My hda-intel card is also not being recognized after installing alsa-driver-unstable. The only thing that works is my usb microphone. > >> I think that the program is not giving the instruction to have > >> modules inserted on bootup. > >> I have attached sample modprobe, aplay, asoundconf, modules, > >> alsa-base.conf > > > >Check /proc/asound/version. Is it the latest one? > > > >> $ modprobe snd-card-hda-intel > > > >modprobe snd-hda-intel > > > > > >Takashi > > $ sudo cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.19. > Compiled on Apr 15 2009 for kernel 2.6.28-11-generic (SMP). Then it looks OK. The problem might be a patch currently included in the unstable tree, which could be broken. I'll check it later. Meanwhile, if ca0110 worked fine with unstable tree before, it means that the hda-ca0110 patch can go in to the sound git tree. This would be a better option for both of us. thanks, Takashi

1 2

Re: [alsa-devel] SALSA library and recording
by reetesh chandran 16 Apr '09

16 Apr '09

Hi Manisha, Sorry this is not a response which will help you. I am facing similar issue. I am using SALSA and I am trying to record from a USB headset. The USB headset does have a mixer elem "Mic". But when I use snd_mixer_selem_get_capture_dB_range() or snd_mixer_selem_get_capture_volume_range() I just get zeroes. I did not do any set() functions like snd_mixer_selem_set_capture_switch_all(). Do I need to do that ? What is your source of info about how to set up the mixer for capture ? Thanks in advance Reetesh --- On Wed, 4/15/09, Manisha Sankpal <manisha.sankpal(a)gmail.com> wrote: From: Manisha Sankpal <manisha.sankpal(a)gmail.com> Subject: [alsa-devel] SALSA library and recording To: alsa-devel(a)alsa-project.org Date: Wednesday, April 15, 2009, 9:32 AM Hi All, I am using salsa library to capture the data from the sound card. The requirement is that the recording should be done from line-in port of the sound card. The selection of "Line-in" port should be done programatically. With ALSA library, I am able to select the line-in by calling snd_mixer_selem_set_capture_switch_all() function. However, this function is returning an error when SALSA library is used. Is there any other function/way to select the line-in mixer element in the SALSA API? The help would be appreciated. Thanks & regards, manisha _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

[alsa-devel] SALSA library and recording
by Manisha Sankpal 16 Apr '09

16 Apr '09

Hi All, I am using salsa library to capture the data from the sound card. The requirement is that the recording should be done from line-in port of the sound card. The selection of "Line-in" port should be done programatically. With ALSA library, I am able to select the line-in by calling snd_mixer_selem_set_capture_switch_all() function. However, this function is returning an error when SALSA library is used. Is there any other function/way to select the line-in mixer element in the SALSA API? The help would be appreciated. Thanks & regards, manisha

3 3

[alsa-devel] [PATCH] Intel8x0: Remove conflicting quirk for SSID 0x103c0934
by Daniel T Chen 16 Apr '09

16 Apr '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 While cleaning up quirks, I noticed that there is a duplicated quirk for the SSID 0x103c0934. Looking back through the bug reports, I've concluded that there is only one necessary quirk (hp_mute_led), so this patch removes the conflicting one. Thanks, Daniel T Chen -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iD8DBQFJ5nEoe9GwFciKvaMRAqf+AJ45j7iM7XYnV+bDO9sa7fj4XjpHlQCfSSAw mOJvx87TYvsMbGw5g/aS8Ds= =rvle -----END PGP SIGNATURE-----

2 1

[alsa-devel] [PATCH] Intel8x0: Add hp_only quirk for SSID 0x1028016a (Dell Inspiron 8600)
by Daniel T Chen 16 Apr '09

16 Apr '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Subject says it all. Briefly, use hp_only for another Dell Inspiron 8600. Thanks, Daniel T Chen -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iD8DBQFJ5nI5e9GwFciKvaMRAqyEAKDEzBbqoTVX84Mnh3G/5NtRsy7hbACcC/8a xoAY8903z+8x7oIB3WKoZkA= =r3JR -----END PGP SIGNATURE-----

1 0

[alsa-devel] Alsa Pulse plugin pcm->ptr initialization problem
by Dake Gu 16 Apr '09

16 Apr '09

Hi all, By looking through the pcm_pulse.c(1.0.18), I found there is a possible error in initializing the pcm->ptr. Some background: see update_ptr() function in pcm_pulse.c: Basically the pcm->ptr is increased when pa_stream_writable_size() increases. The last pa_stream_writable_size() is saved in pcm->last_size, so if new pa_stream_writable_size() value is larger than pcm->last_size, the pcm->ptr is increased. For example, the pa_stream_writable_size() increases from 800 to 1400, then pcm->ptr is increased by 400. Alsa-lib's snd_pcm_ioplug_hw_ptr_update() function is calling pulse_pointer() to get pcm->ptr, then pcm_ioplug.c calculates the hw.ptr from the changes of pulse_pointer(). The hw.ptr is used in snd_pcm_mmap_playback_avail(). However, there is an issue that during the pulse is started, the pcm->last_size jumps from 0 to a positive value, which should not be counted as pcm->ptr increase. It's causing the hw.ptr wrongly calculated. Wrong senario pa_writable (N/A) 1000 900 800 1200 pcm->ptr 0 1000 1000 1000 1400 ..... hw.ptr 0 1000 1000 1000 1400 Correct scenario: pa_writable (N/A) 1000 900 800 1200 pcm->ptr 0 0 0 0 400 ..... hw.ptr 0 0 0 0 400 If hw.ptr is incorrectly set larger than actual size, it causes avail>"real avail" and client sends more data than pulse can accept. Dake

1 0

[alsa-devel] [PATCH] ASoC: pxa-ssp: allow setting of dai format 0
by Daniel Mack 16 Apr '09

16 Apr '09

pxa_ssp_set_dai_fmt() currently has an early exit if the desired format equals the current configuration. This is correct behaviour unless this function is called with a zero value parameter for the first time. Zero is a valid value for this function, but the early exit is bogus in this case. Hence, set priv->dai_fmt to -1 in the beginning so we can configure the port. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)sirena.org.uk> Cc: pHilipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657..de22544 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -806,6 +806,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, goto err_priv; } + priv->dai_fmt = (unsigned int) -1; dai->private_data = priv; return 0; -- 1.6.2.1

2 1

[alsa-devel] [PATCH] hda_intel.c - Consolidate bitfields
by Joe Perches 15 Apr '09

15 Apr '09

Previously sent to linux-kernel. Commit fa00e046b41663cbda9b1affc0594669e5f14219 added a new bitfield not adjacent to other bitfields in the same struct. Moved the new one. Signed-off-by: Joe Perches <joe(a)perches.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bc882f8..21e99cf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -312,7 +312,6 @@ struct azx_dev { unsigned int period_bytes; /* size of the period in bytes */ unsigned int frags; /* number for period in the play buffer */ unsigned int fifo_size; /* FIFO size */ - unsigned int start_flag: 1; /* stream full start flag */ unsigned long start_jiffies; /* start + minimum jiffies */ unsigned long min_jiffies; /* minimum jiffies before position is valid */ @@ -333,6 +332,7 @@ struct azx_dev { unsigned int opened :1; unsigned int running :1; unsigned int irq_pending :1; + unsigned int start_flag: 1; /* stream full start flag */ /* * For VIA: * A flag to ensure DMA position is 0

1 0

Re: [alsa-devel] 15 April unstable does not handle wave-in
by Susan Cragin 15 Apr '09

15 Apr '09

>>> >> I just downloaded and installed the unstable snapshot 15-April 9:55. >>> >> Both my sound cards are shown by asoundconf, but neither works. The wave-in wave-out is still not recognized. >>> > >>> >I suppose you are referring to CA0110? Could you elaborate? >>> > >>> >The ca0110 codec support itself has no big change. The last one was >>> >in Feb. 18. The rest is either HD-audio core or PCM core. But these >>> >have little to do with the probing... >>> > >>> > >>> >Takashi >>> >>> Please excuse my technical ignorance. Here is everything I know. >>> My hda-intel card is also not being recognized after installing alsa-driver-unstable. The only thing that works is my usb microphone. >>> I think that the program is not giving the instruction to have >>> modules inserted on bootup. >>> I have attached sample modprobe, aplay, asoundconf, modules, >>> alsa-base.conf >> >>Check /proc/asound/version. Is it the latest one? >> >>> $ modprobe snd-card-hda-intel >> >>modprobe snd-hda-intel >> >> >>Takashi > >$ sudo cat /proc/asound/version >Advanced Linux Sound Architecture Driver Version 1.0.19. >Compiled on Apr 15 2009 for kernel 2.6.28-11-generic (SMP). >susan@ubuntu:~$ modprobe snd-hda-intel >WARNING: All config files need .conf: /etc/modprobe.d/oss-compat, it will be ignored in a future release. >susan@ubuntu:~$ I should mention something else. This happened once before, a couple of weeks ago, I think. Then header updates came out of the Ubuntu-Jaunty repository, and then the problem went away. I wish I had been paying more attention to what happened. Susan

1 0

[alsa-devel] [PATCH] add missing definitions(letters) to HD-Audio.txt
by Justin Mattock 15 Apr '09

15 Apr '09

impact: Add missing definitions(letters). Signed-off-by: Justin P. Mattock <justinmattock(a)gmail.com> --- Documentation/sound/alsa/HD-Audio.txt | 3 +++ 1 files changed,3 insertions(+),0 deletions(-) =========================================================================================== ---- Documentation/sound/alsa/HD-Audio.txt 2009-04-14 17:03:14.000000000 -0700 ++++ Documentation/sound/alsa/HD-Audio.txt 2009-04-14 21:15:22.000000000 -0700 @@ -162,7 +162,7 @@ What `model` option values are available depends on the codec chip. Check your codec chip from the codec proc file (see "Codec Proc-File" section below). It will show the vendor/product name of your codec -chip. Then, see Documentation/sound/alsa/HD-Audio-Modelstxt file, +chip. Then, see Documentation/sound/alsa/HD-Audio-Models.txt file, the section of HD-audio driver. You can find a list of codecs and `model` options belonging to each codec. For example, for Realtek ALC262 codec chip, pass `model=ultra` for devices that are compatible @@ -170,7 +170,7 @@ Thus, the first thing you can do for any brand-new, unsupported and non-working HD-audio hardware is to check HD-audio codec and several -different `model` option values. If you have a luck, some of them +different `model` option values. If you have any luck, some of them might suit with your device well. Some codecs such as ALC880 have a special model option `model=test`. -- Justin P. Mattock

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[alsa-devel] Acer 5920G: HDMI Sound
by tifereth 15 Apr '09

15 Apr '09

hi, i'm the owner of an "Acer 5920g" Laptop with the following specs: - Nvidia 8600m GT - HDMI Port - afaik. Realtek Sound Chip i've tried a lot to get the hdmi sound working, but still without success. As i can only find "requests for help" for this sound chip, perhaps you could help the lost acer 5920 owners and me. things i've tried yet: - updated kernel to vanilla 2.6.29.1 ( alsa 1.8a ) - updated alsa to 1.9 ( stable ) - updated alsa to "1.0.19.46.g516fa.506.gb750c" ( snapshot from 14.04.2009 ) - added the followng line to /etc/modprobe.d/alsa-base: "options snd-hda-intel model=auto" result of aplay -L && aplay -l ist still the same: aplay -l **** Liste von PLAYBACK Geräten **** Karte 0: Intel [HDA Intel], Gerät 0: ALC1200 Analog [ALC1200 Analog] Untergeordnete Geräte: 1/1 Untergeordnetes Gerät '0: subdevice #0 Karte 0: Intel [HDA Intel], Gerät 1: ALC1200 Digital [ALC1200 Digital] Untergeordnete Geräte: 1/1 Untergeordnetes Gerät '0: subdevice #0 aplay -L default:CARD=Intel HDA Intel, ALC1200 Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC1200 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, ALC1200 Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) alsa_info.sh output: http://www.alsa-project.org/db/?f=aa10ba4429b9cc24826d8ce1a79a0e898fd69dea Hopefully you have anymore ideas i could try, if you need anymore hardware information please contact me thanks in advance, tifereth

1 1

Re: [alsa-devel] 15 April unstable does not handle wave-in
by Susan Cragin 15 Apr '09

15 Apr '09

PLEASE IGNORE THE ONE I JUST SENT OFF-LIST. THANKS. >> >At Wed, 15 Apr 2009 07:22:08 -0400 (GMT-04:00), >> >Susan Cragin wrote: >> >> >> >> I just downloaded and installed the unstable snapshot 15-April 9:55. >> >> Both my sound cards are shown by asoundconf, but neither works. The wave-in wave-out is still not recognized. >> > >> >I suppose you are referring to CA0110? Could you elaborate? >> > >> >The ca0110 codec support itself has no big change. The last one was >> >in Feb. 18. The rest is either HD-audio core or PCM core. But these >> >have little to do with the probing... >> > >> > >> >Takashi >> >> Please excuse my technical ignorance. Here is everything I know. >> My hda-intel card is also not being recognized after installing alsa-driver-unstable. The only thing that works is my usb microphone. >> I think that the program is not giving the instruction to have >> modules inserted on bootup. >> I have attached sample modprobe, aplay, asoundconf, modules, >> alsa-base.conf > >Check /proc/asound/version. Is it the latest one? > >> $ modprobe snd-card-hda-intel > >modprobe snd-hda-intel > > >Takashi $ sudo cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.19. Compiled on Apr 15 2009 for kernel 2.6.28-11-generic (SMP). susan@ubuntu:~$ modprobe snd-hda-intel WARNING: All config files need .conf: /etc/modprobe.d/oss-compat, it will be ignored in a future release. susan@ubuntu:~$

1 0

[alsa-devel] hda-intel channels hibernate bug
by Johannes Berg 15 Apr '09

15 Apr '09

Hi, On my macbook 5,1 I need to use model=mbp3, as you may still remember. I also need to set channels to 6 to hear anything at all, probably due to the codec mismatch -- it's not really an mbp3 so that only works 80%, but I haven't gotten around to writing a better codec description. However, when I hibernate with channels set to 6, I always need to reset the channels to 6 again after resume to hear anything -- it seems the channels control isn't reset to the hardware at resume time. I suppose this bugs affects any users of this codec, regardless of whether it is actually right or wrong. johannes PS: I wanted to file a bug on the alsa bugtracker but mantisbt decided I had not entered all fields correctly and didn't give me an option to edit, so I had to reenter, which frustrated me enough to give up on mantis.

3 14

[alsa-devel] Gigabyte ALC889A Function ID: 0x01
by Jerry Geis 15 Apr '09

15 Apr '09

I am using the gigabyte MA78GM-US2H motherboard. it has the ALC889A audio. I am trying 1.0.19 and audio works out the analog back port but not over HDMI. Codec: Realtek ALC889A Address: 0 Function Id: 0x1 <--------------------------------- this is different Vendor Id: 0x10ec0885 Subsystem Id: 0x1458a002 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo I have been looking but havent found a solution yet. I tried grabbing the snapshot tar ball for today but I get errors on the compile. I had a similiar motherboard but it does not report Function ID: 01 : - HDMI audio on this board works correctly. Codec: Realtek ALC889A Address: 0 Vendor Id: 0x10ec0885 Subsystem Id: 0x1458a002 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 Any thoughts???? Jerry

2 9

Re: [alsa-devel] 15 April unstable does not handle wave-in
by Susan Cragin 15 Apr '09

15 Apr '09

>At Wed, 15 Apr 2009 07:22:08 -0400 (GMT-04:00), >Susan Cragin wrote: >> >> I just downloaded and installed the unstable snapshot 15-April 9:55. >> Both my sound cards are shown by asoundconf, but neither works. The wave-in wave-out is still not recognized. > >I suppose you are referring to CA0110? Could you elaborate? > >The ca0110 codec support itself has no big change. The last one was >in Feb. 18. The rest is either HD-audio core or PCM core. But these >have little to do with the probing... > > >Takashi Please excuse my technical ignorance. Here is everything I know. My hda-intel card is also not being recognized after installing alsa-driver-unstable. The only thing that works is my usb microphone. I think that the program is not giving the instruction to have modules inserted on bootup. I have attached sample modprobe, aplay, asoundconf, modules, alsa-base.conf Susan ---------------------------------------------------------------------- $ modprobe snd-card-hda-intel WARNING: All config files need .conf: /etc/modprobe.d/oss-compat, it will be ignored in a future release. FATAL: Module snd_card_hda_intel not found. oss-compat is already the newest version. ----------------------------------------------------------------------- $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: system [iMic USB audio system], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ------------------------------------------------------------------------ $ asoundconf list Names of available sound cards: Intel Generic (this is my Creative X-fi) system (this is my USB microphone) ------------------------------------------------------------------------ # /etc/modules: kernel modules to load at boot time. # # This file contains the names of kernel modules that should be loaded # at boot time, one per line. Lines beginning with "#" are ignored. loop lp ------------------------------------------------------------------------ alsabase.conf # autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 # Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-se$ # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } # Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } # Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 options snd-usb-caiaq index=-2 # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 # Keep snd-pcsp from beeing loaded as first soundcard options snd-pcsp index=-2

2 1

[alsa-devel] [PATCH] Automatic driver detection for Amilo 3540
by Michalis Pappas 15 Apr '09

15 Apr '09

A patch to fix the issue described in #4344 and #4352. Signed-off-by: Michalis Pappas <mepapp(a)noc.uoa.gr> --- a/alsa-kernel/pci/hda/patch_realtek.c 2009-01-19 13:08:58.000000000 +0200 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2009-04-11 14:15:44.000000000 +0300 @@ -8516,6 +8516,8 @@ SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530", ALC888_FUJITSU_XA3530), + SND_PCI_QUIRK(0x1734, 0x1137, "Fujitsu AMILO Xa3540", + ALC888_FUJITSU_XA3530), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),

2 1

Re: [alsa-devel] Ubuntu sound problem - http://ubuntuforums.org/showthread.php?t=1073090
by crimsunkg＠yahoo.com 15 Apr '09

15 Apr '09

--- On Mon, 4/13/09, jesse fillion <jessefillion24(a)hotmail.com> wrote: > I have attached my alsa-info output. You should try the latest alsa-driver snapshot (http://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sn…), which contains quite a few fixes.

2 1

[alsa-devel] [PATCH] AOA: Convert onyx and tas codecs to new-style i2c drivers (v3)
by Jean Delvare 15 Apr '09

15 Apr '09

The legacy i2c binding model is going away soon, so convert the AOA codec drivers to the new model or they'll break. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> Tested-by: Andreas Schwab <schwab(a)linux-m68k.org> --- Johannes, this is a reworked patch which assumes that the onyx codec probing can fail. It's not exactly clean but that's the easiest approach for now. Can you please give it a try? Thanks. sound/aoa/codecs/onyx.c | 76 ++++++++++++++++++++++++++++++++--------------- sound/aoa/codecs/tas.c | 66 +++++++++++++++++++++++++--------------- 2 files changed, 95 insertions(+), 47 deletions(-) --- linux-2.6.30-rc2.orig/sound/aoa/codecs/onyx.c 2009-04-15 09:55:20.000000000 +0200 +++ linux-2.6.30-rc2/sound/aoa/codecs/onyx.c 2009-04-15 14:07:50.000000000 +0200 @@ -47,7 +47,7 @@ MODULE_DESCRIPTION("pcm3052 (onyx) codec struct onyx { /* cache registers 65 to 80, they are write-only! */ u8 cache[16]; - struct i2c_client i2c; + struct i2c_client *i2c; struct aoa_codec codec; u32 initialised:1, spdif_locked:1, @@ -72,7 +72,7 @@ static int onyx_read_register(struct ony *value = onyx->cache[reg-FIRSTREGISTER]; return 0; } - v = i2c_smbus_read_byte_data(&onyx->i2c, reg); + v = i2c_smbus_read_byte_data(onyx->i2c, reg); if (v < 0) return -1; *value = (u8)v; @@ -84,7 +84,7 @@ static int onyx_write_register(struct on { int result; - result = i2c_smbus_write_byte_data(&onyx->i2c, reg, value); + result = i2c_smbus_write_byte_data(onyx->i2c, reg, value); if (!result) onyx->cache[reg-FIRSTREGISTER] = value; return result; @@ -996,12 +996,45 @@ static void onyx_exit_codec(struct aoa_c onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); } -static struct i2c_driver onyx_driver; - static int onyx_create(struct i2c_adapter *adapter, struct device_node *node, int addr) { + struct i2c_board_info info; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "aoa_codec_onyx", I2C_NAME_SIZE); + info.addr = addr; + info.platform_data = node; + client = i2c_new_device(adapter, &info); + if (!client) + return -ENODEV; + + /* + * We know the driver is already loaded, so the device should be + * already bound. If not it means binding failed, which suggests + * the device doesn't really exist and should be deleted. + * Ideally this would be replaced by better checks _before_ + * instantiating the device. + */ + if (!client->driver) { + i2c_unregister_device(client); + return -ENODEV; + } + + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&client->detected, &client->driver->clients); + return 0; +} + +static int onyx_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.platform_data; struct onyx *onyx; u8 dummy; @@ -1011,20 +1044,12 @@ static int onyx_create(struct i2c_adapte return -ENOMEM; mutex_init(&onyx->mutex); - onyx->i2c.driver = &onyx_driver; - onyx->i2c.adapter = adapter; - onyx->i2c.addr = addr & 0x7f; - strlcpy(onyx->i2c.name, "onyx audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&onyx->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } + onyx->i2c = client; + i2c_set_clientdata(client, onyx); /* we try to read from register ONYX_REG_CONTROL * to check if the codec is present */ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) { - i2c_detach_client(&onyx->i2c); printk(KERN_ERR PFX "failed to read control register\n"); goto fail; } @@ -1036,14 +1061,14 @@ static int onyx_create(struct i2c_adapte onyx->codec.node = of_node_get(node); if (aoa_codec_register(&onyx->codec)) { - i2c_detach_client(&onyx->i2c); goto fail; } printk(KERN_DEBUG PFX "created and attached onyx instance\n"); return 0; fail: + i2c_set_clientdata(client, NULL); kfree(onyx); - return -EINVAL; + return -ENODEV; } static int onyx_i2c_attach(struct i2c_adapter *adapter) @@ -1080,28 +1105,33 @@ static int onyx_i2c_attach(struct i2c_ad return onyx_create(adapter, NULL, 0x47); } -static int onyx_i2c_detach(struct i2c_client *client) +static int onyx_i2c_remove(struct i2c_client *client) { - struct onyx *onyx = container_of(client, struct onyx, i2c); - int err; + struct onyx *onyx = i2c_get_clientdata(client); - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&onyx->codec); of_node_put(onyx->codec.node); if (onyx->codec_info) kfree(onyx->codec_info); + i2c_set_clientdata(client, onyx); kfree(onyx); return 0; } +static const struct i2c_device_id onyx_i2c_id[] = { + { "aoa_codec_onyx", 0 }, + { } +}; + static struct i2c_driver onyx_driver = { .driver = { .name = "aoa_codec_onyx", .owner = THIS_MODULE, }, .attach_adapter = onyx_i2c_attach, - .detach_client = onyx_i2c_detach, + .probe = onyx_i2c_probe, + .remove = onyx_i2c_remove, + .id_table = onyx_i2c_id, }; static int __init onyx_init(void) --- linux-2.6.30-rc2.orig/sound/aoa/codecs/tas.c 2009-04-15 09:55:20.000000000 +0200 +++ linux-2.6.30-rc2/sound/aoa/codecs/tas.c 2009-04-15 09:56:39.000000000 +0200 @@ -82,7 +82,7 @@ MODULE_DESCRIPTION("tas codec driver for struct tas { struct aoa_codec codec; - struct i2c_client i2c; + struct i2c_client *i2c; u32 mute_l:1, mute_r:1 , controls_created:1 , drc_enabled:1, @@ -108,9 +108,9 @@ static struct tas *codec_to_tas(struct a static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data) { if (len == 1) - return i2c_smbus_write_byte_data(&tas->i2c, reg, *data); + return i2c_smbus_write_byte_data(tas->i2c, reg, *data); else - return i2c_smbus_write_i2c_block_data(&tas->i2c, reg, len, data); + return i2c_smbus_write_i2c_block_data(tas->i2c, reg, len, data); } static void tas3004_set_drc(struct tas *tas) @@ -882,12 +882,34 @@ static void tas_exit_codec(struct aoa_co } -static struct i2c_driver tas_driver; - static int tas_create(struct i2c_adapter *adapter, struct device_node *node, int addr) { + struct i2c_board_info info; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "aoa_codec_tas", I2C_NAME_SIZE); + info.addr = addr; + info.platform_data = node; + + client = i2c_new_device(adapter, &info); + if (!client) + return -ENODEV; + + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&client->detected, &client->driver->clients); + return 0; +} + +static int tas_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.platform_data; struct tas *tas; tas = kzalloc(sizeof(struct tas), GFP_KERNEL); @@ -896,17 +918,11 @@ static int tas_create(struct i2c_adapter return -ENOMEM; mutex_init(&tas->mtx); - tas->i2c.driver = &tas_driver; - tas->i2c.adapter = adapter; - tas->i2c.addr = addr; + tas->i2c = client; + i2c_set_clientdata(client, tas); + /* seems that half is a saner default */ tas->drc_range = TAS3004_DRC_MAX / 2; - strlcpy(tas->i2c.name, "tas audio codec", I2C_NAME_SIZE); - - if (i2c_attach_client(&tas->i2c)) { - printk(KERN_ERR PFX "failed to attach to i2c\n"); - goto fail; - } strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN); tas->codec.owner = THIS_MODULE; @@ -915,14 +931,12 @@ static int tas_create(struct i2c_adapter tas->codec.node = of_node_get(node); if (aoa_codec_register(&tas->codec)) { - goto detach; + goto fail; } printk(KERN_DEBUG "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - addr, node->full_name); + (unsigned int)client->addr, node->full_name); return 0; - detach: - i2c_detach_client(&tas->i2c); fail: mutex_destroy(&tas->mtx); kfree(tas); @@ -970,14 +984,11 @@ static int tas_i2c_attach(struct i2c_ada return -ENODEV; } -static int tas_i2c_detach(struct i2c_client *client) +static int tas_i2c_remove(struct i2c_client *client) { - struct tas *tas = container_of(client, struct tas, i2c); - int err; + struct tas *tas = i2c_get_clientdata(client); u8 tmp = TAS_ACR_ANALOG_PDOWN; - if ((err = i2c_detach_client(client))) - return err; aoa_codec_unregister(&tas->codec); of_node_put(tas->codec.node); @@ -989,13 +1000,20 @@ static int tas_i2c_detach(struct i2c_cli return 0; } +static const struct i2c_device_id tas_i2c_id[] = { + { "aoa_codec_tas", 0 }, + { } +}; + static struct i2c_driver tas_driver = { .driver = { .name = "aoa_codec_tas", .owner = THIS_MODULE, }, .attach_adapter = tas_i2c_attach, - .detach_client = tas_i2c_detach, + .probe = tas_i2c_probe, + .remove = tas_i2c_remove, + .id_table = tas_i2c_id, }; static int __init tas_init(void) -- Jean Delvare

2 2

[alsa-devel] 15 April unstable does not handle wave-in
by Susan Cragin 15 Apr '09

15 Apr '09

I just downloaded and installed the unstable snapshot 15-April 9:55. Both my sound cards are shown by asoundconf, but neither works. The wave-in wave-out is still not recognized. (USB sound does work.) System: Asus G1S running Ubuntu Jaunty with all updates (most recent kernel is only 2.6.28-11-Generic). Sound cards present: hda-intel / Realtek ALC660-VD hda-audio-generic / Creative CA0110-IBG (Creative X-fi)

2 1

[alsa-devel] ASoC: DMA Platform driver for i.mx27
by javier Martin 15 Apr '09

15 Apr '09

This still does not support bidirectional audio but works when used playback or capture separatelly. SDMA API used in i.mx31 dma driver has been moved to DMA API. Please, have patience with me, this is a very experimental version and I am new to this world (audio codecs and linux in general). Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> /* * linux/sound/arm/mx27-pcm.c -- ALSA SoC interface for the Freescale i.MX27 CPU * * Copyright 2009 Vista Silicon S.L. * Author: Javier Martin * javier.martin(a)vista-silicon.com * * Based on mxc-pcm.c by Liam Girdwood, (C) 2006 Wolfson Microelectronics PLC. * and on mxc-alsa-mc13783 (C) 2006 Freescale. * * 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. * * Revision history * 14th April 2009 Initial version. * */ #include <linux/module.h> #include <linux/init.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dma-mapping.h> #include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <asm/dma.h> #include <asm/hardware.h> #include "imx27-pcm.h" #include "imx27-ssi.h" //For debugging /* debug */ #define IMX_DEBUG 0 #if IMX_DEBUG #define dbg(format, arg...) printk(format, ## arg) #else #define dbg(format, arg...) #endif /* Taken from drivers/mxc/ssi/registers.h just provisionally */ #define MXC_SSI1STX0 0x00 #define MXC_SSI1SRX0 0x08 #define MXC_SSI2STX0 0x00 #define MXC_SSI2SRX0 0x08 static const struct snd_pcm_hardware mx27_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S16_LE, .buffer_bytes_max = 32 * 1024, .period_bytes_min = 64, .period_bytes_max = 8 * 1024, .periods_min = 2, .periods_max = 255, .fifo_size = 0, }; struct mx27_runtime_data { int dma_ch_tx; //DMA channel number for tx int dma_ch_rx; //DMA channel number for rx int active; //Is the stream active? unsigned int period; //Period number unsigned int periods; //¿? int tx_spin; //¿? spinlock_t dma_lock; //DMA spinlock struct mx27_pcm_dma_param *dma_params; //FIXME this is SDMA api }; /*! * This function stops the current dma transfer for playback * and clears the dma pointers. * * @param substream pointer to the structure of the current stream. * */ static int audio_stop_dma(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size); unsigned int offset = dma_size * prtd->periods; unsigned long flags; spin_lock_irqsave(&prtd->dma_lock, flags); // dbg("MX27 : audio_stop_dma active = 0\n"); prtd->active = 0; prtd->period = 0; prtd->periods = 0; /* this stops the dma channel and clears the buffer ptrs */ if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if(mxc_dma_disable(prtd->dma_ch_tx)<0) printk("audio_stop_dma: error when stopping dma transfer in channel %d \n", prtd->dma_ch_tx); } else { if(mxc_dma_disable(prtd->dma_ch_rx)<0) printk("audio_stop_dma: error when stopping dma transfer in channel %d \n", prtd->dma_ch_rx); } spin_unlock_irqrestore(&prtd->dma_lock, flags); return 0; } /*! * This function is called whenever a new audio block needs to be * transferred to wm8974. The function receives the address and the size * of the new block and start a new DMA transfer. * * @param substream pointer to the structure of the current stream. * */ static int dma_new_period(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; unsigned int dma_size; unsigned int offset; int ret=0; mxc_dma_requestbuf_t dma_request; if (prtd->active){ memset(&dma_request, 0, sizeof(mxc_dma_requestbuf_t));; dma_size = frames_to_bytes(runtime, runtime->period_size); printk("dma_new_period: prtd->period (%x) runtime->periods (%d)\n", prtd->period,runtime->periods); printk("dma_new_period: runtime->period_size (%d) dma_size (%d)\n", (unsigned int)runtime->period_size, runtime->dma_bytes); offset = dma_size * prtd->period; snd_assert(dma_size <= mx27_pcm_hardware.period_bytes_max); if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { printk("\ndma_new_period (playback): runtime->dma_area = %x\noffset = %x\ndma_size = %x\n", runtime->dma_area, offset, dma_size); printk("\ndma_new_period : dma_addr is %x\n", (runtime->dma_addr + offset)); dma_request.src_addr = (dma_addr_t)(runtime->dma_addr + offset); #ifdef CONFIG_SND_SOC_MX27_SSI1 dma_request.dst_addr = (dma_addr_t) (SSI1_BASE_ADDR + MXC_SSI1STX0); printk("dst_addr is %x\n", dma_request.dst_addr); printk("src_addr is %x\n", dma_request.src_addr); #else dma_request.dst_addr = (dma_addr_t) (SSI2_BASE_ADDR + MXC_SSI2STX0); #endif } else { printk("\ndma_new_period (capture): runtime->dma_area = %x\noffset = %x\ndma_size = %x\n", runtime->dma_area, offset, dma_size); printk("\ndma_new_period: dma_addr is %x\n", (runtime->dma_addr + offset)); dma_request.dst_addr = (dma_addr_t) (runtime->dma_addr + offset); #ifdef CONFIG_SND_SOC_MX27_SSI1 dma_request.src_addr = (dma_addr_t) (SSI1_BASE_ADDR + MXC_SSI1SRX0); #else dma_request.src_addr = (dma_addr_t) (SSI2_BASE_ADDR + MXC_SSI2SRX0); #endif } dma_request.num_of_bytes = dma_size; //8 printk("dma_new_period: Start DMA offset (%d) size (%d)\n", offset, runtime->dma_bytes); if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { printk("dma_neq_period: configuring tx dma\n"); ret = mxc_dma_config(prtd->dma_ch_tx, &dma_request, 1, MXC_DMA_MODE_WRITE); if(ret<0) { printk("dma_new_period: buffer could not be added for playback transfer\n"); return ret; } printk("dma_new_period: enable tx dma transfer\n"); ret = mxc_dma_enable(prtd->dma_ch_tx); if(ret<0) { printk("dma_new_period: cannot queue DMA buffer\ (%i)\n", ret); return ret; } } else { ret = mxc_dma_config(prtd->dma_ch_rx, &dma_request, 1, MXC_DMA_MODE_READ); if(ret<0) { printk("dma_new_period: buffer could not be added for capture transfer\n"); return ret; } printk("dma_new_period: enable dma transfer\n"); ret = mxc_dma_enable(prtd->dma_ch_rx); if(ret<0) { printk("dma_new_period: cannot queue DMA buffer\ (%i)\n", ret); return ret; } } printk("dma_new_period: transfer enabled\n src_addr = %x\n dst_addr = %x\n num_of_byes = %d\n", (void *) dma_request.src_addr, (void *) dma_request.dst_addr, dma_request.num_of_bytes); prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ prtd->period++; prtd->period %= runtime->periods; } return ret; } /*! * This is a callback which will be called * when a TX transfer finishes. The call occurs * in interrupt context. * * @param dat pointer to the structure of the current stream. * */ static void audio_dma_irq(void *data, int error, unsigned int count) { struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; struct mx27_runtime_data *prtd; unsigned int dma_size; unsigned int previous_period; unsigned int offset; substream = data; runtime = substream->runtime; prtd = runtime->private_data; previous_period = prtd->periods; dma_size = frames_to_bytes(runtime, runtime->period_size); offset = dma_size * previous_period; prtd->tx_spin = 0; prtd->periods++; prtd->periods %= runtime->periods; // printk("audio_dma_irq: irq per %d offset %x\n", prtd->periods, offset); /* * Give back to the CPU the access to the non cached memory */ // if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) // dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, // DMA_TO_DEVICE); // else // dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, // DMA_FROM_DEVICE); /* * If we are getting a callback for an active stream then we inform * the PCM middle layer we've finished a period */ if (prtd->active) snd_pcm_period_elapsed(substream); /* * Trig next DMA transfer */ dma_new_period(substream); } /*! * This function configures the hardware to allow audio * playback operations. It is called by ALSA framework. * * @param substream pointer to the structure of the current stream. * * @return 0 on success, -1 otherwise. */ static int snd_mxc_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; prtd->period = 0; prtd->periods = 0; return 0; } static int mxc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; int ret; if((ret=snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) { printk("imx27-pcm: failed to malloc pcm pages\n"); return ret; } printk("mxc_pcm_hw_params: snd_pcm_lib_malloc pages has return OK\n"); printk("IMX27: snd_imx27_audio_hw_params runtime->dma_addr 0x(%x)\n", (unsigned int)runtime->dma_addr); printk("IMX27: snd_imx27_audio_hw_params runtime->dma_area 0x(%x)\n", (unsigned int)runtime->dma_area); printk("IMX27: snd_imx27_audio_hw_params runtime->dma_bytes 0x(%x)\n", (unsigned int)runtime->dma_bytes); return ret; } static int mxc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; mxc_dma_free(prtd->dma_ch_tx); mxc_dma_free(prtd->dma_ch_rx); snd_pcm_lib_free_pages(substream); return 0; } static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct mx27_runtime_data *prtd = substream->runtime->private_data; int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: prtd->tx_spin = 0; /* requested stream startup */ prtd->active = 1; printk("mxc_pcm_trigger: starting dma_new_period\n"); ret = dma_new_period(substream); break; case SNDRV_PCM_TRIGGER_STOP: /* requested stream shutdown */ printk("mxc_pcm_trigger: stopping dma transfer\n"); ret = audio_stop_dma(substream); break; default: ret = -EINVAL; break; } return ret; } static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd = runtime->private_data; unsigned int offset = 0; printk("mxc_pcm_pointer:\n runtime->period_size=%d\n prtd->periods=%d\n runtime->buffer_size=%d\n prtd->tx_spin=%d\n", runtime->period_size, prtd->periods, runtime->buffer_size, prtd->tx_spin); /* tx_spin value is used here to check if a transfer is active */ if (prtd->tx_spin){ offset = (runtime->period_size * (prtd->periods)) + (runtime->period_size >> 1); if (offset >= runtime->buffer_size) offset = runtime->period_size >> 1; } else { offset = (runtime->period_size * (prtd->periods)); if (offset >= runtime->buffer_size) offset = 0; } printk("mxc_pcm_pointer: pointer offset %x\n", offset); return offset; } //Alocation of substream->runtime->private_data static int mxc_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mx27_runtime_data *prtd; int ret; snd_soc_set_runtime_hwparams(substream, &mx27_pcm_hardware); //?? if ((ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return ret; if((prtd = kzalloc(sizeof(struct mx27_runtime_data), GFP_KERNEL)) == NULL) { ret = -ENOMEM; goto out; } runtime->private_data = prtd; if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { #ifdef CONFIG_SND_SOC_MX27_SSI1 printk("mxc_pcm_open: Requesting MXC_DMA_SSI1_16BIT_TX0 dma channel\n"); prtd->dma_ch_tx = mxc_dma_request(MXC_DMA_SSI1_16BIT_TX0, "ALSA TX DMA"); //MXC_DMA_TEST_RAM2RAM #else prtd->dma_ch_tx = mxc_dma_request(MXC_DMA_SSI2_16BIT_TX0, "ALSA TX DMA"); #endif if(prtd->dma_ch_tx < 0) { printk("error requesting a write(dma to device) dma channel\n"); return ret; } //Set dma callback printk("mxc_pcm_open: Setting tx dma callback function\n"); ret = mxc_dma_callback_set(prtd->dma_ch_tx, (mxc_dma_callback_t) audio_dma_irq, (void *)substream); if(ret<0) { printk("error setting dma callback function\n"); return ret; } return 0; } else { #ifdef CONFIG_SND_SOC_MX27_SSI1 printk("mxc_pcm_open: Requesting MXC_DMA_SSI1_16BIT_RX0 dma channel\n"); prtd->dma_ch_rx = mxc_dma_request(MXC_DMA_SSI1_16BIT_RX0, "ALSA RX DMA"); #else prtd->dma_ch_rx = mxc_dma_request(MXC_DMA_SSI2_16BIT_RX0, "ALSA RX DMA"); #endif if(prtd->dma_ch_rx < 0) { printk("error requesting a read(dma from device) dma channel\n"); return ret; } //Set dma callback printk("mxc_pcm_open: Setting rx dma callback function\n"); if(mxc_dma_callback_set(prtd->dma_ch_rx, (mxc_dma_callback_t) audio_dma_irq, (void *)substream)<0) printk("error setting dma callback function\n"); return 0; } out: return ret; } static int mxc_pcm_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct mxc_runtime_data *prtd = runtime->private_data; kfree(prtd); return 0; } static int mxc_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { struct snd_pcm_runtime *runtime = substream->runtime; return dma_mmap_writecombine(substream->pcm->card->dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); } struct snd_pcm_ops mxc_pcm_ops = { .open = mxc_pcm_open, .close = mxc_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = mxc_pcm_hw_params, .hw_free = mxc_pcm_hw_free, .prepare = snd_mxc_prepare, .trigger = mxc_pcm_trigger, .pointer = mxc_pcm_pointer, .mmap = mxc_pcm_mmap, }; static u64 mxc_pcm_dmamask = 0xffffffff; static int imx31_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = mx27_pcm_hardware.buffer_bytes_max; buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; buf->private_data = NULL; //Reserve uncached-buffered memory area for DMA buf->area = dma_alloc_writecombine(pcm->card->dev, size, &buf->addr, GFP_KERNEL); printk("imx27_pcm: preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", (void *) buf->area, (void *) buf->addr, size); if (!buf->area) return -ENOMEM; buf->bytes = size; return 0; } static void imx31_pcm_free_dma_buffers(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; int stream; for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; if (!substream) continue; buf = &substream->dma_buffer; if (!buf->area) continue; dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; } } int mxc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; if (!card->dev->dma_mask) card->dev->dma_mask = &mxc_pcm_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (dai->playback.channels_min) { ret = imx31_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); printk("mxc_pcm_new: preallocate playback buffer\n"); if (ret) goto out; } if (dai->capture.channels_min) { ret = imx31_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); printk("mxc_pcm_new: preallocate capture buffer\n"); if (ret) goto out; } out: return ret; } struct snd_soc_platform imx27_soc_platform = { .name = "mxc-audio", .pcm_ops = &mxc_pcm_ops, .pcm_new = mxc_pcm_new, .pcm_free = imx31_pcm_free_dma_buffers, }; EXPORT_SYMBOL_GPL(imx27_soc_platform); MODULE_AUTHOR("Javier Martin"); MODULE_DESCRIPTION("Freescale i.MX27 PCM DMA module"); MODULE_LICENSE("GPL"); /* * mx27-pcm.h :- ASoC platform header for Freescale i.MX27 * * 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 _MXC_PCM_H #define _MXC_PCM_H #include <asm/arch/dma.h> /* AUDMUX regs definition - MOVE to asm/arch when stable #define AUDMUX_IO_BASE_ADDR IO_ADDRESS(AUDMUX_BASE_ADDR) #define DAM_PTCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x00))) #define DAM_PDCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x04))) #define DAM_PTCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x08))) #define DAM_PDCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x0C))) #define DAM_PTCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x10))) #define DAM_PDCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x14))) #define DAM_PTCR4 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x18))) #define DAM_PDCR4 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x1C))) #define DAM_PTCR5 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x20))) #define DAM_PDCR5 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x24))) #define DAM_PTCR6 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x28))) #define DAM_PDCR6 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x2C))) #define DAM_PTCR7 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x30))) #define DAM_PDCR7 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x34))) #define DAM_CNMCR (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x38))) #define DAM_PTCR(a) (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + a*8))) #define DAM_PDCR(a) (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 4 + a*8))) #define AUDMUX_PTCR_TFSDIR (1 << 31) #define AUDMUX_PTCR_TFSSEL(x, y) ((x << 30) | (((y - 1) & 0x7) << 27)) #define AUDMUX_PTCR_TCLKDIR (1 << 26) #define AUDMUX_PTCR_TCSEL(x, y) ((x << 25) | (((y - 1) & 0x7) << 22)) #define AUDMUX_PTCR_RFSDIR (1 << 21) #define AUDMUX_PTCR_RFSSEL(x, y) ((x << 20) | (((y - 1) & 0x7) << 17)) #define AUDMUX_PTCR_RCLKDIR (1 << 16) #define AUDMUX_PTCR_RCSEL(x, y) ((x << 15) | (((y - 1) & 0x7) << 12)) #define AUDMUX_PTCR_SYN (1 << 11) #define AUDMUX_FROM_TXFS 0 #define AUDMUX_FROM_RXFS 1 #define AUDMUX_PDCR_RXDSEL(x) (((x - 1) & 0x7) << 13) #define AUDMUX_PDCR_TXDXEN (1 << 12) #define AUDMUX_PDCR_MODE(x) (((x) & 0x3) << 8) #define AUDMUX_PDCR_INNMASK(x) (((x) & 0xff) << 0) #define AUDMUX_CNMCR_CEN (1 << 18) #define AUDMUX_CNMCR_FSPOL (1 << 17) #define AUDMUX_CNMCR_CLKPOL (1 << 16) #define AUDMUX_CNMCR_CNTHI(x) (((x) & 0xff) << 8) #define AUDMUX_CNMCR_CNTLOW(x) (((x) & 0xff) << 0) */ /* Previous defines are only valid for imx31 */ #define AUDMUX_IO_BASE_ADDR IO_ADDRESS(AUDMUX_BASE_ADDR) #define DAM_HPCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x00))) #define DAM_HPCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x04))) #define DAM_HPCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x08))) #define DAM_PPCR1 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x10))) #define DAM_PPCR2 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x14))) #define DAM_PPCR3 (*((volatile u32 *)(AUDMUX_IO_BASE_ADDR + 0x1C))) #define AUDMUX_HPCR_TFSDIR (1 << 31) #define AUDMUX_HPCR_TCLKDIR (1 << 30) #define AUDMUX_HPCR_TFCSEL(x) (((x) & 0xff) << 26) #define AUDMUX_HPCR_RXDSEL(x) (((x) & 0x7) << 13) #define AUDMUX_HPCR_SYN (1 << 12) #define AUDMUX_PPCR_TFSDIR (1 << 31) #define AUDMUX_PPCR_TCLKDIR (1 << 30) #define AUDMUX_PPCR_TFCSEL(x) (((x) & 0xff) << 26) #define AUDMUX_PPCR_RXDSEL(x) (((x) & 0x7) << 13) #define AUDMUX_PPCR_SYN (1 << 12) struct mx27_pcm_dma_params { char *name; /* stream identifier */ mxc_dma_requestbuf_t buf_params; /*DMA buffer params */ }; extern struct snd_soc_dai mxc_ssi_dai[3]; /* platform data */ extern struct snd_soc_platform imx27_soc_platform; #endif

2 5

[alsa-devel] General protection fault when unloading snd_hda_intel on 2.6.30-rc1+
by Carlos R. Mafra 15 Apr '09

15 Apr '09

If I try on my Vaio VGN-FZ240E, using the latest kernel 2.6.30-rc1-00191-gd848223 modprobe -r snd_hda_intel I get the following trace (copied from dmesg): general protection fault: 0000 [#1] SMP last sysfs file: /sys/devices/system/cpu/cpu0/cpufreq/scaling_cur_freq CPU 1 Modules linked in: nvram uvcvideo videodev v4l1_compat v4l2_compat_ioctl32 snd_hda_codec_idt usbhid snd_hda_intel(-) snd_hda_codec snd_hwdep snd_pcm sr_mod sky2 i2c_i801 iwlagn snd_timer sg snd_page_alloc evdev ata_piix ahci libata sd_mod scsi_mod uhci_hcd ohci_hcd ehci_hcd usbcore [last unloaded: scsi_wait_scan] Pid: 9063, comm: modprobe Not tainted 2.6.30-rc1-00191-gd848223 #48 VGN-FZ240E RIP: 0010:[<ffffffff8044bf90>] [<ffffffff8044bf90>] input_event+0x40/0xa0 RSP: 0018:ffff88006f283a78 EFLAGS: 00010293 RAX: 0000000000000008 RBX: 0000000000000003 RCX: 0000000000000000 RDX: 0000000000000007 RSI: 0000000000000005 RDI: 697665646632785c RBP: ffff88006f283aa8 R08: 0000000004d6f76c R09: 0000000000000001 R10: 0000000000000000 R11: 0000000000000001 R12: ffffffff805bb97c R13: 0000000000000005 R14: 697665646632785c R15: 0000000000000000 FS: 00007fc6c1e1c6f0(0000) GS:ffff880001028000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b CR2: 000000000071bf40 CR3: 000000007e2d4000 CR4: 00000000000006e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process modprobe (pid: 9063, threadinfo ffff88006f282000, task ffff88006f206840) Stack: ffff88006f283aa8 0000000000000003 ffffffff805bb97c ffff88007e811cc0 0000000000000001 0000000000000000 ffff88006f283ae8 ffffffff80475612 ffff88007f241ae0 0000000000000002 ffff88007d5eb010 0000000000000000 Call Trace: [<ffffffff80475612>] snd_jack_report+0x92/0xa0 [<ffffffffa0160047>] stac92xx_unsol_event+0x2e7/0x5e0 [snd_hda_codec_idt] [<ffffffffa015f528>] stac_issue_unsol_event+0x58/0x70 [snd_hda_codec_idt] [<ffffffffa0160796>] stac92xx_init+0x186/0x660 [snd_hda_codec_idt] [<ffffffffa0160c90>] stac92xx_resume+0x20/0x60 [snd_hda_codec_idt] [<ffffffffa013c973>] hda_call_codec_resume+0x53/0x90 [snd_hda_codec] [<ffffffffa013ca50>] snd_hda_power_up+0x40/0x70 [snd_hda_codec] [<ffffffffa013c2b5>] snd_hda_codec_write+0x45/0xa0 [snd_hda_codec] [<ffffffffa013c37a>] restore_pincfgs+0x6a/0xa0 [snd_hda_codec] [<ffffffffa013ed75>] snd_hda_codec_free+0x35/0x120 [snd_hda_codec] [<ffffffffa01403f8>] snd_hda_bus_free+0x68/0xb0 [snd_hda_codec] [<ffffffffa01405e4>] snd_hda_bus_dev_free+0x14/0x20 [snd_hda_codec] [<ffffffff80473e61>] snd_device_free+0x71/0xd0 [<ffffffff80473f24>] snd_device_free_all+0x64/0x70 [<ffffffff8046e16a>] snd_card_do_free+0x3a/0xd0 [<ffffffff8046ee39>] snd_card_free+0xa9/0xd0 [<ffffffffa015031e>] azx_remove+0x18/0x2a [snd_hda_intel] [<ffffffff8036506f>] pci_device_remove+0x2f/0x60 [<ffffffff8040ec8c>] __device_release_driver+0x6c/0xb0 [<ffffffff8040ed78>] driver_detach+0xa8/0xb0 [<ffffffff8040de48>] bus_remove_driver+0x98/0xd0 [<ffffffff8040f357>] driver_unregister+0x47/0x60 [<ffffffff8036530c>] pci_unregister_driver+0x4c/0xc0 [<ffffffffa0150304>] alsa_card_azx_exit+0x10/0x12 [snd_hda_intel] [<ffffffff8026787b>] sys_delete_module+0x18b/0x240 [<ffffffff80257be9>] ? up_write+0x9/0x10 [<ffffffff8020be2b>] system_call_fastpath+0x16/0x1b Code: 89 5d d8 4c 89 65 e0 49 89 fe 41 89 f5 41 89 cf 83 fe 1f 76 16 48 8b 5d d8 4c 8b 65 e0 4c 8b 6d e8 4c 8b 75 f0 4c 8b 7d f8 c9 c3 <0f> a3 77 20 19 c0 85 c0 74 e0 48 8d 9f f8 06 00 00 89 55 d0 48 RIP [<ffffffff8044bf90>] input_event+0x40/0xa0 RSP <ffff88006f283a78> ---[ end trace 7fa0517f56c40085 ]--- After that, 'cat /proc/modules' contains this line: snd_hda_intel 28648 0 - Unloading 0xffffffffa014e000 and the HDA driver never finishes that advertised "Unloading". It also stops working, [mafra@Pilar:linux-2.6]$ alsamixer alsamixer: function snd_ctl_open failed for default: No such file or directory The complete dmesg is available at http://www.aei.mpg.de/~crmafra/dmesg_hda_intel.txt and from lspci -vvnn I see this 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) Subsystem: Sony Corporation Vaio VGN-FZ260E [104d:9005] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at fc400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel modules: snd-hda-intel When the audio is working alsamixer says it is using Sigmatel STAC9872AK. Anything else I can do to help? -- 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

4 8

[alsa-devel] getting HDAudio IDs
by Scott Serr 15 Apr '09

15 Apr '09

Hi All, I've found a great way to get HDAudio vendor/device IDs when audio drivers are loaded: /proc/asound/card?/codec? But to remake my particular (barebones) distro with with audio support, it will be repetitive and time consuming. I'm wondering what would be involved in making a user space utility that would return the HDAudio ID? A guy on #alsa came up with some good ideas. Potentially following the lead of others directly accessing the hardware from user space http://samba.org/picturebook/ but using libpci instead. Maybe we can use "single cmd mode" so we don't need interrupts. My original idea: It might be that a smart developer could grab a bunch of alsa/hdaudio driver source and torture it into userland.... just to get these IDs. Should I quit now? Is it doable? Are there people here with the skills/experience to do this? BTW, in the kernel config: #CONFIG_SOUND is not set Thanks, Scott

2 1

[alsa-devel] [PATCH 1/2] hdspm: metering ioctl fixed
by Florian Faber 15 Apr '09

15 Apr '09

Metering didn't work before. I know of no application that used this ioctl before, so I assume it is safe to modify the structure. diff a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c 10a11,13 > * Modified 2009-04-13 for proper metering by Florian Faber > * <faber(a)faberman.de> > * 82c85,86 < "Remy Bruno <remy.bruno(a)trinnov.com>"); --- > "Remy Bruno <remy.bruno(a)trinnov.com>, " > "Florian Faber <faber(a)faberman.de>"); 136c140,151 < #define HDSPM_MADI_peakrmsbase 4096 /* 4096-8191 2x64x32Bit Meters */ --- > > #define HDSPM_MADI_INPUT_PEAK 4096 > #define HDSPM_MADI_PLAYBACK_PEAK 4352 > #define HDSPM_MADI_OUTPUT_PEAK 4608 > > #define HDSPM_MADI_INPUT_RMS_L 6144 > #define HDSPM_MADI_PLAYBACK_RMS_L 6400 > #define HDSPM_MADI_OUTPUT_RMS_L 6656 > > #define HDSPM_MADI_INPUT_RMS_H 7168 > #define HDSPM_MADI_PLAYBACK_RMS_H 7424 > #define HDSPM_MADI_OUTPUT_RMS_H 7680 707c722 < /* maby a madi input (which is taken if sel sync is madi) */ --- > /* mayby a madi input (which is taken if sel sync is madi) */ 4108a4124,4128 > static inline int copy_u32_le(void __user *dest, void __iomem *src) > { > u32 val = readl(src); > return copy_to_user(dest, &val, 4); > } 4111c4131 < unsigned int cmd, unsigned long arg) --- > unsigned int cmd, unsigned long __user arg) 4113,4136c4133,4172 < struct hdspm *hdspm = hw->private_data; < struct hdspm_mixer_ioctl mixer; < struct hdspm_config_info info; < struct hdspm_version hdspm_version; < struct hdspm_peak_rms_ioctl rms; < < switch (cmd) { < < case SNDRV_HDSPM_IOCTL_GET_PEAK_RMS: < if (copy_from_user(&rms, (void __user *)arg, sizeof(rms))) < return -EFAULT; < /* maybe there is a chance to memorymap in future < * so dont touch just copy < */ < if(copy_to_user_fromio((void __user *)rms.peak, < hdspm->iobase+HDSPM_MADI_peakrmsbase, < sizeof(struct hdspm_peak_rms)) != 0 ) < return -EFAULT; < < break; < < < case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO: < --- > void __user *argp = (void __user *)arg; > struct hdspm *hdspm = hw->private_data; > struct hdspm_mixer_ioctl mixer; > struct hdspm_config_info info; > struct hdspm_version hdspm_version; > struct hdspm_peak_rms levels; > long unsigned int s; > int i=0; > > switch (cmd) { > > case SNDRV_HDSPM_IOCTL_GET_PEAK_RMS: > for (i=0; i<64; i++) { > levels.input_peaks[i] = readl(hdspm->iobase +HDSPM_MADI_INPUT_PEAK+i*4); > levels.playback_peaks[i] = readl(hdspm->iobase +HDSPM_MADI_PLAYBACK_PEAK+i*4); > levels.output_peaks[i] = readl(hdspm->iobase +HDSPM_MADI_OUTPUT_PEAK+i*4); > levels.input_rms[i] = (readl(hdspm->iobase +HDSPM_MADI_INPUT_RMS_H+i*4) << 32) | readl(hdspm->iobase +HDSPM_MADI_INPUT_RMS_L+i*4); > levels.playback_rms[i] = (readl(hdspm->iobase +HDSPM_MADI_PLAYBACK_RMS_H+i*4) << 32) | readl(hdspm->iobase +HDSPM_MADI_PLAYBACK_RMS_L+i*4); > levels.output_rms[i] = (readl(hdspm->iobase +HDSPM_MADI_OUTPUT_RMS_H+i*4) << 32) | readl(hdspm->iobase +HDSPM_MADI_OUTPUT_RMS_L+i*4); > } > > if (hdspm->system_sample_rate>96000) { > levels.speed = qs; > } else if (hdspm->system_sample_rate>48000) { > levels.speed = ds; > } else { > levels.speed = ss; > } > levels.status2 = hdspm_read(hdspm, HDSPM_statusRegister2); > > s = copy_to_user(argp, &levels, sizeof(struct hdspm_peak_rms)); > if (0 != s) { > snd_printk(KERN_ERR "copy_to_user(.., .., %lu): %lu\n", sizeof(struct hdspm_peak_rms), s); > return -EFAULT; > } > break; > > > case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO: > Flo -- Machines can do the work, so people have time to think. public key 6C002249 x-hkp://wwwkeys.eu.pgp.net

2 1

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 15 Apr '09

15 Apr '09

The following changes since commit 34e51ce60a210094bd66cf0a75dd8512247618ca: Takashi Iwai (1): Merge branch 'for-2.6.30' of git://git.kernel.org/.../broonie/sound-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-2.6.31 Alexander Beregalov (1): ASoC: n810: replace BUG() with BUG_ON() Daniel Glöckner (3): ASoC: Add driver for s6000 I2S interface ASoC: s6105 IP camera machine specific ASoC code ASoC: correct s6000 I2S clock polarity Daniel Ribeiro (1): ASoC: pxa-ssp.c fix clock/frame invert Mark Brown (11): ASoC: Display return code when failing to add a DAPM kcontrol ASoC: Provide core support for symmetric sample rates ASoC: Add WM8988 CODEC driver Merge branch 's6000' into for-2.6.31 ASoC: WM9713 requires symmetric rates on the voice DAI ASoC: Factor out application of power for generic widgets ASoC: Support DAPM events for DACs and ADCs ASoC: Move the WM9713 voice DAC powerdown to a DAPM event ASoC: Add WM8960 CODEC driver Merge branch 'for-2.6.30' into for-2.6.31 Merge branch 'for-2.6.30' into for-2.6.31 Peter Ujfalusi (1): ASoC: tlv320aic23: add DSP_A format support include/sound/soc-dai.h | 1 + include/sound/soc-dapm.h | 10 + include/sound/soc.h | 6 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/tlv320aic23.c | 2 + sound/soc/codecs/wm8960.c | 969 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8960.h | 127 +++++ sound/soc/codecs/wm8988.c | 1097 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8988.h | 60 +++ sound/soc/codecs/wm9713.c | 40 +- sound/soc/omap/n810.c | 7 +- sound/soc/pxa/pxa-ssp.c | 11 +- sound/soc/s6000/Kconfig | 19 + sound/soc/s6000/Makefile | 11 + sound/soc/s6000/s6000-i2s.c | 629 +++++++++++++++++++++++ sound/soc/s6000/s6000-i2s.h | 25 + sound/soc/s6000/s6000-pcm.c | 497 ++++++++++++++++++ sound/soc/s6000/s6000-pcm.h | 35 ++ sound/soc/s6000/s6105-ipcam.c | 244 +++++++++ sound/soc/soc-core.c | 38 ++ sound/soc/soc-dapm.c | 131 +++--- 24 files changed, 3893 insertions(+), 80 deletions(-) create mode 100644 sound/soc/codecs/wm8960.c create mode 100644 sound/soc/codecs/wm8960.h create mode 100644 sound/soc/codecs/wm8988.c create mode 100644 sound/soc/codecs/wm8988.h create mode 100644 sound/soc/s6000/Kconfig create mode 100644 sound/soc/s6000/Makefile create mode 100644 sound/soc/s6000/s6000-i2s.c create mode 100644 sound/soc/s6000/s6000-i2s.h create mode 100644 sound/soc/s6000/s6000-pcm.c create mode 100644 sound/soc/s6000/s6000-pcm.h create mode 100644 sound/soc/s6000/s6105-ipcam.c

3 18

[alsa-devel] [PATCH 1/2] keywest: Convert to new-style i2c driver
by Jean Delvare 15 Apr '09

15 Apr '09

The legacy i2c binding model is going away soon, so convert the ppc keywest sound driver to the new model or it will break. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> --- sound/ppc/keywest.c | 81 +++++++++++++++++++++++++-------------------------- 1 file changed, 40 insertions(+), 41 deletions(-) --- linux-2.6.30-rc1.orig/sound/ppc/keywest.c 2009-04-10 15:22:18.000000000 +0200 +++ linux-2.6.30-rc1/sound/ppc/keywest.c 2009-04-10 16:41:36.000000000 +0200 @@ -33,26 +33,25 @@ static struct pmac_keywest *keywest_ctx; -static int keywest_attach_adapter(struct i2c_adapter *adapter); -static int keywest_detach_client(struct i2c_client *client); - -struct i2c_driver keywest_driver = { - .driver = { - .name = "PMac Keywest Audio", - }, - .attach_adapter = &keywest_attach_adapter, - .detach_client = &keywest_detach_client, -}; - - #ifndef i2c_device_name #define i2c_device_name(x) ((x)->name) #endif +static int keywest_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + i2c_set_clientdata(client, keywest_ctx); + return 0; +} + +/* + * This is kind of a hack, best would be to turn powermac to fixed i2c + * bus numbers and declare the sound device as part of platform + * initialization + */ static int keywest_attach_adapter(struct i2c_adapter *adapter) { - int err; - struct i2c_client *new_client; + struct i2c_board_info info; if (! keywest_ctx) return -EINVAL; @@ -60,46 +59,46 @@ static int keywest_attach_adapter(struct if (strncmp(i2c_device_name(adapter), "mac-io", 6)) return 0; /* ignored */ - new_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); - if (! new_client) - return -ENOMEM; - - new_client->addr = keywest_ctx->addr; - i2c_set_clientdata(new_client, keywest_ctx); - new_client->adapter = adapter; - new_client->driver = &keywest_driver; - new_client->flags = 0; - - strcpy(i2c_device_name(new_client), keywest_ctx->name); - keywest_ctx->client = new_client; + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "keywest", I2C_NAME_SIZE); + info.addr = keywest_ctx->addr; + keywest_ctx->client = i2c_new_device(adapter, &info); - /* Tell the i2c layer a new client has arrived */ - if (i2c_attach_client(new_client)) { - snd_printk(KERN_ERR "tumbler: cannot attach i2c client\n"); - err = -ENODEV; - goto __err; - } - + /* + * Let i2c-core delete that device on driver removal. + * This is safe because i2c-core holds the core_lock mutex for us. + */ + list_add_tail(&keywest_ctx->client->detected, + &keywest_ctx->client->driver->clients); return 0; - - __err: - kfree(new_client); - keywest_ctx->client = NULL; - return err; } -static int keywest_detach_client(struct i2c_client *client) +static int keywest_remove(struct i2c_client *client) { if (! keywest_ctx) return 0; if (client == keywest_ctx->client) keywest_ctx->client = NULL; - i2c_detach_client(client); - kfree(client); return 0; } + +static const struct i2c_device_id keywest_i2c_id[] = { + { "keywest", 0 }, + { } +}; + +struct i2c_driver keywest_driver = { + .driver = { + .name = "PMac Keywest Audio", + }, + .attach_adapter = keywest_attach_adapter, + .probe = keywest_probe, + .remove = keywest_remove, + .id_table = keywest_i2c_id, +}; + /* exported */ void snd_pmac_keywest_cleanup(struct pmac_keywest *i2c) { -- Jean Delvare

2 4

[alsa-devel] Rewrite of hw_ptr updaters makes things worse?
by Chuck Ebbert 14 Apr '09

14 Apr '09

After adding these patches from 2.6.30 to a 2.6.29 kernel, pulseaudio problems are worse: ALSA: pcm - Reset invalid position even without debug option ALSA: pcm - Fix delta calculation at boundary overlap ALSA: pcm - Safer boundary checks ALSA: Rewrite hw_ptr updaters Mar 28 14:28:20 platon pulseaudio[3040]: alsa-sink.c: Increasing wakeup watermark to 60,00 ms Mar 28 14:31:35 platon pulseaudio[3040]: alsa-sink.c: Increasing wakeup watermark to 70,00 ms Mar 28 14:31:35 platon pulseaudio[3040]: alsa-sink.c: Increasing wakeup watermark to 80,00 ms Mar 28 14:31:35 platon pulseaudio[3040]: alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 4294952932 bytes (24347805 ms). Mar 28 14:31:35 platon pulseaudio[3040]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 28 14:31:35 platon pulseaudio[3040]: rtpoll.c: Assertion 'usec <= ((pa_usec_t) 1000000ULL)*60ULL*60ULL' failed at pulsecore/rtpoll.c:548, function pa_rtpoll_set_timer_relative(). Aborting.

2 2

[alsa-devel] HP tx2100er headphone jack problem
by Alex Pyattaev 14 Apr '09

14 Apr '09

Hello! I have this laptop for a year now, and ever since i've been trying to fix the sound. Playback works out-of-the box, but as for specific features... Hope you can help a bit. So, to be exact Hardware: HP tx2100er laptop, AMD64, nforce2 chipset. Audio: see alsa-info output in attachement Soft: 2.6.29-gentoo, alsa-driver either kernel.org or realtek's alsa-driver-1.0.19-5.11. I've tried different model settings, eg: model=hp : playback - ok record - none (either internal and external bus) switching to headphone on jack insert - ok model=3stack: playback-ok record - extremely silent(but works) or noisy switching to headphone on jack insert - no model=toshiba (it is for another chip, however, it waqs reported to wor for tx2500) playback - ok record - ok switching to headphone on jack insert - no So, as you see, the codec seems to work, but for some reason it is never fully functional. I have tried to fix it, but I just do not know how to do it. If you need any additional info - just mail me. Thank you in advance! Alex Pyattaev I just don't know what to do next.

2 2

[alsa-devel] [PATCH 2/2] hdspm: modified include file
by Florian Faber 14 Apr '09

14 Apr '09

Ckeaned up meter structure and modified the PEAK_RMS ioctl to reflect the modifications. diff a/sound/include/sound/hdspm.h b/sound/include/sound/hdspm.h 28,36d27 < /* peam rms level structure like we get from hardware < < maybe in future we can memory map it so I just copy it < to user on ioctl call now an dont change anything < rms are made out of low and high values < where (long) ????_rms = (????_rms_l >> 8) + ((????_rms_h & 0xFFFFFF00)<<24) < (i asume so from the code) < */ < 37a29,35 > uint32_t input_peaks[64]; > uint32_t playback_peaks[64]; > uint32_t output_peaks[64]; > > uint64_t input_rms[64]; > uint64_t playback_rms[64]; > uint64_t output_rms[64]; 39,56c37,38 < unsigned int level_offset[1024]; < < unsigned int input_peak[64]; < unsigned int playback_peak[64]; < unsigned int output_peak[64]; < unsigned int xxx_peak[64]; /* not used */ < < unsigned int reserved[256]; /* not used */ < < unsigned int input_rms_l[64]; < unsigned int playback_rms_l[64]; < unsigned int output_rms_l[64]; < unsigned int xxx_rms_l[64]; /* not used */ < < unsigned int input_rms_h[64]; < unsigned int playback_rms_h[64]; < unsigned int output_rms_h[64]; < unsigned int xxx_rms_h[64]; /* not used */ --- > enum {ss, ds, qs} speed; > int status2; 59,61d40 < struct hdspm_peak_rms_ioctl { < struct hdspm_peak_rms *peak; < }; 63d41 < /* use indirect access due to the limit of ioctl bit size */ 65c43 < _IOR('H', 0x40, struct hdspm_peak_rms_ioctl) --- > _IOR('H', 0x40, struct hdspm_peak_rms) Flo -- Machines can do the work, so people have time to think. public key 6C002249 x-hkp://wwwkeys.eu.pgp.net

1 0

Re: [alsa-devel] Help required. Slax OS, no audio, multiple cards, hdmi
by James Gauld 14 Apr '09

14 Apr '09

Yes, I tried the following Slax module first of all: http://www.slax.org/modules.php?action=detail&id=1063 with no success. any other suggestions? i've enabled iec958 mixer control, but cannot adjst it's volume. the HDMI device does not have the "pvolume" capability. I don't know if that's of any significance, but I'm in dark, uncharted territory here. Lee Revell wrote: > On Mon, Apr 13, 2009 at 11:09 AM, James Gauld <james(a)thundermonkey.net> wrote: >> However, running slax (from usb) I cannot get any sound from either >> device, but I'm only interested in the HDMI card (card #1) which >> delivers output to the integrated speakers (as far as I can tell from >> windows). >> >> I've tried from console and various widgets in KDE with no success. > > Try installing the proprietary ATI video driver. Thanks to DRM the > video driver has veto power over whether you can output audio via > HDMI. > > Lee > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

1 0

Re: [alsa-devel] alsa-driver-unstable not updated since Mar 24th?
by Susan Cragin 14 Apr '09

14 Apr '09

>At Tue, 14 Apr 2009 09:52:42 -0400 (GMT-04:00), >Susan Cragin wrote: >> >> [1 <text/plain; UTF-8 (7bit)>] >> >> I know there were problems on the 25th or 26th, either with the >> >> drivers or between the drivers and changes made to the kernel. >> > >> >It's been broken during my vacation. >> >Now it should be fixed. Let me know if it's still broken. >> > >> > >> >thanks, >> > >> >Takashi >> >> I just downloaded and installed the unstable snapshot 14-April 11:48. Install worked on my Asus G1S running Ubuntu Jaunty with all updates (most recent kernel is only 2.6.28-11-Generic). >> >> But sound is not working correctly on my machine. Neither sound card works. >> Sound cards present: >> hda-intel / Realtek ALC660-VD >> hda-audio-generic / Creative CA0110-IBG (Creative X-fi) > >Is it a regression from older version? > >How about alsa-driver snapshot (stable one)? >It won't work with CA0110, though. > > >Takashi Today's stable snapshot did not compile for me. I have attached the terminal output. My current problems with unstable are indeed a regression. CA0110 (my Creative X-fi) used to work very well with alsa-driver-unstable, as did my hda-intel. I last installed unstable perhaps two weeks ago? and used it at that time to run Audacity and sound programs in wine. I was getting very good sound. Susan

1 0

Re: [alsa-devel] alsa-driver-unstable not updated since Mar 24th?
by Susan Cragin 14 Apr '09

14 Apr '09

>> I know there were problems on the 25th or 26th, either with the >> drivers or between the drivers and changes made to the kernel. > >It's been broken during my vacation. >Now it should be fixed. Let me know if it's still broken. > > >thanks, > >Takashi I just downloaded and installed the unstable snapshot 14-April 11:48. Install worked on my Asus G1S running Ubuntu Jaunty with all updates (most recent kernel is only 2.6.28-11-Generic). But sound is not working correctly on my machine. Neither sound card works. Sound cards present: hda-intel / Realtek ALC660-VD hda-audio-generic / Creative CA0110-IBG (Creative X-fi) Have attached lspci-vv. Anything else, just ask. Any testing, etc. Susan

2 1

Re: [alsa-devel] Sound through HDMI
by Francois Marti 14 Apr '09

14 Apr '09

Hi! I try to change mixer and mute/unmute channel. I upgrade ubuntu from 8.10 to 9.04. I download/compile/install last alsa driver. I add in my modprob.conf this lines : options snd-hda-intel model=3stack-dig alias snd-card-0 snd-hda-intel alias sound-slot-0 snd-hda-intel alias sound-slot-0 snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss that's all. What is the next step ? Regards, On Tue, Apr 14, 2009 at 3:42 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Tue, 14 Apr 2009 15:39:18 +0200, > Francois Marti wrote: >> >> Ho ! >> >> I find something : > > Please add alsa-devel ML to Cc so that other people can follow. > > Also, describe what actually you have tried and what didn't work > precisely. "ABC doesn't work" is too ambiguous to investigate. > > > thanks, > > Takashi >

1 0

[alsa-devel] Help for writting an ASoC driver
by Ernesto Torres 14 Apr '09

14 Apr '09

Hi, i'm new to alsa and i would like some advice on where to get information about writing an ASoC driver it would be for the beagle board, since is the board i have available. I have already made some progress initializing and loading info to the codec via I2C, but I think there are still many things I don't know. I still don't know the exact way for playing via pcm with aplay or the order of the callbacks. I have checked the web for info but not much about ASoC, just the documentation with the linux kernel. My driver actually tries to play with aplay, but gets stuck in : omap-pcm.c: omap_pcm_pointer It just won't send the frames. Thank you for your help. - Ernesto

2 4

[alsa-devel] [PATCH] snd-usb-caiaq: rename files to remove redundant information in file pathes
by Daniel Mack 14 Apr '09

14 Apr '09

Cleanup only, no functional change. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/Makefile | 4 ++-- sound/usb/caiaq/{caiaq-audio.c => audio.c} | 4 ++-- sound/usb/caiaq/{caiaq-audio.h => audio.h} | 0 sound/usb/caiaq/{caiaq-control.c => control.c} | 4 ++-- sound/usb/caiaq/{caiaq-control.h => control.h} | 0 sound/usb/caiaq/{caiaq-device.c => device.c} | 10 +++++----- sound/usb/caiaq/{caiaq-device.h => device.h} | 0 sound/usb/caiaq/{caiaq-input.c => input.c} | 5 +++-- sound/usb/caiaq/{caiaq-input.h => input.h} | 0 sound/usb/caiaq/{caiaq-midi.c => midi.c} | 4 ++-- sound/usb/caiaq/{caiaq-midi.h => midi.h} | 0 11 files changed, 16 insertions(+), 15 deletions(-) rename sound/usb/caiaq/{caiaq-audio.c => audio.c} (99%) rename sound/usb/caiaq/{caiaq-audio.h => audio.h} (100%) rename sound/usb/caiaq/{caiaq-control.c => control.c} (99%) rename sound/usb/caiaq/{caiaq-control.h => control.h} (100%) rename sound/usb/caiaq/{caiaq-device.c => device.c} (99%) rename sound/usb/caiaq/{caiaq-device.h => device.h} (100%) rename sound/usb/caiaq/{caiaq-input.c => input.c} (99%) rename sound/usb/caiaq/{caiaq-input.h => input.h} (100%) rename sound/usb/caiaq/{caiaq-midi.c => midi.c} (98%) rename sound/usb/caiaq/{caiaq-midi.h => midi.h} (100%) diff --git a/sound/usb/caiaq/Makefile b/sound/usb/caiaq/Makefile index 23dadd5..3889996 100644 --- a/sound/usb/caiaq/Makefile +++ b/sound/usb/caiaq/Makefile @@ -1,4 +1,4 @@ -snd-usb-caiaq-y := caiaq-device.o caiaq-audio.o caiaq-midi.o caiaq-control.o -snd-usb-caiaq-$(CONFIG_SND_USB_CAIAQ_INPUT) += caiaq-input.o +snd-usb-caiaq-y := device.o audio.o midi.o control.o +snd-usb-caiaq-$(CONFIG_SND_USB_CAIAQ_INPUT) += input.o obj-$(CONFIG_SND_USB_CAIAQ) += snd-usb-caiaq.o diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/audio.c similarity index 99% rename from sound/usb/caiaq/caiaq-audio.c rename to sound/usb/caiaq/audio.c index cf37331..3f45c0f 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/audio.c @@ -22,8 +22,8 @@ #include <sound/core.h> #include <sound/pcm.h> -#include "caiaq-device.h" -#include "caiaq-audio.h" +#include "device.h" +#include "audio.h" #define N_URBS 32 #define CLOCK_DRIFT_TOLERANCE 5 diff --git a/sound/usb/caiaq/caiaq-audio.h b/sound/usb/caiaq/audio.h similarity index 100% rename from sound/usb/caiaq/caiaq-audio.h rename to sound/usb/caiaq/audio.h diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/control.c similarity index 99% rename from sound/usb/caiaq/caiaq-control.c rename to sound/usb/caiaq/control.c index 52685cf..305f3c3 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/control.c @@ -23,8 +23,8 @@ #include <sound/core.h> #include <sound/pcm.h> -#include "caiaq-device.h" -#include "caiaq-control.h" +#include "device.h" +#include "control.h" #define CNT_INTVAL 0x10000 diff --git a/sound/usb/caiaq/caiaq-control.h b/sound/usb/caiaq/control.h similarity index 100% rename from sound/usb/caiaq/caiaq-control.h rename to sound/usb/caiaq/control.h diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/device.c similarity index 99% rename from sound/usb/caiaq/caiaq-device.c rename to sound/usb/caiaq/device.c index 89f8b68..6d51770 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/device.c @@ -28,11 +28,11 @@ #include <sound/core.h> #include <sound/pcm.h> -#include "caiaq-device.h" -#include "caiaq-audio.h" -#include "caiaq-midi.h" -#include "caiaq-control.h" -#include "caiaq-input.h" +#include "device.h" +#include "audio.h" +#include "midi.h" +#include "control.h" +#include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13"); diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/device.h similarity index 100% rename from sound/usb/caiaq/caiaq-device.h rename to sound/usb/caiaq/device.h diff --git a/sound/usb/caiaq/caiaq-input.c b/sound/usb/caiaq/input.c similarity index 99% rename from sound/usb/caiaq/caiaq-input.c rename to sound/usb/caiaq/input.c index 4451775..a48d309 100644 --- a/sound/usb/caiaq/caiaq-input.c +++ b/sound/usb/caiaq/input.c @@ -20,8 +20,9 @@ #include <linux/usb.h> #include <linux/usb/input.h> #include <sound/pcm.h> -#include "caiaq-device.h" -#include "caiaq-input.h" + +#include "device.h" +#include "input.h" static unsigned short keycode_ak1[] = { KEY_C, KEY_B, KEY_A }; static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4, diff --git a/sound/usb/caiaq/caiaq-input.h b/sound/usb/caiaq/input.h similarity index 100% rename from sound/usb/caiaq/caiaq-input.h rename to sound/usb/caiaq/input.h diff --git a/sound/usb/caiaq/caiaq-midi.c b/sound/usb/caiaq/midi.c similarity index 98% rename from sound/usb/caiaq/caiaq-midi.c rename to sound/usb/caiaq/midi.c index 79424c1..8fa8cd8 100644 --- a/sound/usb/caiaq/caiaq-midi.c +++ b/sound/usb/caiaq/midi.c @@ -21,8 +21,8 @@ #include <sound/core.h> #include <sound/pcm.h> -#include "caiaq-device.h" -#include "caiaq-midi.h" +#include "device.h" +#include "midi.h" static int snd_usb_caiaq_midi_input_open(struct snd_rawmidi_substream *substream) { diff --git a/sound/usb/caiaq/caiaq-midi.h b/sound/usb/caiaq/midi.h similarity index 100% rename from sound/usb/caiaq/caiaq-midi.h rename to sound/usb/caiaq/midi.h -- 1.6.2.1

2 1

[alsa-devel] [PATCH] sc6000: add support for SC-6600 and SC-7000
by Krzysztof Helt 14 Apr '09

14 Apr '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add support for cards based on CompuMedia ASC-9408 chipsets. These cards were produced by Gallant. This patch makes the OSS aedsp16 driver redundant. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/Kconfig | 7 ++- sound/isa/sc6000.c | 127 ++++++++++++++++++++++++++++++++++++++++------------ 2 files changed, 103 insertions(+), 31 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index ed798bf..f9cae35 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -188,15 +188,18 @@ config SND_ES18XX will be called snd-es18xx. config SND_SC6000 - tristate "Gallant SC-6000, Audio Excel DSP 16" + tristate "Gallant SC-6000/6600/7000 and Audio Excel DSP 16" depends on HAS_IOPORT select SND_WSS_LIB select SND_OPL3_LIB select SND_MPU401_UART help - Say Y here to include support for Gallant SC-6000 card and clones: + Say Y here to include support for Gallant SC-6000, SC-6600, SC-7000 + cards and clones: Audio Excel DSP 16 and Zoltrix AV302. + These cards are based on CompuMedia ASC-9308 or ASC-9408 chips. + To compile this driver as a module, choose M here: the module will be called snd-sc6000. diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index bbc5369..37210c5 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -2,6 +2,8 @@ * Driver for Gallant SC-6000 soundcard. This card is also known as * Audio Excel DSP 16 or Zoltrix AV302. * These cards use CompuMedia ASC-9308 chip + AD1848 codec. + * SC-6600 and SC-7000 cards are also supported. They are based on + * CompuMedia ASC-9408 chip and CS4231 codec. * * Copyright (C) 2007 Krzysztof Helt <krzysztof.h1(a)wp.pl> * @@ -191,7 +193,7 @@ static __devinit unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq) return val; } -static __devinit int sc6000_wait_data(char __iomem *vport) +static int sc6000_wait_data(char __iomem *vport) { int loop = 1000; unsigned char val = 0; @@ -206,7 +208,7 @@ static __devinit int sc6000_wait_data(char __iomem *vport) return -EAGAIN; } -static __devinit int sc6000_read(char __iomem *vport) +static int sc6000_read(char __iomem *vport) { if (sc6000_wait_data(vport)) return -EBUSY; @@ -215,7 +217,7 @@ static __devinit int sc6000_read(char __iomem *vport) } -static __devinit int sc6000_write(char __iomem *vport, int cmd) +static int sc6000_write(char __iomem *vport, int cmd) { unsigned char val; int loop = 500000; @@ -276,8 +278,33 @@ static int __devinit sc6000_dsp_reset(char __iomem *vport) } /* detection and initialization */ -static int __devinit sc6000_cfg_write(char __iomem *vport, - unsigned char softcfg) +static int __devinit sc6000_hw_cfg_write(char __iomem *vport, const int *cfg) +{ + if (sc6000_write(vport, COMMAND_6C) < 0) { + snd_printk(KERN_WARNING "CMD 0x%x: failed!\n", COMMAND_6C); + return -EIO; + } + if (sc6000_write(vport, COMMAND_5C) < 0) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", COMMAND_5C); + return -EIO; + } + if (sc6000_write(vport, cfg[0]) < 0) { + snd_printk(KERN_ERR "DATA 0x%x: failed!\n", cfg[0]); + return -EIO; + } + if (sc6000_write(vport, cfg[1]) < 0) { + snd_printk(KERN_ERR "DATA 0x%x: failed!\n", cfg[1]); + return -EIO; + } + if (sc6000_write(vport, COMMAND_C5) < 0) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", COMMAND_C5); + return -EIO; + } + + return 0; +} + +static int sc6000_cfg_write(char __iomem *vport, unsigned char softcfg) { if (sc6000_write(vport, WRITE_MDIRQ_CFG)) { @@ -291,7 +318,7 @@ static int __devinit sc6000_cfg_write(char __iomem *vport, return 0; } -static int __devinit sc6000_setup_board(char __iomem *vport, int config) +static int sc6000_setup_board(char __iomem *vport, int config) { int loop = 10; @@ -334,16 +361,38 @@ static int __devinit sc6000_init_mss(char __iomem *vport, int config, return 0; } -static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, - char __iomem *vmss_port, int mpu_irq) +static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, + long xport, long xmpu, + long xmss_port) +{ + cfg[0] = 0; + cfg[1] = 0; + if (xport == 0x240) + cfg[0] |= 1; + if (xmpu != SNDRV_AUTO_PORT) { + cfg[0] |= (xmpu & 0x30) >> 2; + cfg[1] |= 0x20; + } + if (xmss_port == 0xe80) + cfg[0] |= 0x10; + cfg[0] |= 0x40; /* always set */ + cfg[1] |= 0x80; /* enable WSS system */ + cfg[1] &= ~0x40; /* disable IDE */ + snd_printd("hw cfg %x, %x\n", cfg[0], cfg[1]); +} + +static int __devinit sc6000_init_board(char __iomem *vport, + char __iomem *vmss_port, int dev) { char answer[15]; char version[2]; - int mss_config = sc6000_irq_to_softcfg(irq) | - sc6000_dma_to_softcfg(dma); + int mss_config = sc6000_irq_to_softcfg(irq[dev]) | + sc6000_dma_to_softcfg(dma[dev]); int config = mss_config | - sc6000_mpu_irq_to_softcfg(mpu_irq); + sc6000_mpu_irq_to_softcfg(mpu_irq[dev]); int err; + int cfg[2]; + int old = 0; err = sc6000_dsp_reset(vport); if (err < 0) { @@ -360,7 +409,6 @@ static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, /* * My SC-6000 card return "SC-6000" in DSPCopyright, so * if we have something different, we have to be warned. - * Mine returns "SC-6000A " - KH */ if (strncmp("SC-6000", answer, 7)) snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n"); @@ -372,13 +420,29 @@ static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, printk(KERN_INFO PFX "Detected model: %s, DSP version %d.%d\n", answer, version[0], version[1]); - /* - * 0x0A == (IRQ 7, DMA 1, MIRQ 0) - */ - err = sc6000_cfg_write(vport, 0x0a); + /* set configuration */ + sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], + mss_port[dev]); + if (sc6000_hw_cfg_write(vport, cfg) < 0) { + snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); + return -EIO; + } + err = sc6000_setup_board(vport, config); if (err < 0) { - snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); - return -EFAULT; + snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); + return -ENODEV; + } + + sc6000_dsp_reset(vport); + sc6000_write(vport, COMMAND_5C); + if (sc6000_read(vport) < 0) + old = 1; + sc6000_dsp_reset(vport); + + if (!old) { + sc6000_write(vport, COMMAND_60); + sc6000_write(vport, 0x02); + sc6000_dsp_reset(vport); } err = sc6000_setup_board(vport, config); @@ -386,10 +450,9 @@ static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); return -ENODEV; } - err = sc6000_init_mss(vport, config, vmss_port, mss_config); if (err < 0) { - snd_printk(KERN_ERR "Can not initialize " + snd_printk(KERN_ERR "Cannot initialize " "Microsoft Sound System mode.\n"); return -ENODEV; } @@ -485,14 +548,14 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) struct snd_card *card; struct snd_wss *chip; struct snd_opl3 *opl3; - char __iomem *vport; + char __iomem **vport; char __iomem *vmss_port; - - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(vport)); if (!card) return -ENOMEM; + vport = card->private_data; if (xirq == SNDRV_AUTO_IRQ) { xirq = snd_legacy_find_free_irq(possible_irqs); if (xirq < 0) { @@ -517,8 +580,8 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) err = -EBUSY; goto err_exit; } - vport = devm_ioport_map(devptr, port[dev], 0x10); - if (!vport) { + *vport = devm_ioport_map(devptr, port[dev], 0x10); + if (*vport == NULL) { snd_printk(KERN_ERR PFX "I/O port cannot be iomaped.\n"); err = -EBUSY; @@ -533,7 +596,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) goto err_unmap1; } vmss_port = devm_ioport_map(devptr, mss_port[dev], 4); - if (!vport) { + if (!vmss_port) { snd_printk(KERN_ERR PFX "MSS port I/O cannot be iomaped.\n"); err = -EBUSY; @@ -544,7 +607,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) port[dev], xirq, xdma, mpu_irq[dev] == SNDRV_AUTO_IRQ ? 0 : mpu_irq[dev]); - err = sc6000_init_board(vport, xirq, xdma, vmss_port, mpu_irq[dev]); + err = sc6000_init_board(*vport, vmss_port, dev); if (err < 0) goto err_unmap2; @@ -552,7 +615,6 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) WSS_HW_DETECT, 0, &chip); if (err < 0) goto err_unmap2; - card->private_data = chip; err = snd_wss_pcm(chip, 0, NULL); if (err < 0) { @@ -608,6 +670,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) return 0; err_unmap2: + sc6000_setup_board(*vport, 0); release_region(mss_port[dev], 4); err_unmap1: release_region(port[dev], 0x10); @@ -618,11 +681,17 @@ err_exit: static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev) { + struct snd_card *card = dev_get_drvdata(devptr); + char __iomem **vport = card->private_data; + + if (sc6000_setup_board(*vport, 0) < 0) + snd_printk(KERN_WARNING "sc6000_setup_board failed on exit!\n"); + release_region(port[dev], 0x10); release_region(mss_port[dev], 4); - snd_card_free(dev_get_drvdata(devptr)); dev_set_drvdata(devptr, NULL); + snd_card_free(card); return 0; } -- 1.6.0.3 ---------------------------------------------------------------------- Twoj znajomy gwiazda filmu XXX? Wkrec go! ;) Kliknij >>> http://link.interia.pl/f20f1

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[alsa-devel] iMac 9,1 sound patch
by Micah Bucy 14 Apr '09

14 Apr '09

I tried a git snapshot, but it wasn't working at all, so this patch was developed against the 1.0.19 stable release. This is my first time delving into alsa, so I'm not sure if everything is how it should be, including if the patch was correctly created. I can tell you this. I was able to get jack detection working correctly as long as the 6 channel option is set. I had tried modifying it to only allow a 6ch option as 2ch is for line-out (headphones) only, but the internal speakers wouldn't work, so I left it alone. My guess is, this same setup could be used by the macbook pro 5,1 I'll give it a try over the weekend. I have not tested the built-in mic or the line-in. -Micah

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[alsa-devel] Sound through HDMI
by Francois Marti 14 Apr '09

14 Apr '09

Hi ! I have no sound through HDMI. My platform : DELL Hybrid Station / Ubuntu 9.04 64bit (beta) with ACL888 chipset I give you informations collected by your script. regards

2 1

[alsa-devel] alsa-driver-unstable not updated since Mar 24th?
by Susan Cragin 14 Apr '09

14 Apr '09

Just checking. I know there were problems on the 25th or 26th, either with the drivers or between the drivers and changes made to the kernel. Susan

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[alsa-devel] [PATCH 0/2] ASoC: DSP mode corrections for omap-mcbsp
by Peter Ujfalusi 14 Apr '09

14 Apr '09

Hello, As a follow-up for the before easter mail thread on the omap DSP_A/_B mode support... This series should fix the current DSP_B mode and also introduces support for DSP_A mode in omap-mcbsp. Since omap's mcbsp module is quite flexible, one who is writing a board file for the omap platform has to check both the modes (DSP_A/_B) and the clock polarity settings against the codec requirements and pick the correct pair. For example in osk5912 board file: To use DSP_B mode (as it is now, no changes needed): snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBM_CFM); To use the DSP_A mode for the same board, codec: snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); --- Peter Ujfalusi (2): ASoC: omap-mcbsp: Correct the DSP_B mode ASoC: omap-mcbsp: Add DSP_A support sound/soc/omap/omap-mcbsp.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-)

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[alsa-devel] alsa driver for digigram lx6464es
by Tim Blechmann 14 Apr '09

14 Apr '09

hi all, i have implemented the prototype of an alsa driver for the digigram lx6464es audio interface. audio i/o is basically working, although not very well tested, yet. the driver is available from my git repository [1]. features: - i/o support for 64 channels via ethersound [2] missing features: - handling of xrun notification - configuration of the ethersound network needs to be done externally takashi, i would like to submit the patch to alsa soon ... how should i proceed? can you do a review of the code? thanks, tim [1] http://tim.klingt.org/git?p=digigram.git;a=shortlog;h=refs/heads/lx6464es [2] http://www.ethersound.com/ -- tim(a)klingt.org http://tim.klingt.org I don't write music for sissy ears. Charles Ives

2 11

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 14 Apr '09

14 Apr '09

The following changes since commit d6648da122870ff42bc720da25483bdb8bc868f8: Peter Ujfalusi (1): ASoC: TWL4030: Compillation error fix are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Eric's patch is a build fix for the magician after mainline merge. Eric Miao (1): ASoC: magician: remove un-necessary #include of pxa-regs.h and hardware.h Mark Brown (1): ASoC: Disable S3C64xx support in Kconfig sound/soc/pxa/magician.c | 2 -- sound/soc/s3c24xx/Kconfig | 6 +++--- 2 files changed, 3 insertions(+), 5 deletions(-)

2 1

[alsa-devel] Help required. Slax OS, no audio, multiple cards, hdmi
by James Gauld 14 Apr '09

14 Apr '09

Hi all, My first post and of course it's a help request, sorry ... I've got a fresh install of Slax 6.10 with the latest alsa packages as available on slackware. Here's the result of alsa-info.sh: http://www.alsa-project.org/db/?f=01356b864296bea4beb140b73cf41cbf84158997 My hardware: Samsung TV with integrated PC (model: LH32MGTLBC) 2 onboard audio devices, one of which is Radeon HD3200/RS780 This machine has windows xp embedded through which I can get audio output via the HDMI device. However, running slax (from usb) I cannot get any sound from either device, but I'm only interested in the HDMI card (card #1) which delivers output to the integrated speakers (as far as I can tell from windows). I've tried from console and various widgets in KDE with no success. I've searched all week with no solutions working. If anyone can help me out with this I'd be incredibly grateful Thanks James

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Re: [alsa-devel] Ubuntu sound problem - http://ubuntuforums.org/showthread.php?t=1073090
by jesse fillion 14 Apr '09

14 Apr '09

Hello, I wish I was coming back to you with good news. I have tried the suggested addition to my modprobe. I added the line you gave me to the alsa-base file in my modprobe folder. I have attached it and you will notice that I took "mobile" out and replaced it with m4. Both options allow me to use audio through my headphone jack, but notice that the audio buttons on my laptop funtion properly when I have m4 instead. 1. Ubuntu 8.10 2. kernel 2.6.27-14-generic 3. Alsa 1.0.19 4. AlsaUpgrade-1.0.x-rev-1.16 I notice that when I try and update Alsa snapshot, I get errors. See alsa upgrade log. I have also included my alsa-base file, uxchecker for addtional alsa related information and the actual alsa updater script that I used. Nothing is muted from what I can see, I can hear audio through my headphone jack perfectly. Any help I could receive is truly greatly appreciated. Thank-you Date: Thu, 9 Apr 2009 21:11:03 -0400 From: geisj(a)MessageNetSystems.com To: rob.krakora(a)MessageNetSystems.com CC: jessefillion24(a)hotmail.com Subject: Re: [alsa-devel] Ubuntu sound problem - http://ubuntuforums.org/showthread.php?t=1073090 We added the following to /etc/modprobe.conf: options snd-hda-intel enable_msi=1 model=mobile Jerry Robert Krakora wrote: Jesse: My boss and I just had the same issue with the same hardware (a couple of months ago). We were able to resolve the issue. However, I am not at work right now so I will have to get back with you tomorrow on our resolution. On Thu, Apr 9, 2009 at 12:19 PM, jesse fillion <jessefillion24(a)hotmail.com> wrote: Good afternoon, My name is Jesse. Forgive me if this is not the correct address to be sending this type of request to, but I am looking for support in getting the Alsa drivers to work correctly with HP DV4 laptops. The DV4 1228ca has a Azalia RS780 soundcard. I have the ability to hear music through the headphone jack, but not the internal speakers (please see ubuntu forum for additional information). I would truly appreciate any information you could provide. Thank you. -Jesse Hi, I'm relatively new to Ubuntu and would be thrilled for any assistance anyone can provide me with this problem. I have an HP DV4 1228ca. Just like everyone, I did have sound through my front audio jack and could listen to with headphones. Unfortunately after trying all of your helpful suggestions, I don't even have audio in my front jack anymore. If I go into my sound properties, it doesnt seem to recognize my sound card anymore. I beleive this has to do with the entry in the alsa base file. Here are my results: "[code]#> ./configure --enable-dynamic-minors #> make" winblows@ubuntu:~/Desktop/AlsaDriver/alsa-driver$ make make dep make[1]: Entering directory `/home/winblows/Desktop/AlsaDriver/alsa-driver' make[2]: Entering directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' copying file alsa-kernel/core/info.c /home/winblows/Desktop/AlsaDriver/alsa-driver/utils/patch-alsa: 24: patch: not found make[2]: *** [info.c] Error 1 make[2]: Leaving directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' make[1]: *** [dep] Error 1 make[1]: Leaving directory `/home/winblows/Desktop/AlsaDriver/alsa-driver' make: *** [include/sndversions.h] Error 2 #> sudo make install-modules This is where my noobishness may shine through. I wasn't quite sure what to take from the make process above so I continued... winblows@ubuntu:~/Desktop/AlsaDriver/alsa-driver$ sudo make install-modules [sudo] password for winblows: find /lib/modules/2.6.27-11-generic/kernel/sound -name 'snd*.*o' | xargs rm -f find /lib/modules/2.6.27-11-generic/kernel/sound -name 'snd*.*o.gz' | xargs rm -f find /lib/modules/2.6.27-11-generic/kernel/sound -name 'ac97_bus.*o' | xargs rm -f find /lib/modules/2.6.27-11-generic/kernel/sound -name 'ac97_bus.*o.gz' | xargs rm -f make[1]: Entering directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' mkdir -p /lib/modules/2.6.27-11-generic/kernel/sound/acore cp snd-hrtimer.ko snd-hwdep.ko snd-page-alloc.ko snd-pcm.ko snd-rawmidi.ko snd-timer.ko snd.ko /lib/modules/2.6.27-11-generic/kernel/sound/acore cp: cannot stat `snd-hrtimer.ko': No such file or directory cp: cannot stat `snd-hwdep.ko': No such file or directory cp: cannot stat `snd-page-alloc.ko': No such file or directory cp: cannot stat `snd-pcm.ko': No such file or directory cp: cannot stat `snd-rawmidi.ko': No such file or directory cp: cannot stat `snd-timer.ko': No such file or directory cp: cannot stat `snd.ko': No such file or directory make[1]: *** [modules_install] Error 1 make[1]: Leaving directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' make: *** [install-modules] Error 1 #> sudo vim /etc/modprobe.d/alsa-base [add 'options snd_hda_intel model=hp-dv5' sans quotes to the bottom of the file, even if you have a dv4, etc.] I have tried each of the following, rebooting in between them: options snd_hda_intel model=hp-dv5 options snd_hda_intel model=hp-dv4 # Keep snd-pcsp from beeing loaded as first soundcard options snd-pcsp index=-2 alias snd-card-0 snd-hda-intel alias sound-slot-0 snd-hda-intel options snd-hda-intel model=dell-m4-1 options snd-hda-intel enable_msi=1 # Keep snd-pcsp from beeing loaded as first soundcard options snd-pcsp index=-2 alias snd-card-0 snd-hda-intel alias sound-slot-0 snd-hda-intel options snd-hda-intel model=hp=dv4 options snd-hda-intel enable_msi=1 I have also tried using the 3.4 compiler and "unstable" alsa snapshot package recommended by uticamarco. If anyone has any suggestions, I would be most grateful. Thank you very much. Jesse _________________________________________________________________ Internet Explorer 8 helps keep your personal info safe. http://go.microsoft.com/?linkid=9655581 _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Best Regards, -- Jerry Geis MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 (317)663-0808 Fax _________________________________________________________________ Create a cool, new character for your Windows Live™ Messenger. http://go.microsoft.com/?linkid=9656621

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[alsa-devel] [PATCH - patch_sigmatel.c for HP dv4-1222nr laptop 9/9Patch to patch_sigmatel.c for HP dv4-1222nr laptop
by renidragsemaj＠yahoo.com 14 Apr '09

14 Apr '09

From: James Gardiner <renidragsemaj(a)yahoo.com> This patch to patch_sigmatel.c is for the HP dv4-1222nr laptop. It activates the included microphone(s) and the mute status change light for this computer. I don't know if this will work for any other dv4 laptop. The commit isn't perfect, for it doesn't seem to work too well with pulseaudio - the mic boost doesn't work with that audioserver. Otherwise, it works well with esound, even across reboots and suspend/hibernate/resume. The built-in microphone sits on node 0x18 and feeds through node 0x1c. The gpio for the mute light is GPIO #0. I borrowed the code from the HP HDX. Occasionally if you press the mute button too fast in succession it lowers the actual volume to 0 and you'll have to adjust it again...But if you are doing that you probably are getting a kick out of making me feel bad about the behavior. :) No autodetect of the laptop seems to work for me, but "model=hp-dv4-1222nr" as an option to modprobe will make the changes. Signed-off-by: James Gardiner <renidragsemaj(a)yahoo.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61996a2..4286155 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_M4, STAC_HP_DV5, STAC_HP_HDX, + STAC_HP_DV4_1222NR, STAC_92HD71BXX_MODELS }; @@ -1847,6 +1848,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", [STAC_HP_HDX] = "hp-hdx", + [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -4534,6 +4536,28 @@ static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, } #endif +static int stac92xx_hp_dv4_1222nr_check_power_status(struct hda_codec *codec, + hda_nid_t nid) +{ + /* Shamelessly stolen from the function stac92xx_hp_hdx_check_power_status */ + /* by James Gardiner */ + struct sigmatel_spec *spec = codec->spec; + + if (nid == 0x10) { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE) + spec->gpio_data &= ~0x01; /* orange */ + else + spec->gpio_data |= 0x01; /* white */ + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, + spec->gpio_data); + } + + return 0; +} + static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { struct sigmatel_spec *spec = codec->spec; @@ -5202,6 +5226,19 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 1; break; + case STAC_HP_DV4_1222NR: /* James Gardiner */ + spec->num_dmics = 1; + spec->num_dmuxes = 2; + stac92xx_auto_set_pinctl(codec, 0x1c, 2); + stac92xx_auto_set_pinctl(codec, 0x18, AC_PINCTL_IN_EN); +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + spec->gpio_data |= 0x01; + codec->patch_ops.check_power_status = + stac92xx_hp_dv4_1222nr_check_power_status; +#endif + /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); -- 1.5.6.3

1 0

Re: [alsa-devel] no sound with the new imac
by Clemens Ladisch 13 Apr '09

13 Apr '09

Rafael J. Wysocki wrote: > Takashi, what's the appropriate list for ALSA-related things? alsa-devel (Takashi is on vacation until after the Easter weekend). > On Wednesday 08 April 2009, Justin Mattock wrote: > > Can somebody point me in the right direction > > to creating the correct pin numbers etc... > > for the new imac so I can have sound. What kernel version are you using? Are there any messages in the system log when you load the snd-hda-intel driver? What model do you mean with "the new imac"? What are the contents of /proc/asound/card0/codec#0? Best regards, Clemens

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[alsa-devel] Via vt8237 sometimes fails to load
by Manuel Naranjo 13 Apr '09

13 Apr '09

Hello guys, I have a KT880 based mobo made by MSI, I haven't used this computer for a while (a year or so) I remember that before the audio was working, even though sometimes it was initializated with noise and required me restarting alsa. Now it doesn't even load, and I get some weird errors in dmesg. I've checked the wiki and the bug tracker but still couldn't find a solution. This is lspci -vv -nn 00:11.5 Multimedia audio controller [0401]: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller [1106:3059] (rev 60) Subsystem: Micro-Star International Co., Ltd. Device [1462:0470] Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin C routed to IRQ 22 Region 0: I/O ports at a800 [size=256] Capabilities: <access denied> Kernel modules: snd-via82xx And dmesg ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:581: codec_read: codec 0 is not valid [0x1fc0000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:581: codec_read: codec 0 is not valid [0x1fe0000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:523: codec_ready: codec 0 is not ready [0x1000000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:523: codec_ready: codec 0 is not ready [0x1000000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:581: codec_read: codec 0 is not valid [0x1fc0000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/via82xx.c:581: codec_read: codec 0 is not valid [0x1fe0000] ALSA /tmp/alsa/alsa-driver-1.0.19/pci/ac97/ac97_codec.c:2077: AC'97 0 access is not valid [0xffffffff], removing mixer. VIA 82xx Audio 0000:00:11.5: PCI INT C disabled VIA 82xx Audio: probe of 0000:00:11.5 failed with error -5 Thanks, Manuel

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[alsa-devel] [PATCH] ASoC: pxa-ssp.c fix clock/frame invert
by Daniel Ribeiro 13 Apr '09

13 Apr '09

>From the pxa27x developers manual, Table 8-8 SSPSP Bit Definitions: SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low) SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low) SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High) SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High) SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0). This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and DSP_B modes. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657..6768e4e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -568,7 +568,10 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_NB_IF: break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(3); + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; break; default: return -EINVAL; @@ -585,7 +588,13 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_NB_NF: sspsp |= SSPSP_SFRMP; break; + case SND_SOC_DAIFMT_NB_IF: + break; case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; break; default: return -EINVAL; -- tg: (577c9c4..) asoc/pxa-ssp-invert-clock.patch (depends on: master) total: 0 errors, 0 warnings, 24 lines checked asoc-pxa-ssp-invert-clock.patch has no obvious style problems and is ready for submission.

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Re: [alsa-devel] Question about pxa2xx-ac97-lib.c
by Mark Brown 12 Apr '09

12 Apr '09

On Sun, Apr 12, 2009 at 04:47:39PM +0200, Marek Vasut wrote: > One more thing, I'd like to ask how to change the reset_gpio in the above > file. Will I have to make copy of pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI] in > palm27x.c and change the .probe pointer to my own function that sets the > proper platform data? That is, something like the following? Thanks BTW, I meant to say in reply to the previous mail - you really should copy all audio discussion to at least alsa-devel. > +/* Palms use GPIO95 for AC97 reset */ > +static int palm_ac97_probe(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct platform_device *pd = to_platform_device(dai->dev); > + static struct pxa2xx_ac97_platform_data pdata = { > + .reset_gpio = 95, > + }; > + printk("%s[%i]\n", __FUNCTION__, __LINE__); > + pd->dev.platform_data = &pdata; > + return pxa2xx_ac97_hw_probe(pd); > +} Look in mainline - pxa2xx-ac97-lib already provides platform data based configuration of the GPIO to use for reset.

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Re: [alsa-devel] Question about pxa2xx-ac97-lib.c
by Robert Jarzmik 12 Apr '09

12 Apr '09

Marek Vasut <marek.vasut(a)gmail.com> writes: > One more thing, I'd like to ask how to change the reset_gpio in the above > file. Will I have to make copy of pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI] in > palm27x.c and change the .probe pointer to my own function that sets the > proper platform data? That is, something like the following? Thanks I'll add Mark to the discussion here, for the ASoC point of view. Mark, the original mail is in [1]. I originaly thought we would change pxa_set_ac97_info(), to add a parameter which will be the platform data for the gpio. IOW, have something like : void __init pxa_set_ac97_info(struct pxa2xx_ac97_platform_data *pdata) { pxa_register_device(&pxa_device_ac97, ops); } The trouble is, I saw references to the platform data as pxa2xx_audio_ops_t*, mostly to suspend and resume a card. I've seen them in pxa2xx_ac97_do_suspend(), pxa2xx_ac97_do_resume(). Mark, could you please have a glance at it, and tell us the right approach to have the struct pxa2xx_ac97_platform_data passed to pxa2xx-ac97 please ? Cheers. -- Robert [1] : > diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c > index 48a73f6..f134f95 100644 > --- a/sound/soc/pxa/palm27x.c > +++ b/sound/soc/pxa/palm27x.c > @@ -19,11 +19,13 @@ > #include <linux/gpio.h> > #include <linux/interrupt.h> > #include <linux/irq.h> > +#include <linux/platform_device.h> > > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/soc.h> > #include <sound/soc-dapm.h> > +#include <sound/pxa2xx-lib.h> > > #include <asm/mach-types.h> > #include <mach/audio.h> > @@ -168,6 +170,21 @@ static int palm27x_ac97_init(struct snd_soc_codec *codec) > return 0; > } > // this is the altered probe function > +/* Palms use GPIO95 for AC97 reset */ > +static int palm_ac97_probe(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct platform_device *pd = to_platform_device(dai->dev); > + static struct pxa2xx_ac97_platform_data pdata = { > + .reset_gpio = 95, > + }; > + printk("%s[%i]\n", __FUNCTION__, __LINE__); > + pd->dev.platform_data = &pdata; > + return pxa2xx_ac97_hw_probe(pd); > +} > + > +static struct snd_soc_dai hifi_cpu_dai; > + > static struct snd_soc_dai_link palm27x_dai[] = { > { > .name = "AC97 HiFi", > @@ -204,6 +221,8 @@ static int __init palm27x_asoc_init(void) > { > int ret; > // here it'll be copied > + pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI].probe = palm_ac97_probe; > + > if (!(machine_is_palmtx() || machine_is_palmt5() || > machine_is_palmld())) > return -ENODEV; > > > ------------------------------------------------------------------- > List admin: http://lists.arm.linux.org.uk/mailman/listinfo/linux-arm-kernel > FAQ: http://www.arm.linux.org.uk/mailinglists/faq.php > Etiquette: http://www.arm.linux.org.uk/mailinglists/etiquette.php

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[alsa-devel] [PATCH] ASoC: n810: replace BUG() with BUG_ON()
by Alexander Beregalov 12 Apr '09

12 Apr '09

Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/soc/omap/n810.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index a6d1178..e54e1c2 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -383,10 +383,9 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) - BUG(); - if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) - BUG(); + BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) || + (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);

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Re: [alsa-devel] Alsa driver compatible with 2.6.18 kernel
by reetesh chandran 11 Apr '09

11 Apr '09

Thanks a lot Lee. Which Alsa driver version is the latest stable version, or which is most widely used ? --- On Fri, 4/10/09, Lee Revell <rlrevell(a)joe-job.com> wrote: From: Lee Revell <rlrevell(a)joe-job.com> Subject: Re: [alsa-devel] Alsa driver compatible with 2.6.18 kernel To: "reetesh chandran" <reeteshchandran(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Date: Friday, April 10, 2009, 5:45 PM On Fri, Apr 10, 2009 at 12:54 PM, reetesh chandran <reeteshchandran(a)yahoo.com> wrote: > Hello all, > > I have a 2.6.18 kernel and I am using 1.0.12rc1 alsa driver. I am facing some issues and would like to get a newer alsa driver. I cant change the linux kernel version. Do you know which is the latest alsa driver which is supported on linux 2.6.18 ? How do I find that out ? The latest should work. New ALSA releases are backwards compatible with old kernels. Lee

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[alsa-devel] DELL SP2309W Monitor with Integrated microphone
by Phil Gorbett 11 Apr '09

11 Apr '09

Any luck anyone with this one yet? I too have the same issue as that posted previously (last month), and get the same results with the indicative testing. I have tried several fixes via googling - module loading and sound "card" indexing via alsa.base modprobe file and LOTS of others. Still no luck. I am running FC10 with pulse, but think it's probably a snd-usb-audio problem as the driver appears to be correctly loading and the webcam is clearly indicated in the sound prefs app and gconf/gstreamer. I have run thru the usb audio device classes, and all of the reported class values reported via lsusb -v seem to correctly correspond to possible values from the audio class parameters. I'm now stumped. I do get the following errors in the gnome sound recorder: "Could not negotiate format" followed by "Could not get/set settings from/on resource.". This jives with the error from the sound preference panel: "gconfaudiosrc ! audioconvert ! audioresample ! gconfaudiosink profile=chat: Could not get/set settings from/on resource." which does not reveal anything specific enough to track down the problem from searching the net. All other sound devices and services are performing correctly via pulseaudio. I have at one time removed p/a, but this did not correct the problem. I am running the webcam mic as device 0, but changing that to any other device number fails to solve the problem. Alsamixer shows the card "Monitor Webcam" mic, the chip "USB mixer", the recording levels at 74, the capture enabled, and the AGC and loudness parameters enabled, and although I have tried various combinations of these, none have worked. I am running version 1.0.19 of the alsa drivers, and compiled these directly with all everything enabled as well as using the rpm builds. Alsainfo has been uploaded to: http://www.alsa-project.org/db/?f=803829378929c58bcc86351c091719a12c79075e The video portion of the camera functions entirely correctly via uvc and v4l, but the audio listing from the video test app gucveiw does not list the webcam mic at all under the audio device list. The uvc dev group sent me here, naturally. Cheers, PG

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[alsa-devel] oss emul: powerbook5,3 microphone to mono
by Robert T. Drury 11 Apr '09

11 Apr '09

On machine powerbook5,3 the built-in microphone is attached to the right channel of the sound chip. In the alsa oss emulation in linux-2.6.23.9 (alsa-*-1.0.12) the plugin layer copies the left input channel to a one-channel buffer when calling ioctl SNDCTL_DSP_CHANNELS with one channel. So no mic to mono on the powerbook5,3. In the kernel source: linux-2.6.23.9/sound/core/oss/pcm_plugin.c: The function snd_pcm_plug_read_transfer copies from the two channel hardware input buffer to the one channel user space buffer. The left source channel appears to be selected for copy in function snd_pcm_plug_format_plugins. One approach is to query line/mic input there for powerbook5,3 and select right channel source for copy. Any ideas?

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[alsa-devel] Alsa driver compatible with 2.6.18 kernel
by reetesh chandran 11 Apr '09

11 Apr '09

Hello all, I have a 2.6.18 kernel and I am using 1.0.12rc1 alsa driver. I am facing some issues and would like to get a newer alsa driver. I cant change the linux kernel version. Do you know which is the latest alsa driver which is supported on linux 2.6.18 ? How do I find that out ? Thanks Reetesh

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[alsa-devel] [PATCH 2/2] keywest: Get rid of useless i2c_device_name macro
by Jean Delvare 10 Apr '09

10 Apr '09

The i2c_device_name macro doesn't have much value, get rid of it. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> --- sound/ppc/keywest.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) --- linux-2.6.30-rc1.orig/sound/ppc/keywest.c 2009-04-10 16:22:08.000000000 +0200 +++ linux-2.6.30-rc1/sound/ppc/keywest.c 2009-04-10 16:25:16.000000000 +0200 @@ -33,10 +33,6 @@ static struct pmac_keywest *keywest_ctx; -#ifndef i2c_device_name -#define i2c_device_name(x) ((x)->name) -#endif - static int keywest_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -56,7 +52,7 @@ static int keywest_attach_adapter(struct if (! keywest_ctx) return -EINVAL; - if (strncmp(i2c_device_name(adapter), "mac-io", 6)) + if (strncmp(adapter->name, "mac-io", 6)) return 0; /* ignored */ memset(&info, 0, sizeof(struct i2c_board_info)); -- Jean Delvare

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[alsa-devel] Ubuntu sound problem - http://ubuntuforums.org/showthread.php?t=1073090
by jesse fillion 09 Apr '09

09 Apr '09

Good afternoon, My name is Jesse. Forgive me if this is not the correct address to be sending this type of request to, but I am looking for support in getting the Alsa drivers to work correctly with HP DV4 laptops. The DV4 1228ca has a Azalia RS780 soundcard. I have the ability to hear music through the headphone jack, but not the internal speakers (please see ubuntu forum for additional information). I would truly appreciate any information you could provide. Thank you. -Jesse Hi, I'm relatively new to Ubuntu and would be thrilled for any assistance anyone can provide me with this problem. I have an HP DV4 1228ca. Just like everyone, I did have sound through my front audio jack and could listen to with headphones. Unfortunately after trying all of your helpful suggestions, I don't even have audio in my front jack anymore. If I go into my sound properties, it doesnt seem to recognize my sound card anymore. I beleive this has to do with the entry in the alsa base file. Here are my results: "[code]#> ./configure --enable-dynamic-minors #> make" winblows@ubuntu:~/Desktop/AlsaDriver/alsa-driver$ make make dep make[1]: Entering directory `/home/winblows/Desktop/AlsaDriver/alsa-driver' make[2]: Entering directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' copying file alsa-kernel/core/info.c /home/winblows/Desktop/AlsaDriver/alsa-driver/utils/patch-alsa: 24: patch: not found make[2]: *** [info.c] Error 1 make[2]: Leaving directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' make[1]: *** [dep] Error 1 make[1]: Leaving directory `/home/winblows/Desktop/AlsaDriver/alsa-driver' make: *** [include/sndversions.h] Error 2 #> sudo make install-modules This is where my noobishness may shine through. I wasn't quite sure what to take from the make process above so I continued... winblows@ubuntu:~/Desktop/AlsaDriver/alsa-driver$ sudo make install-modules [sudo] password for winblows: find /lib/modules/2.6.27-11-generic/kernel/sound -name 'snd*.*o' | xargs rm -f find /lib/modules/2.6.27-11-generic/kernel/sound -name 'snd*.*o.gz' | xargs rm -f find /lib/modules/2.6.27-11-generic/kernel/sound -name 'ac97_bus.*o' | xargs rm -f find /lib/modules/2.6.27-11-generic/kernel/sound -name 'ac97_bus.*o.gz' | xargs rm -f make[1]: Entering directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' mkdir -p /lib/modules/2.6.27-11-generic/kernel/sound/acore cp snd-hrtimer.ko snd-hwdep.ko snd-page-alloc.ko snd-pcm.ko snd-rawmidi.ko snd-timer.ko snd.ko /lib/modules/2.6.27-11-generic/kernel/sound/acore cp: cannot stat `snd-hrtimer.ko': No such file or directory cp: cannot stat `snd-hwdep.ko': No such file or directory cp: cannot stat `snd-page-alloc.ko': No such file or directory cp: cannot stat `snd-pcm.ko': No such file or directory cp: cannot stat `snd-rawmidi.ko': No such file or directory cp: cannot stat `snd-timer.ko': No such file or directory cp: cannot stat `snd.ko': No such file or directory make[1]: *** [modules_install] Error 1 make[1]: Leaving directory `/home/winblows/Desktop/AlsaDriver/alsa-driver/acore' make: *** [install-modules] Error 1 #> sudo vim /etc/modprobe.d/alsa-base [add 'options snd_hda_intel model=hp-dv5' sans quotes to the bottom of the file, even if you have a dv4, etc.] I have tried each of the following, rebooting in between them: options snd_hda_intel model=hp-dv5 options snd_hda_intel model=hp-dv4 # Keep snd-pcsp from beeing loaded as first soundcard options snd-pcsp index=-2 alias snd-card-0 snd-hda-intel alias sound-slot-0 snd-hda-intel options snd-hda-intel model=dell-m4-1 options snd-hda-intel enable_msi=1 # Keep snd-pcsp from beeing loaded as first soundcard options snd-pcsp index=-2 alias snd-card-0 snd-hda-intel alias sound-slot-0 snd-hda-intel options snd-hda-intel model=hp=dv4 options snd-hda-intel enable_msi=1 I have also tried using the 3.4 compiler and "unstable" alsa snapshot package recommended by uticamarco. If anyone has any suggestions, I would be most grateful. Thank you very much. Jesse _________________________________________________________________ Internet Explorer 8 helps keep your personal info safe. http://go.microsoft.com/?linkid=9655581

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[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Add support Voice DAI
by Joonyoung Shim 09 Apr '09

09 Apr '09

Hi, I added Voice DAI to support the PCM voice interface of the twl4030 codec. I tried this work with TRM, but there can be some miscomprehension. Please review. Thanks.

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[alsa-devel] [PATCH 0/3] ASoC: omap-mcbsp, osk5912 and tlv320aic23 DSP mode change
by Peter Ujfalusi 09 Apr '09

09 Apr '09

Hello, While I was trying to implement the four channel mode for the twl4030 codec, I have noticed, that the DSP_B mode implementation in omap-mcbsp and tlv320aic23 is not coorect, also for me it seams that the osk5912 configuration for the DSP mode is incorrect (based on the tlv320aic32 datasheet). The following series tries to correct this by: - Adding DSP_A support for the codec (setting the LRP bit for the DSP mode) - Changing the DSP_B mode to DSP_A mode in omap-mcbsp - Correcting the dai format configuration for the osk5912 board file. Please note, that I don't have access to osk5912 board, but based on the available documentation, I think the changes in this series are correct. --- Peter Ujfalusi (3): ASoC: tlv320aic32: add DSP_A format support ASoC: omap-mcbsp: change the DSP_B mode to DSP_A mode ASoC: osk5912: Use the correct DSP_A mode sound/soc/codecs/tlv320aic23.c | 2 ++ sound/soc/omap/omap-mcbsp.c | 4 ++-- sound/soc/omap/osk5912.c | 8 ++++---- 3 files changed, 8 insertions(+), 6 deletions(-)

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[alsa-devel] Alsa driver issue with udev events
by reeteshchandran＠yahoo.com 09 Apr '09

09 Apr '09

Hello all, I am using Alsa driver 1.0.12RC1 on ARM platform. I have a volume control application using SALSA (small alsa) running. The app has the control device open (/dev/snd/controlC0). I have another application listening to udev events, something like udev monitor. The device in description is USB headset. The problem is when I unplug the device. When I unplug, I am not getting all udev remove events. I get only 3 remove events from /class/sound/mixer, /class/sound/dsp and /class/sound/audio. But I dont get the remove events for the USB device layer. As a result I am not able to detect that the device has been removed. This problem exists only if the app has the control device open. I closed the control device and I get all unplug events correctly. Has anyone else seen this problem ? Any pointers as to where to start looking ? Thanks and regards, Reetesh

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Re: [alsa-devel] ASoC: Fixed reg cache bugs in wm8974 codec.
by Mark Brown 08 Apr '09

08 Apr '09

On Wed, Apr 08, 2009 at 12:25:49PM +0200, javier Martin wrote: > ASoC: Fixed reg cache bugs in wm8974 codec. Applied, thanks.

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[alsa-devel] Help-Rewind/Forward
by Sudeept Prusti 08 Apr '09

08 Apr '09

Hi all, Can anyone brief how exactly rewind/forward works in alsa-lib API. As alsa uses period based architecture, how exactly it(rewind/forward) happens in driver. Thanks, Sudeept The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it.

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[alsa-devel] 08-Apr unstable snapsot does not compile with Ubuntu kernel released this morning
by Susan Cragin 08 Apr '09

08 Apr '09

Background--Yesterday was the first alsa-driver unstable in a while. It did not compile with my kernel and I notified this list. Yesterday afternoon late there were some updates including a new (I think) alsa-tools. (It might have been alsa-utils.) After that, the 08-Apr drivers compiled. There was another Ubuntu 2.6.28-11 kernel released this morning. After downloading /installing that the 08-Apr unstable snapshot no longer compiles. (I tried both ./configure && make AND ./gitcompile) I don't know if this information is useful to you, or even if alsa-driver-unstable is supposed to compile with the 28 kernel. No need to respond. Susan susan@ubuntu:~/Desktop/alsa-driver-unstable$ ./configure && make checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking for ranlib... ranlib checking for a BSD-compatible install... /usr/bin/install -c checking how to run the C preprocessor... gcc -E checking for grep that handles long lines and -e... /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for an ANSI C-conforming const... yes checking for inline... inline checking whether time.h and sys/time.h may both be included... yes checking whether gcc needs -traditional... no checking for current directory... /home/susan/Desktop/alsa-driver-unstable checking cross compile... checking for directory with kernel source... /lib/modules/2.6.28-11-generic/build checking for directory with kernel build... checking for kernel linux/version.h... yes checking for kernel linux/autoconf.h... yes checking for kernel version... 2.6.28-11-generic checking for GCC version... ./configure: eval: line 5216: syntax error near unexpected token `)' ./configure: eval: line 5216: `my_compiler_version=4.3.3-5ubuntu4)' Kernel compiler: Used compiler: gcc (Ubuntu 4.3.3-5ubuntu4) 4.3.3 *** NO PREDEFINED KERNEL COMPILER IS DETECTED *** Assuming the same compiler is used with the current system compiler. *** Please make sure that the same compiler version was used for building kernel. checking for built-in ALSA... no checking for existing ALSA module... yes checking for Red Hat kernel... auto checking for Red Hat kernel... no checking for SUSE kernel... auto checking for SUSE kernel... no checking for updating alsa-kernel version.h... yes checking for CONFIG_EXPERIMENTAL... yes checking for directory to store kernel modules... /lib/modules/2.6.28-11-generic/kernel/sound checking for verbose procfs... on checking for verbose printk... on checking for debug level... none checking for ISA support in kernel... yes checking for processor type... i586 checking for i386 machine type... default checking for ISA DMA API... yes checking for kernel linux/config.h... no Creating <linux/config.h>... checking for deprecated linux/config.h... checking to modify of kernel linux/kmod.h... no checking for kernel linux/utsrelease.h... yes checking for kernel linux/compiler.h... yes checking for kernel linux/pm.h... yes checking for kernel linux/spinlock.h... yes checking for kernel linux/irq.h... yes checking for kernel linux/threads.h... yes checking for kernel linux/rwsem.h... yes checking for kernel linux/gameport.h... yes checking for kernel media/v4l2-dev.h... yes checking for kernel linux/devfs_fs_kernel.h... no Creating a dummy <linux/devfs_fs_kernel.h>... checking for kernel linux/highmem.h... yes checking for kernel linux/workqueue.h... yes checking for kernel linux/dma-mapping.h... yes checking for kernel asm/hw_irq.h... yes checking for kernel linux/device.h... yes checking for kernel linux/platform_device.h... yes checking for kernel linux/isa.h... yes checking for kernel linux/jiffies.h... yes checking for kernel linux/compat.h... yes checking for kernel linux/log2.h... yes checking for kernel linux/adb.h... yes checking for kernel linux/cuda.h... yes checking for kernel linux/pmu.h... yes checking for kernel linux/io.h... yes checking for kernel linux/moduleparam.h... yes checking for kernel linux/syscalls.h... yes checking for kernel linux/firmware.h... yes checking for kernel linux/err.h... yes checking for kernel linux/bitmap.h... yes checking for kernel linux/mutex.h... yes checking for kernel linux/pm_qos_params.h... yes checking for kernel linux/kthread.h... yes checking for kernel asm/irq_regs.h... yes checking for kernel linux/seq_file.h... yes checking for kernel linux/debugfs.h... yes checking for kernel linux/gpio.h... yes checking for kernel module symbol versions... yes checking for has ioport support... yes checking for PCI support in kernel... yes checking for I2C driver in kernel... yes checking for I2C_POWERMAC in kernel... unknown checking for firmware loader... yes checking for input subsystem in kernel... yes checking for pc-speaker platform in kernel... yes checking for SMP... yes checking for Video device support in kernel... yes checking for ISA PnP driver in kernel... yes checking for PnP driver in kernel... yes checking for Kernel ISA-PnP support... yes checking for strlcpy... yes checking for snprintf... yes checking for vsnprintf... yes checking for scnprintf... yes checking for sscanf... yes checking for vmalloc_to_page... yes checking for old kmod... no checking for PDE... yes checking for pci_set_consistent_dma_mask... yes checking for pci_dev_present... yes checking for msleep... yes checking for msleep_interrupt... yes checking for msecs_to_jiffies... yes checking for tty->count is the atomic type... no checking for video_get_drvdata... yes checking for video_drvdata... yes checking for V4L1 layer... yes checking for io_remap_pfn_range... yes checking for kcalloc... yes checking for kstrdup... yes checking for kstrndup... yes checking for kzalloc... yes checking for create_workqueue with flags... no checking for pci_ioremap_bar... yes checking for saved_config_space in pci_dev... yes checking for new pci_save_state... yes checking for register_sound_special_device... yes checking for driver extra-version... checking for driver version... 1.0.19 checking for dynamic minor numbers... no checking for sequencer support... yes checking for OSS/Free emulation... yes checking for OSS PCM plugin system inclusion... yes checking for RTC support... no checking for HPET support... yes checking for Procfs support... yes checking for USB support... yes checking for class_simple... no checking for old driver suspend/resume callbacks... no checking for removal of page-reservation for nopage/mmap... no checking for nested class_device... no checking for new IRQ handler... yes checking for gfp_t... yes checking for GFP_DMA32... yes checking for page_to_pfn... yes checking for PnP suspend/resume... yes checking for device_create_drvdata... no checking for new unlocked/compat_ioctl... yes checking for builtin _Bool support... yes checking for x86-compatible PC... yes checking for High-Res timers... yes checking for kernel PCMCIA checking for PCMCIA support... yes checking for PC9800 support in kernel... no checking for parallel port support... yes checking for power management... yes checking for CONFIG_HAS_DMA... yes checking for cards to compile driver for... all checking for additonal options to compile driver for... all configure: creating ./config.status config.status: creating version config.status: creating Makefile.conf config.status: WARNING: 'Makefile.conf.in' seems to ignore the --datarootdir setting config.status: creating snddevices config.status: creating utils/alsa-driver.spec config.status: creating utils/buildrpm config.status: creating toplevel.config config.status: creating utils/alsasound config.status: creating utils/alsasound.posix config.status: creating include/pci_ids_compat.h config.status: creating include/i2c-id_compat.h config.status: creating include/config.h config.status: include/config.h is unchanged config.status: creating include/config1.h config.status: include/config1.h is unchanged config.status: creating include/version.h config.status: include/version.h is unchanged config.status: creating include/autoconf-extra.h config.status: include/autoconf-extra.h is unchanged Hacking autoconf.h... make dep make[1]: Entering directory `/home/susan/Desktop/alsa-driver-unstable' make[2]: Entering directory `/home/susan/Desktop/alsa-driver-unstable/acore' copying file alsa-kernel/core/info.c patching file info.c Hunk #2 FAILED at 155. Hunk #3 succeeded at 162 (offset -5 lines). Hunk #4 succeeded at 172 (offset -5 lines). Hunk #5 succeeded at 203 (offset -5 lines). Hunk #6 succeeded at 490 (offset -5 lines). Hunk #7 FAILED at 530. Hunk #8 FAILED at 994. Hunk #9 succeeded at 1018 (offset -10 lines). 3 out of 9 hunks FAILED -- saving rejects to file info.c.rej make[2]: *** [info.c] Error 1 make[2]: Leaving directory `/home/susan/Desktop/alsa-driver-unstable/acore' make[1]: *** [dep] Error 1 make[1]: Leaving directory `/home/susan/Desktop/alsa-driver-unstable' make: *** [include/sndversions.h] Error 2 susan@ubuntu:~/Desktop/alsa-driver-unstable$

1 0

[alsa-devel] Need help on porting ALSA to ARM application baseboard
by Diptopal Basu 08 Apr '09

08 Apr '09

Hi All , I am working on ARM 926EJ-S Application baseboard . The board has a LM4549 codec connected through an AACI prime cell . I am running Linux 2.6.26.5 on this board and the requirement now is to have an ogg vorbis player running on the board , I am planning to use ogg123 for the same purpose , but it requires ALSA driver on the board . Do you have any information as to how to port ALSA on this particular board . Regards Diptopal Basu

2 1

[alsa-devel] [PATCH 0/6] ASoC: 4 channel support for twl4030 and omap-mcbsp
by Peter Ujfalusi 08 Apr '09

08 Apr '09

Hello, This series adds support for four channel DSP_A or DSP_B on omap-mcbsp and four channel support for the twl4030 codec (in Option1). Constrains added to deal with the channel count in twl4030 codec driver. As an example, the board file for Beagle is modified to add support for stereo and TDM mode depending on the stream's channel count, so the stereo and the four channel mode are supported on Beagle. Please note that patch #2 modifies also the DSP_B mode to have single phase configuration, s since I think that is the correct for that mode also. --- Peter Ujfalusi (6): ASoC: OMAP mcbsp: Add DSP_A support ASoC: OMAP mcbsp: Add 4 channel support ASoC: TWL4030: Enable all audio paths ASoC: TWL4030: Add 4 channel TDM support ASoC: TWL4030: Add constraints for 4 channel mode ASoC: Beagle: Add support for 4 channel mode sound/soc/codecs/twl4030.c | 24 ++++++++++++++++++++---- sound/soc/omap/omap-mcbsp.c | 25 +++++++++++++++++++------ sound/soc/omap/omap3beagle.c | 26 ++++++++++++++++++-------- 3 files changed, 57 insertions(+), 18 deletions(-)

4 25

[alsa-devel] 07-Apr snapshot does not compile
by Susan Cragin 07 Apr '09

07 Apr '09

This is FYI only. I assume the problem is either (1) already known or (2) my fault. Susan susan@ubuntu:~/Desktop/alsa-driver-unstable$ ./configure && make checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking for ranlib... ranlib checking for a BSD-compatible install... /usr/bin/install -c checking how to run the C preprocessor... gcc -E checking for grep that handles long lines and -e... /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for an ANSI C-conforming const... yes checking for inline... inline checking whether time.h and sys/time.h may both be included... yes checking whether gcc needs -traditional... no checking for current directory... /home/susan/Desktop/alsa-driver-unstable checking cross compile... checking for directory with kernel source... /lib/modules/2.6.28-11-generic/build checking for directory with kernel build... checking for kernel linux/version.h... yes checking for kernel linux/autoconf.h... yes checking for kernel version... 2.6.28-11-generic checking for GCC version... ./configure: eval: line 5216: syntax error near unexpected token `)' ./configure: eval: line 5216: `my_compiler_version=4.3.3-5ubuntu4)' Kernel compiler: Used compiler: gcc (Ubuntu 4.3.3-5ubuntu4) 4.3.3 *** NO PREDEFINED KERNEL COMPILER IS DETECTED *** Assuming the same compiler is used with the current system compiler. *** Please make sure that the same compiler version was used for building kernel. checking for built-in ALSA... no checking for existing ALSA module... yes checking for Red Hat kernel... auto checking for Red Hat kernel... no checking for SUSE kernel... auto checking for SUSE kernel... no checking for updating alsa-kernel version.h... yes checking for CONFIG_EXPERIMENTAL... yes checking for directory to store kernel modules... /lib/modules/2.6.28-11-generic/kernel/sound checking for verbose procfs... on checking for verbose printk... on checking for debug level... none checking for ISA support in kernel... yes checking for processor type... i586 checking for i386 machine type... default checking for ISA DMA API... yes checking for kernel linux/config.h... no Creating <linux/config.h>... checking for deprecated linux/config.h... checking to modify of kernel linux/kmod.h... no checking for kernel linux/utsrelease.h... yes checking for kernel linux/compiler.h... yes checking for kernel linux/pm.h... yes checking for kernel linux/spinlock.h... yes checking for kernel linux/irq.h... yes checking for kernel linux/threads.h... yes checking for kernel linux/rwsem.h... yes checking for kernel linux/gameport.h... yes checking for kernel media/v4l2-dev.h... yes checking for kernel linux/devfs_fs_kernel.h... no Creating a dummy <linux/devfs_fs_kernel.h>... checking for kernel linux/highmem.h... yes checking for kernel linux/workqueue.h... yes checking for kernel linux/dma-mapping.h... yes checking for kernel asm/hw_irq.h... yes checking for kernel linux/device.h... yes checking for kernel linux/platform_device.h... yes checking for kernel linux/isa.h... yes checking for kernel linux/jiffies.h... yes checking for kernel linux/compat.h... yes checking for kernel linux/log2.h... yes checking for kernel linux/adb.h... yes checking for kernel linux/cuda.h... yes checking for kernel linux/pmu.h... yes checking for kernel linux/io.h... yes checking for kernel linux/moduleparam.h... yes checking for kernel linux/syscalls.h... yes checking for kernel linux/firmware.h... yes checking for kernel linux/err.h... yes checking for kernel linux/bitmap.h... yes checking for kernel linux/mutex.h... yes checking for kernel linux/pm_qos_params.h... yes checking for kernel linux/kthread.h... yes checking for kernel asm/irq_regs.h... yes checking for kernel linux/seq_file.h... yes checking for kernel linux/debugfs.h... yes checking for kernel linux/gpio.h... yes checking for kernel module symbol versions... yes checking for has ioport support... yes checking for PCI support in kernel... yes checking for I2C driver in kernel... yes checking for I2C_POWERMAC in kernel... unknown checking for firmware loader... yes checking for input subsystem in kernel... yes checking for pc-speaker platform in kernel... yes checking for SMP... yes checking for Video device support in kernel... yes checking for ISA PnP driver in kernel... yes checking for PnP driver in kernel... yes checking for Kernel ISA-PnP support... yes checking for strlcpy... yes checking for snprintf... yes checking for vsnprintf... yes checking for scnprintf... yes checking for sscanf... yes checking for vmalloc_to_page... yes checking for old kmod... no checking for PDE... yes checking for pci_set_consistent_dma_mask... yes checking for pci_dev_present... yes checking for msleep... yes checking for msleep_interrupt... yes checking for msecs_to_jiffies... yes checking for tty->count is the atomic type... no checking for video_get_drvdata... yes checking for video_drvdata... yes checking for V4L1 layer... yes checking for io_remap_pfn_range... yes checking for kcalloc... yes checking for kstrdup... yes checking for kstrndup... yes checking for kzalloc... yes checking for create_workqueue with flags... no checking for pci_ioremap_bar... yes checking for saved_config_space in pci_dev... yes checking for new pci_save_state... yes checking for register_sound_special_device... yes checking for driver extra-version... checking for driver version... 1.0.19 checking for dynamic minor numbers... no checking for sequencer support... yes checking for OSS/Free emulation... yes checking for OSS PCM plugin system inclusion... yes checking for RTC support... no checking for HPET support... yes checking for Procfs support... yes checking for USB support... yes checking for class_simple... no checking for old driver suspend/resume callbacks... no checking for removal of page-reservation for nopage/mmap... no checking for nested class_device... no checking for new IRQ handler... yes checking for gfp_t... yes checking for GFP_DMA32... yes checking for page_to_pfn... yes checking for PnP suspend/resume... yes checking for device_create_drvdata... no checking for new unlocked/compat_ioctl... yes checking for builtin _Bool support... yes checking for x86-compatible PC... yes checking for High-Res timers... yes checking for kernel PCMCIA checking for PCMCIA support... yes checking for PC9800 support in kernel... no checking for parallel port support... yes checking for power management... yes checking for CONFIG_HAS_DMA... yes checking for cards to compile driver for... all checking for additonal options to compile driver for... all configure: creating ./config.status config.status: creating version config.status: creating Makefile.conf config.status: WARNING: 'Makefile.conf.in' seems to ignore the --datarootdir setting config.status: creating snddevices config.status: creating utils/alsa-driver.spec config.status: creating utils/buildrpm config.status: creating toplevel.config config.status: creating utils/alsasound config.status: creating utils/alsasound.posix config.status: creating include/pci_ids_compat.h config.status: creating include/i2c-id_compat.h config.status: creating include/config.h config.status: include/config.h is unchanged config.status: creating include/config1.h config.status: include/config1.h is unchanged config.status: creating include/version.h config.status: include/version.h is unchanged config.status: creating include/autoconf-extra.h config.status: include/autoconf-extra.h is unchanged Hacking autoconf.h... make dep make[1]: Entering directory `/home/susan/Desktop/alsa-driver-unstable' make[2]: Entering directory `/home/susan/Desktop/alsa-driver-unstable/acore' copying file alsa-kernel/core/info.c patching file info.c Hunk #2 FAILED at 155. Hunk #3 succeeded at 162 (offset -5 lines). Hunk #4 succeeded at 172 (offset -5 lines). Hunk #5 succeeded at 203 (offset -5 lines). Hunk #6 succeeded at 490 (offset -5 lines). Hunk #7 FAILED at 530. Hunk #8 FAILED at 994. Hunk #9 succeeded at 1018 (offset -10 lines). 3 out of 9 hunks FAILED -- saving rejects to file info.c.rej make[2]: *** [info.c] Error 1 make[2]: Leaving directory `/home/susan/Desktop/alsa-driver-unstable/acore' make[1]: *** [dep] Error 1 make[1]: Leaving directory `/home/susan/Desktop/alsa-driver-unstable' make: *** [include/sndversions.h] Error 2

1 0

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Fix for compilation error
by Peter Ujfalusi 07 Apr '09

07 Apr '09

Hello, I'm really sorry for the trouble that this caused. The patch provides fix for the compillation error, which is introduced by the "ASoC: TWL4030: Add constrains for second stream" patch of mine... --- Peter Ujfalusi (1): ASoC: TWL4030: Compillation error fix sound/soc/codecs/twl4030.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-)

2 2

[alsa-devel] no sound with imac kernel=2.6.29
by Justin Mattock 07 Apr '09

07 Apr '09

Is there a patch that I can test out for the latest imac. At the moment I have no sound in either external/internal speakers. I've applied the imac91.patch submitted a few days ago(everything compiled fine)but didn't realize it for a different card(or so I think) below is lspci -vvv I can attach dmesg and supply info if need be. 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) Subsystem: nVidia Corporation Device cb79 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 17 Region 0: Memory at d3480000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel -- Justin P. Mattock

1 0

[alsa-devel] [PATCH][SOUND]volume control for CSWITCH and CROUTE
by Viral Mehta 07 Apr '09

07 Apr '09

From: Deepika Makhija <deepika.makhija(a)einfochips.com> Added an else part to check SNDRV_MIXER_OSS_PRESENT_CVOLUME for MIC (slot 7) in commit 36c7b833e5d2501142a371e4e75281d3a29fbd6b Similarly, checks and volume control is required for SNDRV_MIXER_OSS_PRESENT_CSWITCH and SNDRV_MIXER_OSS_PRESENT_CROUTE as well. Signed-off-by: Deepika Makhija <deepika.makhija(a)einfochips.com> Signed-off-by: Viral Mehta <viral.mehta(a)einfochips.com> --- --- a/sound/core/oss/mixer-oss.c 2009-04-04 17:06:46.000000000 +0530 +++ b/sound/core/oss/mixer-oss.c 2009-04-04 17:04:33.000000000 +0530 @@ -703,19 +703,27 @@ static int snd_mixer_oss_put_volume1(str if (left || right) { if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0); + if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0); if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0); if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1); + if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE) + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1); if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1); } else { if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) { + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE) { + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1); }

2 2

[alsa-devel] [PATCH] Add VREF powerdown sequence for another board
by Matthew Ranostay 07 Apr '09

07 Apr '09

Add powerdown sequence for VREF using a shared jack when the headphone is present and the microphone isn't on. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 16 ++++++++++++++++ 1 files changed, 16 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b5e108a..a6fc257 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4413,6 +4413,22 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) if (spec->num_pwrs > 0) stac92xx_pin_sense(codec, event->nid); stac92xx_report_jack(codec, event->nid); + + switch (codec->subsystem_id) { + case 0x103c308f: + if (event->nid == 0xb) { + int pin = AC_PINCTL_IN_EN; + + if (get_pin_presence(codec, 0xa) + && get_pin_presence(codec, 0xb)) + pin |= AC_PINCTL_VREF_80; + if (!get_pin_presence(codec, 0xb)) + pin |= AC_PINCTL_VREF_80; + + /* toggle VREF state based on mic + hp pin status */ + stac92xx_auto_set_pinctl(codec, 0x0a, pin); + } + } break; case STAC_VREF_EVENT: data = snd_hda_codec_read(codec, codec->afg, 0, -- 1.5.6.3

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 07 Apr '09

07 Apr '09

The following changes since commit 103f211d0be2bed75b5739de62a10415ef0bbc25: Peter Ujfalusi (1): ASoC: TWL4030: Add actual support for 96KHz playback support are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Anton Vorontsov (1): ASoC: fsl_dma: Pass the proper device for dma mapping routines Dan Carpenter (1): ASoC: Fix null dereference in ak4535_remove() sound/soc/codecs/ak4535.c | 3 ++- sound/soc/fsl/fsl_dma.c | 12 ++++++------ 2 files changed, 8 insertions(+), 7 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: fsl_dma: Pass the proper device for dma mapping routines
by Anton Vorontsov 07 Apr '09

07 Apr '09

The driver should pass a device that specifies internal DMA ops, but substream->pcm is just a logical device, and thus doesn't have arch- specific dma callbacks, therefore following bug appears: Freescale Synchronous Serial Interface (SSI) ASoC Driver ------------[ cut here ]------------ kernel BUG at arch/powerpc/include/asm/dma-mapping.h:237! Oops: Exception in kernel mode, sig: 5 [#1] ... NIP [c02259c4] snd_malloc_dev_pages+0x58/0xac LR [c0225c74] snd_dma_alloc_pages+0xf8/0x108 Call Trace: [df02bde0] [df02be2c] 0xdf02be2c (unreliable) [df02bdf0] [c0225c74] snd_dma_alloc_pages+0xf8/0x108 [df02be10] [c023a100] fsl_dma_new+0x68/0x124 [df02be20] [c02342ac] soc_new_pcm+0x1bc/0x234 [df02bea0] [c02343dc] snd_soc_new_pcms+0xb8/0x148 [df02bed0] [c023824c] cs4270_probe+0x34/0x124 [df02bef0] [c0232fe8] snd_soc_instantiate_card+0x1a4/0x2f4 [df02bf20] [c0233164] snd_soc_instantiate_cards+0x2c/0x68 [df02bf30] [c0234704] snd_soc_register_platform+0x60/0x80 [df02bf50] [c03d5664] fsl_soc_platform_init+0x18/0x28 ... This patch fixes the issue by using card's device instead. Signed-off-by: Anton Vorontsov <avorontsov(a)ru.mvista.com> --- sound/soc/fsl/fsl_dma.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index b3eb857..835e43c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -300,7 +300,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = fsl_dma_dmamask; - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); if (ret) { @@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, return -ENOMEM; } - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { @@ -418,7 +418,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) return -EBUSY; } - dma_private = dma_alloc_coherent(substream->pcm->dev, + dma_private = dma_alloc_coherent(substream->pcm->card->dev, sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL); if (!dma_private) { dev_err(substream->pcm->card->dev, @@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) dev_err(substream->pcm->card->dev, "can't register ISR for IRQ %u (ret=%i)\n", dma_private->irq, ret); - dma_free_coherent(substream->pcm->dev, + dma_free_coherent(substream->pcm->card->dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); return ret; @@ -761,13 +761,13 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) free_irq(dma_private->irq, dma_private); if (dma_private->ld_buf_phys) { - dma_unmap_single(substream->pcm->dev, + dma_unmap_single(substream->pcm->card->dev, dma_private->ld_buf_phys, sizeof(dma_private->link), DMA_TO_DEVICE); } /* Deallocate the fsl_dma_private structure */ - dma_free_coherent(substream->pcm->dev, + dma_free_coherent(substream->pcm->card->dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); substream->runtime->private_data = NULL; -- 1.5.6.5

5 7

[alsa-devel] [PATCH v2 1/2] alsa: ASoC driver for s6000 I2S interface
by Daniel Glöckner 06 Apr '09

06 Apr '09

This patch adds a driver for the I2S interface found on Stretch s6000 family processors. Changes compared to v1 of this patch: - removed superfluous dependency on SND_SOC - moved all function and register definitions from header to .c file - implemented remaining interaction between s6000-pcm and s6000-i2c with function pointers passed in cpu_dai->dma_data - removed SNDRV_PCM_RATE_KNOT - moved snd_soc_dai_ops outside s6000_i2s_dai - made s6000-i2s a platform driver that registers the dai - freeing s6000_i2s_dev _after_ last usage - replaced release_region with release_mem_region - handling dma underflow/overflow with SNDRV_PCM_STATE_XRUN - replaced DPRINTK with dev_dbg - made remaining not exported symbols static - added comments describing dma channel parameters Signed-off-by: Daniel Glöckner <dg(a)emlix.com> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/s6000/Kconfig | 10 + sound/soc/s6000/Makefile | 6 + sound/soc/s6000/s6000-i2s.c | 629 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/s6000/s6000-i2s.h | 25 ++ sound/soc/s6000/s6000-pcm.c | 497 ++++++++++++++++++++++++++++++++++ sound/soc/s6000/s6000-pcm.h | 35 +++ 8 files changed, 1204 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s6000/Kconfig create mode 100644 sound/soc/s6000/Makefile create mode 100644 sound/soc/s6000/s6000-i2s.c create mode 100644 sound/soc/s6000/s6000-i2s.h create mode 100644 sound/soc/s6000/s6000-pcm.c create mode 100644 sound/soc/s6000/s6000-pcm.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index ef025c6..56e5198 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -31,6 +31,7 @@ source "sound/soc/fsl/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" +source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 86a9b1f..6b33d80 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,4 +10,5 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ +obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig new file mode 100644 index 0000000..4bfc8bc --- /dev/null +++ b/sound/soc/s6000/Kconfig @@ -0,0 +1,10 @@ +config SND_S6000_SOC + tristate "SoC Audio for the Stretch s6000 family" + depends on XTENSA_VARIANT_S6000 + help + Say Y or M if you want to add support for codecs attached to + s6000 family chips. You will also need to select the platform + to support below. + +config SND_S6000_SOC_I2S + tristate diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile new file mode 100644 index 0000000..df15f87 --- /dev/null +++ b/sound/soc/s6000/Makefile @@ -0,0 +1,6 @@ +# s6000 Platform Support +snd-soc-s6000-objs := s6000-pcm.o +snd-soc-s6000-i2s-objs := s6000-i2s.o + +obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o +obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c new file mode 100644 index 0000000..dcc7904 --- /dev/null +++ b/sound/soc/s6000/s6000-i2s.c @@ -0,0 +1,629 @@ +/* + * ALSA SoC I2S Audio Layer for the Stretch S6000 family + * + * Author: Daniel Gloeckner, <dg(a)emlix.com> + * Copyright: (C) 2009 emlix GmbH <info(a)emlix.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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/interrupt.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "s6000-i2s.h" +#include "s6000-pcm.h" + +struct s6000_i2s_dev { + dma_addr_t sifbase; + u8 __iomem *scbbase; + unsigned int wide; + unsigned int channel_in; + unsigned int channel_out; + unsigned int lines_in; + unsigned int lines_out; + struct s6000_pcm_dma_params dma_params; +}; + +#define S6_I2S_INTERRUPT_STATUS 0x00 +#define S6_I2S_INT_OVERRUN 1 +#define S6_I2S_INT_UNDERRUN 2 +#define S6_I2S_INT_ALIGNMENT 4 +#define S6_I2S_INTERRUPT_ENABLE 0x04 +#define S6_I2S_INTERRUPT_RAW 0x08 +#define S6_I2S_INTERRUPT_CLEAR 0x0C +#define S6_I2S_INTERRUPT_SET 0x10 +#define S6_I2S_MODE 0x20 +#define S6_I2S_DUAL 0 +#define S6_I2S_WIDE 1 +#define S6_I2S_TX_DEFAULT 0x24 +#define S6_I2S_DATA_CFG(c) (0x40 + 0x10 * (c)) +#define S6_I2S_IN 0 +#define S6_I2S_OUT 1 +#define S6_I2S_UNUSED 2 +#define S6_I2S_INTERFACE_CFG(c) (0x44 + 0x10 * (c)) +#define S6_I2S_DIV_MASK 0x001fff +#define S6_I2S_16BIT 0x000000 +#define S6_I2S_20BIT 0x002000 +#define S6_I2S_24BIT 0x004000 +#define S6_I2S_32BIT 0x006000 +#define S6_I2S_BITS_MASK 0x006000 +#define S6_I2S_MEM_16BIT 0x000000 +#define S6_I2S_MEM_32BIT 0x008000 +#define S6_I2S_MEM_MASK 0x008000 +#define S6_I2S_CHANNELS_SHIFT 16 +#define S6_I2S_CHANNELS_MASK 0x030000 +#define S6_I2S_SCK_IN 0x000000 +#define S6_I2S_SCK_OUT 0x040000 +#define S6_I2S_SCK_DIR 0x040000 +#define S6_I2S_WS_IN 0x000000 +#define S6_I2S_WS_OUT 0x080000 +#define S6_I2S_WS_DIR 0x080000 +#define S6_I2S_LEFT_FIRST 0x000000 +#define S6_I2S_RIGHT_FIRST 0x100000 +#define S6_I2S_FIRST 0x100000 +#define S6_I2S_CUR_SCK 0x200000 +#define S6_I2S_CUR_WS 0x400000 +#define S6_I2S_ENABLE(c) (0x48 + 0x10 * (c)) +#define S6_I2S_DISABLE_IF 0x02 +#define S6_I2S_ENABLE_IF 0x03 +#define S6_I2S_IS_BUSY 0x04 +#define S6_I2S_DMA_ACTIVE 0x08 +#define S6_I2S_IS_ENABLED 0x10 + +#define S6_I2S_NUM_LINES 4 + +#define S6_I2S_SIF_PORT0 0x0000000 +#define S6_I2S_SIF_PORT1 0x0000080 /* docs say 0x0000010 */ + +static inline void s6_i2s_write_reg(struct s6000_i2s_dev *dev, int reg, u32 val) +{ + writel(val, dev->scbbase + reg); +} + +static inline u32 s6_i2s_read_reg(struct s6000_i2s_dev *dev, int reg) +{ + return readl(dev->scbbase + reg); +} + +static inline void s6_i2s_mod_reg(struct s6000_i2s_dev *dev, int reg, + u32 mask, u32 val) +{ + val ^= s6_i2s_read_reg(dev, reg) & ~mask; + s6_i2s_write_reg(dev, reg, val); +} + +static void s6000_i2s_start_channel(struct s6000_i2s_dev *dev, int channel) +{ + int i, j, cur, prev; + + /* + * Wait for WCLK to toggle 5 times before enabling the channel + * s6000 Family Datasheet 3.6.4: + * "At least two cycles of WS must occur between commands + * to disable or enable the interface" + */ + j = 0; + prev = ~S6_I2S_CUR_WS; + for (i = 1000000; --i && j < 6; ) { + cur = s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(channel)) + & S6_I2S_CUR_WS; + if (prev != cur) { + prev = cur; + j++; + } + } + if (j < 6) + printk(KERN_WARNING "s6000-i2s: timeout waiting for WCLK\n"); + + s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_ENABLE_IF); +} + +static void s6000_i2s_stop_channel(struct s6000_i2s_dev *dev, int channel) +{ + s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_DISABLE_IF); +} + +static void s6000_i2s_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data; + int channel; + + channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dev->channel_out : dev->channel_in; + + s6000_i2s_start_channel(dev, channel); +} + +static void s6000_i2s_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data; + int channel; + + channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dev->channel_out : dev->channel_in; + + s6000_i2s_stop_channel(dev, channel); +} + +static int s6000_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + int after) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ^ !after) + s6000_i2s_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!after) + s6000_i2s_stop(substream); + } + return 0; +} + +static unsigned int s6000_i2s_int_sources(struct s6000_i2s_dev *dev) +{ + unsigned int pending; + pending = s6_i2s_read_reg(dev, S6_I2S_INTERRUPT_RAW); + pending &= S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN; + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, pending); + + return pending; +} + +static unsigned int s6000_i2s_check_xrun(struct snd_soc_dai *cpu_dai) +{ + struct s6000_i2s_dev *dev = cpu_dai->private_data; + unsigned int errors; + unsigned int ret; + + errors = s6000_i2s_int_sources(dev); + if (likely(!errors)) + return 0; + + ret = 0; + if (errors & S6_I2S_INT_ALIGNMENT) + printk(KERN_ERR "s6000-i2s: WCLK misaligned\n"); + if (errors & S6_I2S_INT_UNDERRUN) + ret |= 1 << SNDRV_PCM_STREAM_PLAYBACK; + if (errors & S6_I2S_INT_OVERRUN) + ret |= 1 << SNDRV_PCM_STREAM_CAPTURE; + return ret; +} + +static void s6000_i2s_wait_disabled(struct s6000_i2s_dev *dev) +{ + int channel; + int n = 50; + for (channel = 0; channel < 2; channel++) { + while (--n >= 0) { + int v = s6_i2s_read_reg(dev, S6_I2S_ENABLE(channel)); + if ((v & S6_I2S_IS_ENABLED) + || !(v & (S6_I2S_DMA_ACTIVE | S6_I2S_IS_BUSY))) + break; + udelay(20); + } + } + if (n < 0) + printk(KERN_WARNING "s6000-i2s: timeout disabling interfaces"); +} + +static int s6000_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s6000_i2s_dev *dev = cpu_dai->private_data; + u32 w; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + w = S6_I2S_SCK_IN | S6_I2S_WS_IN; + break; + case SND_SOC_DAIFMT_CBS_CFM: + w = S6_I2S_SCK_OUT | S6_I2S_WS_IN; + break; + case SND_SOC_DAIFMT_CBM_CFS: + w = S6_I2S_SCK_IN | S6_I2S_WS_OUT; + break; + case SND_SOC_DAIFMT_CBS_CFS: + w = S6_I2S_SCK_OUT | S6_I2S_WS_OUT; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + w |= S6_I2S_LEFT_FIRST; + break; + case SND_SOC_DAIFMT_IB_NF: + w |= S6_I2S_RIGHT_FIRST; + break; + default: + return -EINVAL; + } + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(0), + S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(1), + S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); + + return 0; +} + +static int s6000_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct s6000_i2s_dev *dev = dai->private_data; + + if (!div || (div & 1) || div > (S6_I2S_DIV_MASK + 1) * 2) + return -EINVAL; + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(div_id), + S6_I2S_DIV_MASK, div / 2 - 1); + return 0; +} + +static int s6000_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct s6000_i2s_dev *dev = dai->private_data; + int interf; + u32 w = 0; + + if (dev->wide) + interf = 0; + else { + w |= (((params_channels(params) - 2) / 2) + << S6_I2S_CHANNELS_SHIFT) & S6_I2S_CHANNELS_MASK; + interf = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ? dev->channel_out : dev->channel_in; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + w |= S6_I2S_16BIT | S6_I2S_MEM_16BIT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + w |= S6_I2S_32BIT | S6_I2S_MEM_32BIT; + break; + default: + printk(KERN_WARNING "s6000-i2s: unsupported PCM format %x\n", + params_format(params)); + return -EINVAL; + } + + if (s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(interf)) + & S6_I2S_IS_ENABLED) { + printk(KERN_ERR "s6000-i2s: interface already enabled\n"); + return -EBUSY; + } + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(interf), + S6_I2S_CHANNELS_MASK|S6_I2S_MEM_MASK|S6_I2S_BITS_MASK, + w); + + return 0; +} + +static int s6000_i2s_dai_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct s6000_i2s_dev *dev = dai->private_data; + struct s6000_snd_platform_data *pdata = pdev->dev.platform_data; + + if (!pdata) + return -EINVAL; + + dev->wide = pdata->wide; + dev->channel_in = pdata->channel_in; + dev->channel_out = pdata->channel_out; + dev->lines_in = pdata->lines_in; + dev->lines_out = pdata->lines_out; + + s6_i2s_write_reg(dev, S6_I2S_MODE, + dev->wide ? S6_I2S_WIDE : S6_I2S_DUAL); + + if (dev->wide) { + int i; + + if (dev->lines_in + dev->lines_out > S6_I2S_NUM_LINES) + return -EINVAL; + + dev->channel_in = 0; + dev->channel_out = 1; + dai->capture.channels_min = 2 * dev->lines_in; + dai->capture.channels_max = dai->capture.channels_min; + dai->playback.channels_min = 2 * dev->lines_out; + dai->playback.channels_max = dai->playback.channels_min; + + for (i = 0; i < dev->lines_out; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_OUT); + + for (; i < S6_I2S_NUM_LINES - dev->lines_in; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), + S6_I2S_UNUSED); + + for (; i < S6_I2S_NUM_LINES; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_IN); + } else { + unsigned int cfg[2] = {S6_I2S_UNUSED, S6_I2S_UNUSED}; + + if (dev->lines_in > 1 || dev->lines_out > 1) + return -EINVAL; + + dai->capture.channels_min = 2 * dev->lines_in; + dai->capture.channels_max = 8 * dev->lines_in; + dai->playback.channels_min = 2 * dev->lines_out; + dai->playback.channels_max = 8 * dev->lines_out; + + if (dev->lines_in) + cfg[dev->channel_in] = S6_I2S_IN; + if (dev->lines_out) + cfg[dev->channel_out] = S6_I2S_OUT; + + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(0), cfg[0]); + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(1), cfg[1]); + } + + if (dev->lines_out) { + if (dev->lines_in) { + if (!dev->dma_params.dma_out) + return -ENODEV; + } else { + dev->dma_params.dma_out = dev->dma_params.dma_in; + dev->dma_params.dma_in = 0; + } + } + dev->dma_params.sif_in = dev->sifbase + (dev->channel_in ? + S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); + dev->dma_params.sif_out = dev->sifbase + (dev->channel_out ? + S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); + dev->dma_params.same_rate = pdata->same_rate | pdata->wide; + return 0; +} + +#define S6000_I2S_RATES (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops s6000_i2s_dai_ops = { + .set_fmt = s6000_i2s_set_dai_fmt, + .set_clkdiv = s6000_i2s_set_clkdiv, + .hw_params = s6000_i2s_hw_params, +}; + +struct snd_soc_dai s6000_i2s_dai = { + .name = "s6000-i2s", + .id = 0, + .probe = s6000_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 8, + .formats = S6000_I2S_FORMATS, + .rates = S6000_I2S_RATES, + .rate_min = 0, + .rate_max = 1562500, + }, + .capture = { + .channels_min = 2, + .channels_max = 8, + .formats = S6000_I2S_FORMATS, + .rates = S6000_I2S_RATES, + .rate_min = 0, + .rate_max = 1562500, + }, + .ops = &s6000_i2s_dai_ops, +} +EXPORT_SYMBOL_GPL(s6000_i2s_dai); + +static int __devinit s6000_i2s_probe(struct platform_device *pdev) +{ + struct s6000_i2s_dev *dev; + struct resource *scbmem, *sifmem, *region, *dma1, *dma2; + u8 __iomem *mmio; + int ret; + + scbmem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!scbmem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_none; + } + + region = request_mem_region(scbmem->start, + scbmem->end - scbmem->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S SCB region already claimed\n"); + ret = -EBUSY; + goto err_release_none; + } + + mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1); + if (!mmio) { + dev_err(&pdev->dev, "can't ioremap SCB region\n"); + ret = -ENOMEM; + goto err_release_scb; + } + + sifmem = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!sifmem) { + dev_err(&pdev->dev, "no second mem resource?\n"); + ret = -ENODEV; + goto err_release_map; + } + + region = request_mem_region(sifmem->start, + sifmem->end - sifmem->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S SIF region already claimed\n"); + ret = -EBUSY; + goto err_release_map; + } + + dma1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dma1) { + dev_err(&pdev->dev, "no dma resource?\n"); + ret = -ENODEV; + goto err_release_sif; + } + + region = request_mem_region(dma1->start, dma1->end - dma1->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S DMA region already claimed\n"); + ret = -EBUSY; + goto err_release_sif; + } + + dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (dma2) { + region = request_mem_region(dma2->start, + dma2->end - dma2->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, + "I2S DMA region already claimed\n"); + ret = -EBUSY; + goto err_release_dma1; + } + } + + dev = kzalloc(sizeof(struct s6000_i2s_dev), GFP_KERNEL); + if (!dev) { + ret = -ENOMEM; + goto err_release_dma2; + } + + s6000_i2s_dai.dev = &pdev->dev; + s6000_i2s_dai.private_data = dev; + s6000_i2s_dai.dma_data = &dev->dma_params; + + dev->sifbase = sifmem->start; + dev->scbbase = mmio; + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, + S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN); + + s6000_i2s_stop_channel(dev, 0); + s6000_i2s_stop_channel(dev, 1); + s6000_i2s_wait_disabled(dev); + + dev->dma_params.check_xrun = s6000_i2s_check_xrun; + dev->dma_params.trigger = s6000_i2s_trigger; + dev->dma_params.dma_in = dma1->start; + dev->dma_params.dma_out = dma2 ? dma2->start : 0; + dev->dma_params.irq = platform_get_irq(pdev, 0); + if (dev->dma_params.irq < 0) { + dev_err(&pdev->dev, "no irq resource?\n"); + ret = -ENODEV; + goto err_release_dev; + } + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, + S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN); + + ret = snd_soc_register_dai(&s6000_i2s_dai); + if (ret) + goto err_release_dev; + + return 0; + +err_release_dev: + kfree(dev); +err_release_dma2: + if (dma2) + release_mem_region(dma2->start, dma2->end - dma2->start + 1); +err_release_dma1: + release_mem_region(dma1->start, dma1->end - dma1->start + 1); +err_release_sif: + release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1); +err_release_map: + iounmap(mmio); +err_release_scb: + release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1); +err_release_none: + return ret; +} + +static void __devexit s6000_i2s_remove(struct platform_device *pdev) +{ + struct s6000_i2s_dev *dev = s6000_i2s_dai.private_data; + struct resource *region; + void __iomem *mmio = dev->scbbase; + + snd_soc_unregister_dai(&s6000_i2s_dai); + + s6000_i2s_stop_channel(dev, 0); + s6000_i2s_stop_channel(dev, 1); + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); + s6000_i2s_dai.private_data = 0; + kfree(dev); + + region = platform_get_resource(pdev, IORESOURCE_DMA, 0); + release_mem_region(region->start, region->end - region->start + 1); + + region = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (region) + release_mem_region(region->start, + region->end - region->start + 1); + + region = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(region->start, (region->end - region->start) + 1); + + iounmap(mmio); + region = platform_get_resource(pdev, IORESOURCE_IO, 0); + release_mem_region(region->start, (region->end - region->start) + 1); +} + +static struct platform_driver s6000_i2s_driver = { + .probe = s6000_i2s_probe, + .remove = __devexit_p(s6000_i2s_remove), + .driver = { + .name = "s6000-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init s6000_i2s_init(void) +{ + return platform_driver_register(&s6000_i2s_driver); +} +module_init(s6000_i2s_init); + +static void __exit s6000_i2s_exit(void) +{ + platform_driver_unregister(&s6000_i2s_driver); +} +module_exit(s6000_i2s_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-i2s.h b/sound/soc/s6000/s6000-i2s.h new file mode 100644 index 0000000..2375fdf --- /dev/null +++ b/sound/soc/s6000/s6000-i2s.h @@ -0,0 +1,25 @@ +/* + * ALSA SoC I2S Audio Layer for the Stretch s6000 family + * + * Author: Daniel Gloeckner, <dg(a)emlix.com> + * Copyright: (C) 2009 emlix GmbH <info(a)emlix.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 _S6000_I2S_H +#define _S6000_I2S_H + +extern struct snd_soc_dai s6000_i2s_dai; + +struct s6000_snd_platform_data { + int lines_in; + int lines_out; + int channel_in; + int channel_out; + int wide; + int same_rate; +}; +#endif diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c new file mode 100644 index 0000000..83b8028 --- /dev/null +++ b/sound/soc/s6000/s6000-pcm.c @@ -0,0 +1,497 @@ +/* + * ALSA PCM interface for the Stetch s6000 family + * + * Author: Daniel Gloeckner, <dg(a)emlix.com> + * Copyright: (C) 2009 emlix GmbH <info(a)emlix.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. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <variant/dmac.h> + +#include "s6000-pcm.h" + +#define S6_PCM_PREALLOCATE_SIZE (96 * 1024) +#define S6_PCM_PREALLOCATE_MAX (2048 * 1024) + +static struct snd_pcm_hardware s6000_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000), + .rate_min = 0, + .rate_max = 1562500, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = 0x7ffffff0, + .period_bytes_min = 16, + .period_bytes_max = 0xfffff0, + .periods_min = 2, + .periods_max = 1024, /* no limit */ + .fifo_size = 0, +}; + +struct s6000_runtime_data { + spinlock_t lock; + int period; /* current DMA period */ +}; + +static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int channel; + unsigned int period_size; + unsigned int dma_offset; + dma_addr_t dma_pos; + dma_addr_t src, dst; + + period_size = snd_pcm_lib_period_bytes(substream); + dma_offset = prtd->period * period_size; + dma_pos = runtime->dma_addr + dma_offset; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = dma_pos; + dst = par->sif_out; + channel = par->dma_out; + } else { + src = par->sif_in; + dst = dma_pos; + channel = par->dma_in; + } + + if (!s6dmac_channel_enabled(DMA_MASK_DMAC(channel), + DMA_INDEX_CHNL(channel))) + return; + + if (s6dmac_fifo_full(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel))) { + printk(KERN_ERR "s6000-pcm: fifo full\n"); + return; + } + + BUG_ON(period_size & 15); + s6dmac_put_fifo(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel), + src, dst, period_size); + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +static irqreturn_t s6000_pcm_irq(int irq, void *data) +{ + struct snd_pcm *pcm = data; + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_runtime_data *prtd; + unsigned int has_xrun; + int i, ret = IRQ_NONE; + u32 channel[2] = { + [SNDRV_PCM_STREAM_PLAYBACK] = params->dma_out, + [SNDRV_PCM_STREAM_CAPTURE] = params->dma_in + }; + + has_xrun = params->check_xrun(runtime->dai->cpu_dai); + + for (i = 0; i < ARRAY_SIZE(channel); ++i) { + struct snd_pcm_substream *substream = pcm->streams[i].substream; + unsigned int pending; + + if (!channel[i]) + continue; + + if (unlikely(has_xrun & (1 << i)) && + substream->runtime && + snd_pcm_running(substream)) { + dev_dbg(pcm->dev, "xrun\n"); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + ret = IRQ_HANDLED; + } + + pending = s6dmac_int_sources(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i])); + + if (pending & 1) { + ret = IRQ_HANDLED; + if (likely(substream->runtime && + snd_pcm_running(substream))) { + snd_pcm_period_elapsed(substream); + dev_dbg(pcm->dev, "period elapsed %x %x\n", + s6dmac_cur_src(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i])), + s6dmac_cur_dst(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i]))); + prtd = substream->runtime->private_data; + spin_lock(&prtd->lock); + s6000_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } + } + + if (unlikely(pending & ~7)) { + if (pending & (1 << 3)) + printk(KERN_WARNING + "s6000-pcm: DMA %x Underflow\n", + channel[i]); + if (pending & (1 << 4)) + printk(KERN_WARNING + "s6000-pcm: DMA %x Overflow\n", + channel[i]); + if (pending & 0x1e0) + printk(KERN_WARNING + "s6000-pcm: DMA %x Master Error " + "(mask %x)\n", + channel[i], pending >> 5); + + } + } + + return ret; +} + +static int s6000_pcm_start(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + unsigned long flags; + int srcinc; + u32 dma; + + spin_lock_irqsave(&prtd->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + srcinc = 1; + dma = par->dma_out; + } else { + srcinc = 0; + dma = par->dma_in; + } + s6dmac_enable_chan(DMA_MASK_DMAC(dma), DMA_INDEX_CHNL(dma), + 1 /* priority 1 (0 is max) */, + 0 /* peripheral requests w/o xfer length mode */, + srcinc /* source address increment */, + srcinc^1 /* destination address increment */, + 0 /* chunksize 0 (skip impossible on this dma) */, + 0 /* source skip after chunk (impossible) */, + 0 /* destination skip after chunk (impossible) */, + 4 /* 16 byte burst size */, + -1 /* don't conserve bandwidth */, + 0 /* low watermark irq descriptor theshold */, + 0 /* disable hardware timestamps */, + 1 /* enable channel */); + + s6000_pcm_enqueue_dma(substream); + s6000_pcm_enqueue_dma(substream); + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +static int s6000_pcm_stop(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + unsigned long flags; + u32 channel; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + channel = par->dma_out; + else + channel = par->dma_in; + + s6dmac_set_terminal_count(DMA_MASK_DMAC(channel), + DMA_INDEX_CHNL(channel), 0); + + spin_lock_irqsave(&prtd->lock, flags); + + s6dmac_disable_chan(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel)); + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int ret; + + ret = par->trigger(substream, cmd, 0); + if (ret < 0) + return ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = s6000_pcm_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = s6000_pcm_stop(substream); + break; + default: + ret = -EINVAL; + } + if (ret < 0) + return ret; + + return par->trigger(substream, cmd, 1); +} + +static int s6000_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + + prtd->period = 0; + + return 0; +} + +static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + unsigned long flags; + unsigned int offset; + dma_addr_t count; + + spin_lock_irqsave(&prtd->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + count = s6dmac_cur_src(DMA_MASK_DMAC(par->dma_out), + DMA_INDEX_CHNL(par->dma_out)); + else + count = s6dmac_cur_dst(DMA_MASK_DMAC(par->dma_in), + DMA_INDEX_CHNL(par->dma_in)); + + count -= runtime->dma_addr; + + spin_unlock_irqrestore(&prtd->lock, flags); + + offset = bytes_to_frames(runtime, count); + if (unlikely(offset >= runtime->buffer_size)) + offset = 0; + + return offset; +} + +static int s6000_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware); + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 16); + if (ret < 0) + return ret; + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16); + if (ret < 0) + return ret; + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + if (par->same_rate) { + int rate; + spin_lock(&par->lock); /* needed? */ + rate = par->rate; + spin_unlock(&par->lock); + if (rate != -1) { + ret = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_RATE, + rate, rate); + if (ret < 0) + return ret; + } + } + + prtd = kzalloc(sizeof(struct s6000_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int s6000_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int ret; + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) { + printk(KERN_WARNING "s6000-pcm: allocation of memory failed\n"); + return ret; + } + + if (par->same_rate) { + spin_lock(&par->lock); + if (par->rate == -1 || + !(par->in_use & ~(1 << substream->stream))) { + par->rate = params_rate(hw_params); + par->in_use |= 1 << substream->stream; + } else if (params_rate(hw_params) != par->rate) { + snd_pcm_lib_free_pages(substream); + par->in_use &= ~(1 << substream->stream); + ret = -EBUSY; + } + spin_unlock(&par->lock); + } + return ret; +} + +static int s6000_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + + spin_lock(&par->lock); + par->in_use &= ~(1 << substream->stream); + if (!par->in_use) + par->rate = -1; + spin_unlock(&par->lock); + + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_pcm_ops s6000_pcm_ops = { + .open = s6000_pcm_open, + .close = s6000_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = s6000_pcm_hw_params, + .hw_free = s6000_pcm_hw_free, + .trigger = s6000_pcm_trigger, + .prepare = s6000_pcm_prepare, + .pointer = s6000_pcm_pointer, +}; + +static void s6000_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + + free_irq(params->irq, pcm); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static u64 s6000_pcm_dmamask = DMA_32BIT_MASK; + +static int s6000_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + int res; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s6000_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_32BIT_MASK; + + if (params->dma_in) { + s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), + DMA_INDEX_CHNL(params->dma_in)); + s6dmac_int_sources(DMA_MASK_DMAC(params->dma_in), + DMA_INDEX_CHNL(params->dma_in)); + } + + if (params->dma_out) { + s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_out), + DMA_INDEX_CHNL(params->dma_out)); + s6dmac_int_sources(DMA_MASK_DMAC(params->dma_out), + DMA_INDEX_CHNL(params->dma_out)); + } + + res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED, + s6000_soc_platform.name, pcm); + if (res) { + printk(KERN_ERR "s6000-pcm couldn't get IRQ\n"); + return res; + } + + res = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, + S6_PCM_PREALLOCATE_SIZE, + S6_PCM_PREALLOCATE_MAX); + if (res) + printk(KERN_WARNING "s6000-pcm: preallocation failed\n"); + + spin_lock_init(&params->lock); + params->in_use = 0; + params->rate = -1; + return 0; +} + +struct snd_soc_platform s6000_soc_platform = { + .name = "s6000-audio", + .pcm_ops = &s6000_pcm_ops, + .pcm_new = s6000_pcm_new, + .pcm_free = s6000_pcm_free, +}; +EXPORT_SYMBOL_GPL(s6000_soc_platform); + +static int __init s6000_pcm_init(void) +{ + return snd_soc_register_platform(&s6000_soc_platform); +} +module_init(s6000_pcm_init); + +static void __exit s6000_pcm_exit(void) +{ + snd_soc_unregister_platform(&s6000_soc_platform); +} +module_exit(s6000_pcm_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-pcm.h b/sound/soc/s6000/s6000-pcm.h new file mode 100644 index 0000000..96f23f6 --- /dev/null +++ b/sound/soc/s6000/s6000-pcm.h @@ -0,0 +1,35 @@ +/* + * ALSA PCM interface for the Stretch s6000 family + * + * Author: Daniel Gloeckner, <dg(a)emlix.com> + * Copyright: (C) 2009 emlix GmbH <info(a)emlix.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 _S6000_PCM_H +#define _S6000_PCM_H + +struct snd_soc_dai; +struct snd_pcm_substream; + +struct s6000_pcm_dma_params { + unsigned int (*check_xrun)(struct snd_soc_dai *cpu_dai); + int (*trigger)(struct snd_pcm_substream *substream, int cmd, int after); + dma_addr_t sif_in; + dma_addr_t sif_out; + u32 dma_in; + u32 dma_out; + int irq; + int same_rate; + + spinlock_t lock; + int in_use; + int rate; +}; + +extern struct snd_soc_platform s6000_soc_platform; + +#endif -- 1.6.1.3

3 6

[alsa-devel] ASoC: Correct DSP/TDM mode (for twl4030 4 channel mode)
by Peter Ujfalusi 06 Apr '09

06 Apr '09

Hello, I have now the implementation for the 4 channel support on the twl4030 codec driver (and also for the omap-mcbsp). It does work nicely, but I'm not sure what Interface format (DSP_A or DSP_B or TDM?) should I use for it. Can someone help me out on this one? The TPS65950 TRM is publicly available, which has exactly the same audio block: http://focus.tij.co.jp/jp/lit/ug/swcu050b/swcu050b.pdf Page 712 (TDM mode, Sample length 16 bits; Word length 16 bits) My bet goes to SND_SOC_DAIFMT_DSP_A, but I'm not 100% sure about it. Thank you, Péter

2 1

[alsa-devel] [PATCH 0/5] ARM: DaVinci: ASoC: Add audio support for DM646x
by Naresh Medisetty 06 Apr '09

06 Apr '09

Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch set adds audio support for DM646x This series is dependant on the following 3 patches submitted to the DaVinci git: [PATCH v2 1/2] ARM: DaVinci: EDMA: Move queue related mappings to dm<soc>.c [PATCH v2 2/2] ARM: DaVinci: EDMA: Enable EDMA support for dm646x [PATCH] ARM: DaVinci: Fix the machine_is_davinci_dm6467_evm() error Patch 1: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 2: Adds McASP clock support for the two instances of mcasp casp0,mcasp1) This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 3: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 4: Add dummy codec support Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 5: Add machine driver support for DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> on Monstavista 2.6.18 kernel.

3 3

[alsa-devel] [PATCH] Add 92HD81B1C device id
by Matthew Ranostay 06 Apr '09

06 Apr '09

Added device id in struct for codec 92HD81B1C (0x111d76d5). Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b5e108a..b34d78b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4895,6 +4895,7 @@ again: switch (codec->vendor_id) { case 0x111d7604: case 0x111d7605: + case 0x111d76d5: if (spec->board_config == STAC_92HD83XXX_PWR_REF) break; spec->num_pwrs = 0; @@ -5707,6 +5708,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, -- 1.5.6.3

2 1

[alsa-devel] [PATCH] snd-atmel-abdac: increase periods_min to 6 instead of 4
by Hans-Christian Egtvedt 06 Apr '09

06 Apr '09

This patch increases periods_min to 6 from 4, this will remove any hickups where the buffer is not filled fast enough from user space. Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt(a)atmel.com> --- sound/atmel/abdac.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index aeda7cf..232de1e 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -176,7 +176,7 @@ static struct snd_pcm_hardware atmel_abdac_hw = { .buffer_bytes_max = 64 * 4096, .period_bytes_min = 4096, .period_bytes_max = 4096, - .periods_min = 4, + .periods_min = 6, .periods_max = 64, }; -- 1.5.6.3

2 1

[alsa-devel] [PATCH] hda - enable SPDIF output for Intel DX58SO board
by Wu Fengguang 06 Apr '09

06 Apr '09

ALC889 has two SPDIF outputs: 0x06, 0x10. Board vendors can use either or both. DX58SO uses 0x10, but the driver assumes 0x06. The safe solution is to add 0x10 as slave output to the existing 0x06. Reported-by: Jeroen Van Breedam <jeroen.vanbreedam(a)sgr5.be> Tested-by: Jeroen Van Breedam <jeroen.vanbreedam(a)sgr5.be> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -8764,6 +8764,10 @@ static struct snd_pci_quirk alc883_cfg_t {} }; +static hda_nid_t alc883_slave_dig_outs[] = { + ALC1200_DIGOUT_NID, 0, +}; + static hda_nid_t alc1200_slave_dig_outs[] = { ALC883_DIGOUT_NID, 0, }; @@ -8809,6 +8813,7 @@ static struct alc_config_preset alc883_p .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc883_slave_dig_outs, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes), .channel_mode = alc883_3ST_6ch_intel_modes, .need_dac_fix = 1,

2 1

[alsa-devel] [PATCH] opl3sa2: add ZV port control
by Krzysztof Helt 06 Apr '09

06 Apr '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add ZV port control switch. This patch is done after solution given in the ALSA bug #2872 report. The patch resolves the issue. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opl3sa2.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index f395b0f..d161e74 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -477,6 +477,7 @@ OPL3SA2_DOUBLE_TLV("Master Playback Volume", 0, 0x07, 0x08, 0, 0, 15, 1, OPL3SA2_SINGLE("Mic Playback Switch", 0, 0x09, 7, 1, 1), OPL3SA2_SINGLE_TLV("Mic Playback Volume", 0, 0x09, 0, 31, 1, db_scale_5bit_12db_max), +OPL3SA2_SINGLE("ZV Port Switch", 0, 0x02, 0, 1, 0), }; static struct snd_kcontrol_new snd_opl3sa2_tone_controls[] = { -- 1.5.2.2 ---------------------------------------------------------------------- Szukasz pieniedzy? Wez podwojny limit zadluzenia w koncie direct. >> http://link.interia.pl/f20a3

2 1

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 06 Apr '09

06 Apr '09

The following changes since commit 1f2186951e02f2a5bcda9459f63136918932385a: Takashi Iwai (1): ALSA: Fix wrong pointer to dev_err() in arm/pxa2xx-ac97-lib.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Philipp Zabel (2): ASoC: pxa-ssp: Use 16-bit DMA for magician stereo ASoC: Add Magician machine support sound/soc/pxa/Kconfig | 10 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 560 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa-ssp.c | 12 +- 4 files changed, 581 insertions(+), 3 deletions(-) create mode 100644 sound/soc/pxa/magician.c

3 3

[alsa-devel] X-FI Datasheets Question
by daedra 05 Apr '09

05 Apr '09

I remember about a year ago James Dutton received the datasheets from creative for the X-FI. I was just wondering if he ever attempted to write a driver? I am just curious because now we have the source driver from creative, but I am hoping for some more features like HWMIX and Wavetable support for the X-FI. Any chance of these features in the future?

1 0

[alsa-devel] Device creation order
by Lennart Poettering 05 Apr '09

05 Apr '09

Heya! PulseAudio listens for hotplugged audio devices via hal/udev. For each card ALSA creates a bunch of device nodes in /dev. Before PA can open the card it needs to make sure that all devices nodes of the card got properly created. I.e. to make sure that surround sound and SPDIF works it is not sufficient to wait until one PCM device is available, but instead that *all* devices that belong to the card are available, i.e. device nodes created with permissions and ACLs set up correctly. Unfortunately there is no signal from udev/hal/kernel that would tell me explicitly that all subdevices of a specific devices are completely enumerated and device files created for. I have been discussing this a bit with Kay Sievers and we came to the conclusion that a simple fix would be if we could rely that the control (i.e. /dev/snd/controlCxx) device is always the last device to be created for a card. Looking at the drivers this seems to be generally the case, so I was wondering if I can rely on this? Do all drivers behave like this? Do all drivers expose a control device? Can we officially declare it part of the userspace API that the control device is the last one to be created? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

7 15

[alsa-devel] Mixer problem
by James Gardiner 05 Apr '09

05 Apr '09

I'm currently in the process of adding support for my HP dv4-1222nr laptop to the drivers. It has an IDT 92HD71B7 codec chip and no matter how hard I try, the mixer wants to default to the ATI SB one. I'm checking my /proc and dmesg and it's telling me the same thing. The weird thing is that using "model=hp-dv5" in my modprobe is fixing the 0x0d node speaker problem and my mods to get the digital microphone operative are working somewhat. However, the mixer in the sigmatel source shows that it's capable of adding dmic boost to the mixer controls. A printk I added in the source at the point where it would start processing the dmic is not even reaching the function. I tried adding my subsystem id to the PCI quirks and that didn't even help. I'm lost... Any ideas?

1 0

[alsa-devel] 0 or EAGAIN from snd_pcm_update_avail()?
by Ludovico Cavedon 05 Apr '09

05 Apr '09

Hi all, if snd_pcm_update_avail() gets called on a PCM device in non-blocking mode when no data is availabe jet, what is the expected return value? -0 -EAGAIN -either is valid I am asking this question because pulseaudio expect to receive 0, but the the bluetooth headset pcm transfer callback returns EAGAIN, so I want to find out against whoms I should open a bug :) Thanks in advance, Ludovico

2 2

[alsa-devel] how to use mcbsp for i2s?
by Ernesto Torres 04 Apr '09

04 Apr '09

I'm currently working in adding an external codec to the beagleboard, so far the i2c control part is functional, now i want to use mcbsp3 to provide i2s transfers Is there any way to test the mcbsp to validate the connection? Also, I've done little modifications to the omap3beagle.c to change from mcbsp2 to mcbsp3, can someone validate this is correct? In: static int __init omap3beagle_soc_init(void) I changed to: *(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 2; /* McBSP3 */ It was: *(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 1; /* McBSP2 */ Is this correct? Or it should be changed in other places too? Thanks

3 7

[alsa-devel] bug report
by Jeremy 04 Apr '09

04 Apr '09

I was told to post this as a bug report I dont have mic boost on my creative blaster Audigy SE . I am in Debian Squeezey. Here is my alsa info from the script: http://www.alsa-project.org/db/?f=a6a639d7543e25379e5ed2e0e9e3fbab35d285cb there is actually another mic on my webcam that is usb, if you are wondering what that is. Thanks, Jeremy

1 0

[alsa-devel] [PATCH] ASoC: TWL4030: Add actuall support for 96KHz playback support
by Peter Ujfalusi 03 Apr '09

03 Apr '09

Adds the needed code to be able to use 96KHz playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 3 +++ sound/soc/codecs/twl4030.h | 1 + 2 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4199498..bfda7a8 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1311,6 +1311,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, case 48000: mode |= TWL4030_APLL_RATE_48000; break; + case 96000: + mode |= TWL4030_APLL_RATE_96000; + break; default: printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", params_rate(params)); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 33dbb14..cb63765 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -109,6 +109,7 @@ #define TWL4030_APLL_RATE_32000 0x80 #define TWL4030_APLL_RATE_44100 0x90 #define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_APLL_RATE_96000 0xE0 #define TWL4030_SEL_16K 0x04 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 -- 1.6.2

2 1

[alsa-devel] [PATCH][TRIVIAL] remove extra space
by Viral Mehta 03 Apr '09

03 Apr '09

Just for the sake of readability, removing extra space Signed-off-by: Viral Mehta <viral.mehta(a)einfochips.com> --- --- a/sound/usb/usbmixer.c 2009-03-24 04:42:14.000000000 +0530 +++ b/sound/usb/usbmixer.c 2009-04-03 12:49:49.000000000 +0530 @@ -394,7 +394,7 @@ static int set_ctl_value(struct usb_mixe value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; - while (timeout -- > 0) + while (timeout-- > 0) if (snd_usb_ctl_msg(cval->mixer->chip->dev, usb_sndctrlpipe(cval->mixer->chip->dev, 0), request,

2 1

[alsa-devel] Missing pcmC0D0p on Intel HDA with Alsa 20090327 snapshot
by Frederik Himpe 03 Apr '09

03 Apr '09

Hi, I'm using Mandriva kernel 2.6.29-tmb-desktop-2mdv, which according to the changelog contains Alsa 1.0.19+ snapshot 20090327 and Alsa 1.0.19+ unstable addons snapshot 20090327. With this kernel, I have no sound working on my Intel HDA device on a Dell Latitude E6400 system, while sound is working fine with standard 2.6.29 kernel. aplay gives this output: ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave aplay: main:590: audio open error: No such file or directory According to strace, this is because of a missing /dev/snd/pcmC0D0p. # ls /dev/snd controlC0 hwC0D0 hwC0D2 seq timer Some dmesg related output: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 HDA Intel 0000:00:1b.0: setting latency timer to 64 input: HDA Digital PCBeep as /class/input/input9 hda_codec: formats == 0 (nid=0x10, val=0xe07e0, ovrd=0, streams=0xe07e0) hda_codec: cannot attach PCM stream 0 for codec #0 hda_codec: formats == 0 (nid=0x2, val=0x1e07f0, ovrd=1, streams=0x1e07f0) hda_codec: cannot attach PCM stream 3 for codec #2 input: HDA Intel Mic at Sep Left Jack as /class/input/input10 input: HDA Intel Mic at Ext Right Jack as /class/input/input11 input: HDA Intel Line Out at Sep Left Jack as /class/input/input12 input: HDA Intel HP Out at Ext Right Jack as /class/input/input13 lspci: 00:1b.0 Audio device [0403]: Intel Corporation 82801I (ICH9 Family) HD Audio Controller [8086:293e] (rev 03) # lsmod | grep snd snd_hda_codec_intelhdmi 13856 1 snd_hda_codec_idt 69004 1 snd_hda_intel 29128 0 snd_hda_codec 68864 3 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel snd_hwdep 8488 1 snd_hda_codec snd_seq_dummy 3140 0 snd_seq_oss 31456 0 snd_seq_midi_event 7296 1 snd_seq_oss snd_seq 56704 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_seq_device 7156 3 snd_seq_dummy,snd_seq_oss,snd_seq snd_pcm_oss 43392 0 snd_pcm 80520 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_timer 22832 2 snd_seq,snd_pcm snd_mixer_oss 16512 1 snd_pcm_oss snd 65192 12 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss soundcore 7648 1 snd snd_page_alloc 9680 2 snd_hda_intel,snd_pcm -- Frederik Himpe

1 0

[alsa-devel] [PATCH 3/3] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Naresh Medisetty 03 Apr '09

03 Apr '09

This patch does the following: (1) Add support for the DM646x machine (2)Introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-evm.c | 129 ++++++++++++++++++++++++++++++++------- 1 files changed, 107 insertions(+), 22 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9b90b34..b89a322 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -20,11 +20,19 @@ #include <sound/soc-dapm.h> #include <asm/dma.h> -#include <mach/hardware.h> +#include <asm/mach-types.h> + +#include <mach/asp.h> +#include <mach/edma.h> +#include <mach/mux.h> +#include <mach/cpu.h> + #include "../codecs/tlv320aic3x.h" +#include "../codecs/davinci_codec_stubs.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-i2s-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -39,7 +47,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -140,6 +148,25 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = davinci_iis_mcasp_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "MCASP SPDIF", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[1], + .codec_dai = dit_stub_dai, + .ops = &evm_ops, + }, +}; + + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -148,12 +175,27 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = 2, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 0, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -161,36 +203,58 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DM644X_DMACH_MCBSP_TX, - .rx_dma_ch = DM644X_DMACH_MCBSP_RX, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; -static int __init evm_init(void) +static int davinci_evm_probe(struct platform_device *pdev) { + struct resource *resources; + struct evm_snd_platform_data *data = pdev->dev.platform_data; + struct snd_soc_device *evm_snd_dev_data; + int index; int ret; + int res_size = pdev->num_resources; + + if (!data) + return -ENODEV; + + resources = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!resources) + return -ENODEV; + + if (machine_is_davinci_evm()) { + davinci_cfg_reg(DM644X_MCBSP); + evm_snd_dev_data = &evm_snd_devdata; + index = 0; + } else if (machine_is_davinci_dm355_evm()) { + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + evm_snd_dev_data = &evm_snd_devdata; + index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; + } else + return -EINVAL; - evm_snd_device = platform_device_alloc("soc-audio", 0); + evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, &evm_snd_data, - sizeof(evm_snd_data)); + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = data; + + ret = platform_device_add_resources(evm_snd_device, + resources, res_size); - ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, - ARRAY_SIZE(evm_snd_resources)); if (ret) { platform_device_put(evm_snd_device); return ret; @@ -203,11 +267,32 @@ static int __init evm_init(void) return ret; } -static void __exit evm_exit(void) +static int davici_evm_remove(struct platform_device *pdev) { platform_device_unregister(evm_snd_device); + return 0; +} +static struct platform_driver davinci_asoc_driver = { + .probe = davinci_evm_probe, + .remove = davici_evm_remove, + .driver = { + .name = "davinci-asoc", + .owner = THIS_MODULE, + }, +}; + +static int __init evm_init(void) +{ + return platform_driver_register(&davinci_asoc_driver); } +static void __exit evm_exit(void) +{ + platform_driver_unregister(&davinci_asoc_driver); +} + + + module_init(evm_init); module_exit(evm_exit); -- 1.5.6

1 0

[alsa-devel] [PATCH 2/5] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by naresh＠ti.com 03 Apr '09

03 Apr '09

From: Naresh Medisetty <naresh(a)ti.com> Adds McASP clock support for the two instances of mcasp (mcasp0,mcasp1). This patch is part of the audio support for dm646x series. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/dm646x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index fbc9daf..6276b31 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -220,6 +220,18 @@ static struct clk gpio_clk = { .lpsc = DM646X_LPSC_GPIO, }; +static struct clk mcasp0_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP0, +}; + +static struct clk mcasp1_clk = { + .name = "mcasp1", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP1, +}; + static struct clk aemif_clk = { .name = "aemif", .parent = &pll1_sysclk3, @@ -307,6 +319,8 @@ struct davinci_clk dm646x_clks[] = { CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK(NULL, "gpio", &gpio_clk), + CLK(NULL, "mcasp0", &mcasp0_clk), + CLK(NULL, "mcasp1", &mcasp1_clk), CLK(NULL, "aemif", &aemif_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK(NULL, "pwm0", &pwm0_clk), -- 1.5.6

1 0

[alsa-devel] iMac 9,1 sound patch
by Micah Bucy 02 Apr '09

02 Apr '09

I tried a git snapshot, but it wasn't working at all, so this patch was developed against the 1.0.19 stable release. This is my first time delving into alsa, so I'm not sure if everything is how it should be, including if the patch was correctly created. I can tell you this. I was able to get jack detection working correctly as long as the 6 channel option is set. I had tried modifying it to only allow a 6ch option as 2ch is for line-out (headphones) only, but the internal speakers wouldn't work, so I left it alone. My guess is, this same setup could be used by the macbook pro 5,1 I'll give it a try over the weekend. I have not tested the built-in mic or the line-in. -Micah

1 0

[alsa-devel] Interleaving and Deinterleaving support
by Singaravelan Nallasellan 02 Apr '09

02 Apr '09

Hi, Does alsa core in the kernel support interleaving and deinterleaving? if the driver receives a deinterleaved stream and wants to interleave for the sound card, should the driver perform the conversion? Cheers, Sing _________________________________________________________________ Rediscover Hotmail®: Get e-mail storage that grows with you. http://windowslive.com/RediscoverHotmail?ocid=TXT_TAGLM_WL_HM_Rediscover_St…

2 2

[alsa-devel] [PATCH 5/5] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Naresh Medisetty 02 Apr '09

02 Apr '09

This patch does the following: (1) Add support for the DM646x machine (2)Introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-evm.c | 129 ++++++++++++++++++++++++++++++++------- 1 files changed, 107 insertions(+), 22 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9b90b34..b89a322 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -20,11 +20,19 @@ #include <sound/soc-dapm.h> #include <asm/dma.h> -#include <mach/hardware.h> +#include <asm/mach-types.h> + +#include <mach/asp.h> +#include <mach/edma.h> +#include <mach/mux.h> +#include <mach/cpu.h> + #include "../codecs/tlv320aic3x.h" +#include "../codecs/davinci_codec_stubs.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-i2s-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -39,7 +47,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -140,6 +148,25 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = davinci_iis_mcasp_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "MCASP SPDIF", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[1], + .codec_dai = dit_stub_dai, + .ops = &evm_ops, + }, +}; + + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -148,12 +175,27 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = 2, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 0, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -161,36 +203,58 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DM644X_DMACH_MCBSP_TX, - .rx_dma_ch = DM644X_DMACH_MCBSP_RX, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; -static int __init evm_init(void) +static int davinci_evm_probe(struct platform_device *pdev) { + struct resource *resources; + struct evm_snd_platform_data *data = pdev->dev.platform_data; + struct snd_soc_device *evm_snd_dev_data; + int index; int ret; + int res_size = pdev->num_resources; + + if (!data) + return -ENODEV; + + resources = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!resources) + return -ENODEV; + + if (machine_is_davinci_evm()) { + davinci_cfg_reg(DM644X_MCBSP); + evm_snd_dev_data = &evm_snd_devdata; + index = 0; + } else if (machine_is_davinci_dm355_evm()) { + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + evm_snd_dev_data = &evm_snd_devdata; + index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; + } else + return -EINVAL; - evm_snd_device = platform_device_alloc("soc-audio", 0); + evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, &evm_snd_data, - sizeof(evm_snd_data)); + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = data; + + ret = platform_device_add_resources(evm_snd_device, + resources, res_size); - ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, - ARRAY_SIZE(evm_snd_resources)); if (ret) { platform_device_put(evm_snd_device); return ret; @@ -203,11 +267,32 @@ static int __init evm_init(void) return ret; } -static void __exit evm_exit(void) +static int davici_evm_remove(struct platform_device *pdev) { platform_device_unregister(evm_snd_device); + return 0; +} +static struct platform_driver davinci_asoc_driver = { + .probe = davinci_evm_probe, + .remove = davici_evm_remove, + .driver = { + .name = "davinci-asoc", + .owner = THIS_MODULE, + }, +}; + +static int __init evm_init(void) +{ + return platform_driver_register(&davinci_asoc_driver); } +static void __exit evm_exit(void) +{ + platform_driver_unregister(&davinci_asoc_driver); +} + + + module_init(evm_init); module_exit(evm_exit); -- 1.5.6

2 1

[alsa-devel] [PATCH 4/5] ARM: DaVinci: ASoC: Add dummy codec support
by Naresh Medisetty 02 Apr '09

02 Apr '09

McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed. This patch provides stub codec that can be used in these configurations. On DM646x EVM the McASP1 is connected to the S/PDIF out. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/davinci_codec_stubs.c | 83 ++++++++++++++++++++++++++++++++ sound/soc/codecs/davinci_codec_stubs.h | 23 +++++++++ 4 files changed, 111 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/davinci_codec_stubs.c create mode 100644 sound/soc/codecs/davinci_codec_stubs.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b6c7f7a..59e50d0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_CODEC_STUBS select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C @@ -91,6 +92,8 @@ config SND_SOC_SSM2602 config SND_SOC_TLV320AIC23 tristate +config SND_SOC_CODEC_STUBS + tristate config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 030d245..70c26a5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -10,6 +10,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-davinci_stubs-objs := davinci_codec_stubs.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o @@ -41,6 +42,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_CODEC_STUBS) += snd-soc-davinci_stubs.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/davinci_codec_stubs.c b/sound/soc/codecs/davinci_codec_stubs.c new file mode 100644 index 0000000..63d56ad --- /dev/null +++ b/sound/soc/codecs/davinci_codec_stubs.c @@ -0,0 +1,83 @@ +/* + * ALSA SoC DaVinci DIT/DIR driver + * + * TI DaVinci audio controller can operate in DIT/DIR (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> +#include <sound/pcm.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +struct snd_soc_dai dit_stub_dai[] = { + { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + } +}; + +struct snd_soc_dai dir_stub_dai[] = { + { + .name = "DIR", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + } +}; + +static int davinci_dit_probe(struct platform_device *pdev) +{ + return snd_soc_register_dai(dit_stub_dai); +} + +static int davici_dit_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(dit_stub_dai); + return 0; +} + +static struct platform_driver davinci_dit_driver = { + .probe = davinci_dit_probe, + .remove = davici_dit_remove, + .driver = { + .name = "davinci-dit", + .owner = THIS_MODULE, + }, +}; + +static int __init dit_modinit(void) +{ + return platform_driver_register(&davinci_dit_driver); +} + +static void __exit dit_exit(void) +{ + platform_driver_unregister(&davinci_dit_driver); +} + +module_init(dit_modinit); +module_exit(dit_exit); + diff --git a/sound/soc/codecs/davinci_codec_stubs.h b/sound/soc/codecs/davinci_codec_stubs.h new file mode 100644 index 0000000..693797b --- /dev/null +++ b/sound/soc/codecs/davinci_codec_stubs.h @@ -0,0 +1,23 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.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 CODEC_STUBS_H +#define CODEC_STUBS_H + +#ifdef CONFIG_SND_SOC_CODEC_STUBS +extern struct snd_soc_dai dit_stub_dai[]; +extern struct snd_soc_dai dir_stub_dai[]; +#else +#define dit_stub_dai NULL +#define dir_stub_dai NULL +#endif + +#endif /* CODEC_STUBS_H */ -- 1.5.6

2 1

[alsa-devel] [PATCH 3/5] ARM: DaVinci: ASoC: Add the platform devices for ASP
by naresh＠ti.com 02 Apr '09

02 Apr '09

From: Naresh Medisetty <naresh(a)ti.com> Registers the platform devices for ASP on dm355, dm644x and dm646x so that the machine driver can probe to get ASP related platform data. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/dm355.c | 24 +++++++++++ arch/arm/mach-davinci/dm644x.c | 24 +++++++++++ arch/arm/mach-davinci/dm646x.c | 67 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 39 ++++++++++++++++- 4 files changed, 152 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index ee92c10..b79e9ac 100644 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -23,6 +23,7 @@ #include <mach/psc.h> #include <mach/mux.h> #include <mach/irqs.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -540,6 +541,28 @@ static struct platform_device dm355_edma_device = { .resource = edma_resources, }; +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm355_evm_snd_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm355_evm_snd_data = { + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +}; + +static struct platform_device dm355_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = &dm355_evm_snd_data, + .num_resources = ARRAY_SIZE(dm355_evm_snd_resources), + .resource = dm355_evm_snd_resources, +}; + /*----------------------------------------------------------------------*/ void __init dm355_init(void) @@ -555,6 +578,7 @@ static int __init dm355_init_devices(void) davinci_cfg_reg(DM355_INT_EDMA_CC); platform_device_register(&dm355_edma_device); + platform_device_register(&dm355_asoc_device); return 0; } postcore_initcall(dm355_init_devices); diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index d1f5790..8921c4f 100644 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -20,6 +20,7 @@ #include <mach/irqs.h> #include <mach/psc.h> #include <mach/mux.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -463,6 +464,28 @@ static struct platform_device dm644x_edma_device = { .resource = edma_resources, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ +static struct resource dm644x_evm_snd_resources[] = { + { + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm644x_evm_snd_data = { + .tx_dma_ch = DAVINCI_DMA_ASP0_TX, + .rx_dma_ch = DAVINCI_DMA_ASP0_RX, +}; + +static struct platform_device dm644x_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = &dm644x_evm_snd_data, + .num_resources = ARRAY_SIZE(dm644x_evm_snd_resources), + .resource = dm644x_evm_snd_resources, +}; + /*----------------------------------------------------------------------*/ #if defined(CONFIG_TI_DAVINCI_EMAC) || defined(CONFIG_TI_DAVINCI_EMAC_MODULE) @@ -495,6 +518,7 @@ static int __init dm644x_init_devices(void) return 0; platform_device_register(&dm644x_edma_device); + platform_device_register(&dm644x_asoc_device); return 0; } postcore_initcall(dm644x_init_devices); diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index 6276b31..f26bfcd 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -20,6 +20,7 @@ #include <mach/irqs.h> #include <mach/psc.h> #include <mach/mux.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -499,6 +500,70 @@ static struct platform_device dm646x_edma_device = { .resource = edma_resources, }; +static u8 dm6467_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm6467_dit_serializer_direction[] = { + TX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct evm_snd_platform_data dm6467_evm_snd_data[] = { + { + .clk_name = "mcasp0", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .rx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = 4, + .tdm_slots = 2, + .serial_dir = dm6467_iis_serializer_direction, + .eventq_no = EVENTQ_0, + }, + { + .clk_name = "mcasp1", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .rx_dma_ch = -1, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = 4, + .tdm_slots = 32, + .serial_dir = dm6467_dit_serializer_direction, + .eventq_no = EVENTQ_0, + }, +}; + +static struct resource dm6467_evm_snd_resources[] = { + { + .name = "mcasp0", + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + { + .name = "mcasp1", + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct platform_device dm646x_asoc_device = { + .name = "davinci-asoc", + .id = -1, + .dev.platform_data = dm6467_evm_snd_data, + .num_resources = ARRAY_SIZE(dm6467_evm_snd_resources), + .resource = dm6467_evm_snd_resources, +}; + +static struct platform_device dm646x_dit_device = { + .name = "davinci-dit", + .id = -1, + +}; + /*----------------------------------------------------------------------*/ #if defined(CONFIG_TI_DAVINCI_EMAC) || defined(CONFIG_TI_DAVINCI_EMAC_MODULE) @@ -532,6 +597,8 @@ static int __init dm646x_init_devices(void) return 0; platform_device_register(&dm646x_edma_device); + platform_device_register(&dm646x_asoc_device); + platform_device_register(&dm646x_dit_device); return 0; } postcore_initcall(dm646x_init_devices); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..1bb1f0b 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -5,21 +5,56 @@ #define __ASM_ARCH_DAVINCI_ASP_H #include <mach/irqs.h> +#include <mach/edma.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ + +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 + +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT #define DAVINCI_ASP1_RX_INT IRQ_MBRINT #define DAVINCI_ASP1_TX_INT IRQ_MBXINT +struct evm_snd_platform_data { + char *clk_name; + int tx_dma_ch; + int rx_dma_ch; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + + #endif /* __ASM_ARCH_DAVINCI_ASP_H */ -- 1.5.6

2 1

[alsa-devel] problem with speaker-test
by Sedji Gaouaou 02 Apr '09

02 Apr '09

Hi, I am trying to use speaker-test on my sam9g20 board, with the Wolfson wm8731, but I can't have any tone... Instead it returns this error: # speaker-test -c 2 -t1 speaker-test 1.0.14 Playback device is default Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 16 to 1073745880 Period size range from 0 to 0 Using max buffer size 1073745880 Periods = 4 was set period_size = 0 was set buffer_size = 8192 Unable to set sw params for playback: Invalid argument Setting of swparams failed: Invalid argument So I tried: # speaker-test -D"name" -c 2 -t1, but I don't know exactly what should I pass as "name". I did: aplay -L to list all the devices, but I don't see anything... Is anybody got an idea here? Regards, Sedji

2 2

[alsa-devel] [PATCH 1/5] ARM: DaVinci: ASoC: Add mcasp support for DM646x
by Naresh Medisetty 02 Apr '09

02 Apr '09

Adds driver support for the two instances of McASP on TI's DM646x. The multichannel audio serial port (McASP) functions as a general-purpose audio serial port optimized for the needs of multichannel audio application. (http://www.ti.com/litv/pdf/spruer1b). There are two instances of McASP on DM646x. The McASP0 module includes up to 4 serializers that can be individually enabled to either transmit or receive in different modes. The McASP1 module is limited with only 1 pinned-out serializer that can be enabled to only transmit in DIT mode (neither receiving in any mode nor transmitting in either Burst or TDM mode is supported). The codec on the DM646x EVM is AIC32 which is connected to Serializer0 and Serializer1 of McASP0. The McASP0 consists of transmit and receive sections that may operate synchronized, or completely independently with separate master clocks, bit clocks, and frame syncs, and using different transmit modes with different bit-stream formats. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/Kconfig | 12 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-i2s-mcasp.c | 877 +++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s-mcasp.h | 56 +++ sound/soc/davinci/davinci-pcm.h | 17 +- 5 files changed, 957 insertions(+), 8 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.c create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index bd7392c..7f35baa 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_I2S_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM @@ -18,6 +21,15 @@ config SND_DAVINCI_SOC_EVM Say Y if you want to add support for SoC audio on TI DaVinci EVM platform. +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM + select SND_DAVINCI_SOC_I2S_MCASP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DaVinci DM6467 EVM platform. + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1..152d59c 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,16 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-i2s-mcasp-objs:= davinci-i2s-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S_MCASP) += snd-soc-davinci-i2s-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-i2s-mcasp.c b/sound/soc/davinci/davinci-i2s-mcasp.c new file mode 100644 index 0000000..6a8d520 --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.c @@ -0,0 +1,877 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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. + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s-mcasp.h" + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left (even TDM Slot) Channel Status Register File*/ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right (odd TDM slot) Channel Status RegisterFile*/ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left (even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right (odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define MCASP_FREE BIT(0) +#define MCASP_SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define DISMOD(val) (val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ + /* and Transmit Bad Clock Detect /32 Counter + Reset */ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + + + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + int i = 0; + mcasp_set_bits(regs, val); + + /* programming GBLCTL needs to read back from GBLCTL and verfiy */ + /* loop count is to avoid the lock-up */ + for (i = 0; (i < 1000) && ((mcasp_get_reg(regs) & val) != val); i++); + +} + +int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + + return 0; +} + +int davinci_i2s_mcasp_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int tmp, link_cnt; + int count = 0; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai; + struct davinci_audio_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *parray = pdev->dev.platform_data; + struct evm_snd_platform_data *pdata; + struct davinci_pcm_dma_params *dma_data; + int ret = 0; + + for (link_cnt = 0; link_cnt < card->num_links; link_cnt++) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, link_cnt); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + } + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + } + + dev = kzalloc(sizeof(struct davinci_audio_dev) * + card->num_links, GFP_KERNEL); + if (!dev) { + return -ENOMEM; + goto err_release_region; + } + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * + (card->num_links << 1), GFP_KERNEL); + if (!dma_data) + goto err_release_dev; + cpu_dai = card->dai_link[link_cnt].cpu_dai; + cpu_dai->private_data = &dev[link_cnt]; + pdata = &parray[link_cnt]; + dev[link_cnt].clk = clk_get(&pdev->dev, pdata->clk_name); + + if (IS_ERR(dev[link_cnt].clk)) { + ret = -ENODEV; + goto err_release_data; + } + clk_enable(dev[link_cnt].clk); + + dev[link_cnt].base = (void __iomem *)IO_ADDRESS(mem->start); + dev[link_cnt].op_mode = pdata->op_mode; + dev[link_cnt].tdm_slots = pdata->tdm_slots; + dev[link_cnt].num_serializer = pdata->num_serializer; + dev[link_cnt].serial_dir = pdata->serial_dir; + dev[link_cnt].codec_fmt = pdata->codec_fmt; + + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].channel = pdata->tx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_PLAYBACK] = + &dma_data[count]; + + + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].channel = pdata->rx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_CAPTURE] = + &dma_data[count]; + count++; + + } + return 0; + +err_release_data: + kfree(dma_data); +err_release_dev: + kfree(dev); +err_release_region: + for (tmp = link_cnt; tmp >= 0; tmp--) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, tmp); + release_mem_region(mem->start, + (mem->end - mem->start) + 1); + } + + return ret; +} + +void davinci_i2s_mcasp_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int i; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct davinci_audio_dev *dev_list = cpu_dai->private_data; + struct davinci_audio_dev *dev; + struct resource *mem; + struct davinci_pcm_dma_params *ptr; + + for (i = 0; i < card->num_links; i++) { + dev = &dev_list[i]; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + } + ptr = dev_list->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(ptr); + kfree(dev_list); +} + +void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_i2s_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct snd_pcm_substream *substream, + int channel_size) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03 << 4; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05 << 4; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07 << 4; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09 << 4; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B << 4; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D << 4; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F << 4; + break; + + default: + return -1; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, fmt, (0x0F << 4)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, fmt, (0x0F << 4)); + return 0; +} + +static void davinci_hw_common_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} +static void davinci_hw_iis_param(struct snd_pcm_substream *substream) +{ + int i, active_slots; + u32 mask = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + + } else { + /* bit stream is MSB first with no delay */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + FSRDLY(1) | RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_i2s_mcasp_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(substream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(substream); + else + davinci_hw_iis_param(substream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(substream, word_length); + + return 0; +} + +static int davinci_i2s_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_iis_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .probe = davinci_i2s_mcasp_probe, + .remove = davinci_i2s_mcasp_remove, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &davinci_i2s_dai_ops, + + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &davinci_i2s_dai_ops, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_iis_mcasp_dai); + +static int __init davinci_i2s_mcasp_init(void) +{ + return snd_soc_register_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + +} +module_init(davinci_i2s_mcasp_init); + +static void __exit davinci_i2s_mcasp_exit(void) +{ + return snd_soc_unregister_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + +} +module_exit(davinci_i2s_mcasp_exit); + + + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McASP) SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/davinci/davinci-i2s-mcasp.h b/sound/soc/davinci/davinci-i2s-mcasp.h new file mode 100644 index 0000000..6469e7d --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.h @@ -0,0 +1,56 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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 DAVINCI_I2S_MCASP_H +#define DAVINCI_I2S_MCASP_H + +#include <linux/io.h> +#include <mach/asp.h> +#include "davinci-pcm.h" + +extern struct snd_soc_dai davinci_iis_mcasp_dai[]; +extern struct snd_soc_dai davinci_dit_mcasp_dai; + + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + + + +#endif /* DAVINCI_I2S_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..3cbe281 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,17 +12,18 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> +#include <mach/asp.h> + + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ + enum dma_event_q eventq_no; /* event queue number */ }; -struct evm_snd_platform_data { - int tx_dma_ch; - int rx_dma_ch; -}; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

2 1

[PATCH] ASoC: Implement suspend and resume operations for WM9705
by Mark Brown 02 Apr '09

02 Apr '09

Compile tested only. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9705.c | 37 +++++++++++++++++++++++++++++++++++++ 1 files changed, 37 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 3265817..2109b50 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -317,6 +317,41 @@ static int wm9705_reset(struct snd_soc_codec *codec) return -EIO; } +#ifdef CONFIG_PM +static int wm9705_soc_suspend(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + soc_ac97_ops.write(codec->ac97, AC97_POWERDONW, 0xffff); + + return 0; +} + +static int wm9705_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i, ret; + u16 *cache = codec->reg_cache; + + ret = wm9705_reset(codec); + if (ret < 0) { + printk(KERN_ERR "could not reset AC97 codec\n"); + return ret; + } + + for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { + soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + } + + return 0; +} +#else +#define wm9705_soc_suspend NULL +#define wm9705_soc_resume NULL +#endif + static int wm9705_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -407,6 +442,8 @@ static int wm9705_soc_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_wm9705 = { .probe = wm9705_soc_probe, .remove = wm9705_soc_remove, + .suspend = wm9705_soc_suspend, + .resume = wm9705_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705); -- 1.6.2.1

1 0

Re: [alsa-devel] [PATCH 3/3] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Medisetty, Naresh 02 Apr '09

02 Apr '09

Please ignore this patch since the patch number is wrong. I will re-submit the same patch with proper numbering. Regards Naresh This patch does the following: (1) Add support for the DM646x machine (2)Introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Naresh Medisetty naresh(a)ti.com --- _______________________________________________ Davinci-linux-open-source mailing list Davinci-linux-open-source(a)linux.davincidsp.com http://linux.davincidsp.com/mailman/listinfo/davinci-linux-open-source

1 0

[PATCH] ASoC: Implement suspend and resume operations for WM9705
by Mark Brown 02 Apr '09

02 Apr '09

Compile tested only. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9705.c | 37 +++++++++++++++++++++++++++++++++++++ 1 files changed, 37 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 3265817..0564476 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -317,6 +317,41 @@ static int wm9705_reset(struct snd_soc_codec *codec) return -EIO; } +#ifdef CONFIG_PM +static int wm9705_soc_suspend(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ac97_write(codec, AC97_POWERDOWN, 0xffff); + + return 0; +} + +static int wm9705_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i, ret; + u16 *cache = codec->reg_cache; + + ret = wm9705_reset(codec); + if (ret < 0) { + printk(KERN_ERR "could not reset AC97 codec\n"); + return ret; + } + + for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { + soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + } + + return 0; +} +#else +#define wm9705_soc_suspend NULL +#define wm9705_soc_resume NULL +#endif + static int wm9705_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -407,6 +442,8 @@ static int wm9705_soc_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_wm9705 = { .probe = wm9705_soc_probe, .remove = wm9705_soc_remove, + .suspend = wm9705_soc_suspend, + .resume = wm9705_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705); -- 1.6.2.1

1 0

[PATCH 2/2] ASoC: cs4270: add power management support
by Daniel Mack 02 Apr '09

02 Apr '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)sirena.org.uk> --- sound/soc/codecs/cs4270.c | 59 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 58 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index bc338df..65c68a0 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -18,7 +18,7 @@ * - The machine driver's 'startup' function must call * cs4270_set_dai_sysclk() with the value of MCLK. * - Only I2S and left-justified modes are supported - * - Power management is not supported + * - Power management is supported */ #include <linux/module.h> @@ -27,6 +27,7 @@ #include <sound/soc.h> #include <sound/initval.h> #include <linux/i2c.h> +#include <linux/delay.h> #include "cs4270.h" @@ -65,6 +66,8 @@ #define CS4270_PWRCTL_PDN_ADC 0x20 #define CS4270_PWRCTL_PDN_DAC 0x02 #define CS4270_PWRCTL_PDN 0x01 +#define CS4270_PWRCTL_PDN_ALL \ + (CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN) #define CS4270_MODE_SPEED_MASK 0x30 #define CS4270_MODE_1X 0x00 #define CS4270_MODE_2X 0x10 @@ -788,6 +791,58 @@ static struct i2c_device_id cs4270_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4270_id); +#ifdef CONFIG_PM + +/* This suspend/resume implementation can handle both - a simple standby + * where the codec remains powered, and a full suspend, where the voltage + * domain the codec is connected to is teared down and/or any other hardware + * reset condition is asserted. + * + * The codec's own power saving features are enabled in the suspend callback, + * and all registers are written back to the hardware when resuming. + */ + +static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) +{ + struct cs4270_private *cs4270 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = &cs4270->codec; + int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} + +static int cs4270_i2c_resume(struct i2c_client *client) +{ + struct cs4270_private *cs4270 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = &cs4270->codec; + int reg; + + /* In case the device was put to hard reset during sleep, we need to + * wait 500ns here before any I2C communication. */ + ndelay(500); + + /* first restore the entire register cache ... */ + for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { + u8 val = snd_soc_read(codec, reg); + + if (i2c_smbus_write_byte_data(client, reg, val)) { + dev_err(codec->dev, "i2c write failed\n"); + return -EIO; + } + } + + /* ... then disable the power-down bits */ + reg = snd_soc_read(codec, CS4270_PWRCTL); + if (reg < 0) + return reg; + + reg &= ~CS4270_PWRCTL_PDN_ALL; + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} +#else +#define cs4270_i2c_suspend NULL +#define cs4270_i2c_resume NULL +#endif /* CONFIG_PM */ + /* * cs4270_i2c_driver - I2C device identification * @@ -802,6 +857,8 @@ static struct i2c_driver cs4270_i2c_driver = { .id_table = cs4270_id, .probe = cs4270_i2c_probe, .remove = cs4270_i2c_remove, + .suspend = cs4270_i2c_suspend, + .resume = cs4270_i2c_resume, }; /* -- 1.6.2.1

1 0

[PATCH v3] ASoC: cs4270: add power management support
by Daniel Mack 02 Apr '09

02 Apr '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)sirena.org.uk> --- sound/soc/codecs/cs4270.c | 58 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 57 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index bc338df..4d4a45e 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -18,7 +18,7 @@ * - The machine driver's 'startup' function must call * cs4270_set_dai_sysclk() with the value of MCLK. * - Only I2S and left-justified modes are supported - * - Power management is not supported + * - Power management is supported */ #include <linux/module.h> @@ -27,6 +27,7 @@ #include <sound/soc.h> #include <sound/initval.h> #include <linux/i2c.h> +#include <linux/delay.h> #include "cs4270.h" @@ -65,6 +66,8 @@ #define CS4270_PWRCTL_PDN_ADC 0x20 #define CS4270_PWRCTL_PDN_DAC 0x02 #define CS4270_PWRCTL_PDN 0x01 +#define CS4270_PWRCTL_PDN_ALL \ + (CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN) #define CS4270_MODE_SPEED_MASK 0x30 #define CS4270_MODE_1X 0x00 #define CS4270_MODE_2X 0x10 @@ -788,6 +791,57 @@ static struct i2c_device_id cs4270_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4270_id); +#ifdef CONFIG_PM + +/* This suspend/resume implementation can handle both - a simple standby + * where the codec remains powered, and a full suspend, where the voltage + * domain the codec is connected to is teared down and/or any other hardware + * reset condition is asserted. + * + * The codec's own power saving features are enabled in the suspend callback, + * and all registers are written back to the hardware when resuming. + */ + +static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) +{ + struct cs4270_private *cs4270 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = &cs4270->codec; + int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} + +static int cs4270_i2c_resume(struct i2c_client *client) +{ + struct cs4270_private *cs4270 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = &cs4270->codec; + int reg; + + /* In case the device was put to hard reset during sleep, we need to + * wait 500ns here before any I2C communication. */ + ndelay(500); + + /* first restore the entire register cache ... */ + for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { + u8 val = snd_soc_read(codec, reg); + + if (i2c_smbus_write_byte_data(client, reg, val)) { + dev_err(codec->dev, "i2c write failed\n"); + return -EIO; + } + } + + /* ... then disable the power-down bits */ + reg = snd_soc_read(codec, CS4270_PWRCTL); + reg &= ~CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} +#else +#define cs4270_i2c_suspend NULL +#define cs4270_i2c_resume NULL +#endif /* CONFIG_PM */ + /* * cs4270_i2c_driver - I2C device identification * @@ -802,6 +856,8 @@ static struct i2c_driver cs4270_i2c_driver = { .id_table = cs4270_id, .probe = cs4270_i2c_probe, .remove = cs4270_i2c_remove, + .suspend = cs4270_i2c_suspend, + .resume = cs4270_i2c_resume, }; /* -- 1.6.2.1

1 0

[alsa-devel] [patch] tlv320aic3x: disable ADC/DAC while changing clock
by Daniel Glöckner 02 Apr '09

02 Apr '09

While using a tlv320aic3101 as a clock master, it often happened that the clock lines got stuck while changing the sampling rate. In a discussion with TI technical support it was suggested to shut down the ADC and DAC while there are changes being made to the clock. Following that rule resolves the problem, but merely writing the registers to switch the ADC/DAC off is not enough. One needs to poll the power status registers to wait until they have completely shut down. The aforementioned rule implies that we can not disable the PLL in stand-by bias level. Signed-off-by: Daniel Glöckner <dg(a)emlix.com> --- sound/soc/codecs/tlv320aic3x.c | 89 ++++++++++++++++++++++++++++------------ sound/soc/codecs/tlv320aic3x.h | 8 ++++ 2 files changed, 71 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index aea0cb7..4c44022 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -740,6 +740,57 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) return 0; } +static u8 aic3x_power_codec(struct snd_soc_codec *codec, u8 new) +{ + u8 val, old; + + val = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL); + old = (val & LADC_PWR_ON) ? 1 : 0; + if ((old ^ new) & 1) { + val ^= LADC_PWR_ON; + aic3x_write(codec, LINE1L_2_LADC_CTRL, val); + } + + val = aic3x_read_reg_cache(codec, LINE1R_2_RADC_CTRL); + old += (val & RADC_PWR_ON) ? 2 : 0; + if ((old ^ new) & 2) { + val ^= RADC_PWR_ON; + aic3x_write(codec, LINE1R_2_RADC_CTRL, val); + } + + val = aic3x_read_reg_cache(codec, DAC_PWR); + old += (val & LDAC_PWR_ON) ? 4 : 0; + old += (val & RDAC_PWR_ON) ? 8 : 0; + if ((old ^ new) & 4) + val ^= LDAC_PWR_ON; + if ((old ^ new) & 8) + val ^= RDAC_PWR_ON; + if ((old ^ new) & 12) + aic3x_write(codec, DAC_PWR, val); + + return old; +} + +static u8 aic3x_power_off_wait(struct snd_soc_codec *codec) +{ + int timeout = 1000; + u8 old, data; + old = aic3x_power_codec(codec, 0); + + do { + aic3x_read(codec, ADC_FLAGS, &data); + } while ((data & (LADC_PWR_STATUS | RADC_PWR_STATUS)) && --timeout > 0); + + do { + aic3x_read(codec, MODULE_PWR_STATUS, &data); + } while ((data & (LDAC_PWR_STATUS | RDAC_PWR_STATUS)) && --timeout > 0); + + if (timeout < 0) + printk(KERN_WARNING "aic3x: timeout waiting for ADC/DAC to shut down"); + + return old; +} + static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -749,9 +800,11 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->codec; struct aic3x_priv *aic3x = codec->private_data; int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; - u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; + u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1, state_saved; u16 pll_d = 1; + state_saved = aic3x_power_off_wait(codec); + /* select data word length */ data = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); @@ -805,8 +858,10 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data |= (data << 4); aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); - if (bypass_pll) + if (bypass_pll) { + aic3x_power_codec(codec, state_saved); return 0; + } /* Use PLL * find an apropriate setup for j, d, r and p by iterating over @@ -847,6 +902,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, if (last_clk == 0) { printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); + aic3x_power_codec(codec, state_saved); return -EINVAL; } @@ -857,7 +913,11 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, aic3x_write(codec, AIC3X_PLL_PROGC_REG, (pll_d >> 6) << PLLD_MSB_SHIFT); aic3x_write(codec, AIC3X_PLL_PROGD_REG, (pll_d & 0x3F) << PLLD_LSB_SHIFT); + data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT) + | PLL_ENABLE); + aic3x_power_codec(codec, state_saved); return 0; } @@ -951,37 +1011,14 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - /* all power is driven by DAPM system */ - if (aic3x->master) { - /* enable pll */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, - reg | PLL_ENABLE); - } break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - /* - * all power is driven by DAPM system, - * so output power is safe if bypass was set - */ - if (aic3x->master) { - /* disable pll */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, - reg & ~PLL_ENABLE); - } break; case SND_SOC_BIAS_OFF: /* force all power off */ - reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL); - aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON); - reg = aic3x_read_reg_cache(codec, LINE1R_2_RADC_CTRL); - aic3x_write(codec, LINE1R_2_RADC_CTRL, reg & ~RADC_PWR_ON); - - reg = aic3x_read_reg_cache(codec, DAC_PWR); - aic3x_write(codec, DAC_PWR, reg & ~(LDAC_PWR_ON | RDAC_PWR_ON)); + aic3x_power_off_wait(codec); reg = aic3x_read_reg_cache(codec, HPLOUT_CTRL); aic3x_write(codec, HPLOUT_CTRL, reg & ~HPLOUT_PWR_ON); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index ac827e5..0b4fedf 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -69,6 +69,8 @@ #define RAGC_CTRL_B 30 #define RAGC_CTRL_C 31 +/* ADC Flag register */ +#define ADC_FLAGS 34 /* DAC Power and Left High Power Output control registers */ #define DAC_PWR 37 #define HPLCOM_CFG 37 @@ -126,6 +128,8 @@ #define DACR1_2_LLOPM_VOL 85 #define LLOPM_CTRL 86 #define RLOPM_CTRL 93 +/* Module Power status register */ +#define MODULE_PWR_STATUS 94 /* GPIO/IRQ registers */ #define AIC3X_STICKY_IRQ_FLAGS_REG 96 #define AIC3X_RT_IRQ_FLAGS_REG 97 @@ -191,6 +195,10 @@ #define MONOLOPM_PWR_ON 0x01 #define LLOPM_PWR_ON 0x01 #define RLOPM_PWR_ON 0x01 +#define LADC_PWR_STATUS 0x40 +#define RADC_PWR_STATUS 0x04 +#define LDAC_PWR_STATUS 0x80 +#define RDAC_PWR_STATUS 0x40 #define INVERT_VOL(val) (0x7f - val) -- 1.6.2.107.ge47ee

3 14

[alsa-devel] [PATCH v2 1/9] snd-atmel-ac97c: cleanup register definitions
by Hans-Christian Egtvedt 02 Apr '09

02 Apr '09

This patch will remove traces of channel B registers, since they are not used by the AC97C driver. Channel B might be used for other purposes. The driver also adds channel status bits TXEMPTY and OVRUN and a AC97C_CH_MASK macro to ease clearing a channel settings. Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt(a)atmel.com> --- sound/atmel/ac97c.h | 14 ++++++++------ 1 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/atmel/ac97c.h b/sound/atmel/ac97c.h index c17bd58..ecbba50 100644 --- a/sound/atmel/ac97c.h +++ b/sound/atmel/ac97c.h @@ -1,5 +1,5 @@ /* - * Register definitions for the Atmel AC97C controller + * Register definitions for Atmel AC97C * * Copyright (C) 2005-2009 Atmel Corporation * @@ -17,10 +17,6 @@ #define AC97C_CATHR 0x24 #define AC97C_CASR 0x28 #define AC97C_CAMR 0x2c -#define AC97C_CBRHR 0x30 -#define AC97C_CBTHR 0x34 -#define AC97C_CBSR 0x38 -#define AC97C_CBMR 0x3c #define AC97C_CORHR 0x40 #define AC97C_COTHR 0x44 #define AC97C_COSR 0x48 @@ -46,8 +42,10 @@ #define AC97C_MR_VRA (1 << 2) #define AC97C_CSR_TXRDY (1 << 0) +#define AC97C_CSR_TXEMPTY (1 << 1) #define AC97C_CSR_UNRUN (1 << 2) #define AC97C_CSR_RXRDY (1 << 4) +#define AC97C_CSR_OVRUN (1 << 5) #define AC97C_CSR_ENDTX (1 << 10) #define AC97C_CSR_ENDRX (1 << 14) @@ -61,11 +59,15 @@ #define AC97C_CMR_DMAEN (1 << 22) #define AC97C_SR_CAEVT (1 << 3) +#define AC97C_SR_COEVT (1 << 2) +#define AC97C_SR_WKUP (1 << 1) +#define AC97C_SR_SOF (1 << 0) +#define AC97C_CH_MASK(slot) \ + (0x7 << (3 * (AC97_SLOT_##slot - 3))) #define AC97C_CH_ASSIGN(slot, channel) \ (AC97C_CHANNEL_##channel << (3 * (AC97_SLOT_##slot - 3))) #define AC97C_CHANNEL_NONE 0x0 #define AC97C_CHANNEL_A 0x1 -#define AC97C_CHANNEL_B 0x2 #endif /* __SOUND_ATMEL_AC97C_H */ -- 1.5.6.3

1 8

[alsa-devel] [PATCH 0/7] snd-atmel-ac97c: misc fixes and error reporting
by Hans-Christian Egtvedt 02 Apr '09

02 Apr '09

This patch series contains six minor fixes and one enhancment of the snd-atmel-ac97c driver. It is based on the for-next branch in sound-2.6 git repo. The enhancment is enabling interrupts for error reporting, this might be useful if you have sound troubles (glitches) on a stressed system. -- Best regards, Hans-Christian Egtvedt

2 10

[alsa-devel] iMac 9,1 sound
by Micah Bucy 02 Apr '09

02 Apr '09

I was able to get sound working on the new iMac by adding the following to alsa-driver-1.0.19/alsa-kernel/pci/hda/patch_realtek.c case 0x106b4900: /* iMac 9,1 */ underneath line 7013 case 0x106b1000: /* iMac 24 */ (this kept it from using the alc883 since the revision_id is 0x100103). I did find though, that using the mbp3 option is what allowed sound to work. The same issues with the macbook pro aluminum apply here. the channel must be switched to 2ch to mute built-in speakers and to 6ch to unmute. There is another issue here though. the playback from the speakers is barely audible, i'm guessing it's similar to the issue with the right speaker on the macbook pro. I'm guessing there needs to be adjustments to the pins defined. My current alsa-info resides at http://www.alsa-project.org/db/?f=5bfaadf03f7134e9ef2c3a934fb4084d36ba2d93 If there's anything else I can do to help this along, let me know. I do have the macbook pro aluminum, so I can test there as well. -Micah

1 0

[PATCH] ASoC: Set parent for AC97 devices we register
by Mark Brown 01 Apr '09

01 Apr '09

Ensure that any AC97 devices that bind to the CODEC are below the ASoC device in the device tree so the suspend and resume code can figure out what order to handle them in. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6c62d4a..99712f6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -98,7 +98,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) int err; codec->ac97->dev.bus = &ac97_bus_type; - codec->ac97->dev.parent = NULL; + codec->ac97->dev.parent = codec->card->dev; codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", -- 1.6.2.1

1 0

[alsa-devel] [PATCH] Add configuration for HP EliteBook 2530p
by Matthew Garrett 01 Apr '09

01 Apr '09

The HP EliteBook 2530p needs the AD1884A_LAPTOP codec setup. Signed-off-by: Matthew Garrett <mjg(a)redhat.com> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 5bb48ee..945b029 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3924,6 +3924,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e1, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} -- Matthew Garrett | mjg59(a)srcf.ucam.org

1 1

[alsa-devel] Bug 2552 typo in ice1724.c
by Alan Horstmann 31 Mar '09

31 Mar '09

Hi, I noticed this trivial typo report on bugtrack, -no 2552 and thought it may as well be fixed, and the bug closed. Patch inline & attached. Not sure if it needs a sign-off? Alan ice1724.c-name-typo-fix.diff Fix trivial name string typo as reported in bug 2552 Signed-off-by: Alan Horstmann <gineera(a)aspect135.co.uk> --- alsa-kernel/pci/ice1712/ice1724-orig.c 2008-06-27 09:11:58.000000000 +0100 +++ alsa-kernel/pci/ice1712/ice1724.c 2009-01-05 11:11:33.000000000 +0000 @@ -1239,7 +1239,7 @@ if (ice->force_pdma4 || ice->force_rdma1) name = "ICE1724 Secondary"; else - name = "IEC1724 IEC958"; + name = "ICE1724 IEC958"; err = snd_pcm_new(ice->card, name, device, play, capt, &pcm); if (err < 0) return err;

1 1

[PATCH] ASoC: Don't defer resume work for AC97 codecs
by Mark Brown 31 Mar '09

31 Mar '09

AC97 devices may have other drivers hanging off them directly so need to have resumed when the resume function returns meaning that we can't defer the resume - complete it immediately for them. Non-AC97 devices should not have other drivers hanging directly off the ASoC devices. We only really need the deferral for non-AC97 devices - it's there since some I2C buses are very slow and non-AC97 codecs often have large numbers of registers to restore and require delays to bring the codec up cleanly leading to a substantial impact on overall resume time. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: stable(a)kernel.org --- sound/soc/soc-core.c | 18 ++++++++++++++---- 1 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6e710f7..6c62d4a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -767,11 +767,21 @@ static int soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; - dev_dbg(socdev->dev, "scheduling resume work\n"); - - if (!schedule_work(&card->deferred_resume_work)) - dev_err(socdev->dev, "resume work item may be lost\n"); + /* AC97 devices might have other drivers hanging off them so + * need to resume immediately. Other drivers don't have that + * problem and may take a substantial amount of time to resume + * due to I/O costs and anti-pop so handle them out of line. + */ + if (cpu_dai->ac97_control) { + dev_dbg(socdev->dev, "Resuming AC97 immediately\n"); + soc_resume_deferred(&card->deferred_resume_work); + } else { + dev_dbg(socdev->dev, "Scheduling resume work\n"); + if (!schedule_work(&card->deferred_resume_work)) + dev_err(socdev->dev, "resume work item may be lost\n"); + } return 0; } -- 1.6.2.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: TWL4030 updates
by Peter Ujfalusi 31 Mar '09

31 Mar '09

Hello, Two small updates for the TWL4030 codec driver: 1. Add support for 96KHz sample rate playback (96KHz capture is not supported) 2. Constrains to protect the already running stream from accidental reconfiguration --- Peter Ujfalusi (2): ASoC: TWL4030: 96KHz playback support ASoC: TWL4030: Add constrains for second stream sound/soc/codecs/twl4030.c | 56 +++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 55 insertions(+), 1 deletions(-)

2 4

[alsa-devel] Non-interleaved multi-channel access, what's required? Oxygen driver...
by olly＠funkster.org 30 Mar '09

30 Mar '09

Hi all, Hope this is the right place for this. I'm working on an app that requires non-interleaved access to 6 output channels, but alas the driver for the card that I'm using (Asus Xonar D1, AV100 / C-Media Oxygen chipset) doesn't support the noninterleaved access type, giving "incorrect access type" in response to this command: if ((err = snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED)) < 0) { fprintf(stderr, "cannot set access type (%s)\n", snd_strerror (err)); return false; } In a bout of optimism, I simply added the SNDRV_PCM_INFO_NONINTERLEAVED flag to the .info part of the snd_pcm_hardware struct that's defined for the card in oxygen_pcm.c and recompiled the kernel, not expecting it to really work. To my surprise, it does work! But, only for the "default" card and with two channels. If I try to use any of the multi-channel devices (e.g. front, surround51) then each channel appears to alternate round the buffers for all the channels. So... how do I find out if this is a limitation of the hardware? Can I do anything to the multi-channel part of the driver to make non-interleaved access work correctly? Or is there some clever use of the .asoundrc that can make it do what I need to? Many thanks in advance for any help, -- Oliver Stephenson http://www.funkster.org/ _______________________________________ Sent using funkster.org webmail service

1 0

[alsa-devel] snd_device_name_hint destroys the config
by Henrik Grunell 30 Mar '09

30 Mar '09

Hi, After I've called snd_device_name_hint the configuration seems destroyed. I can no longer access (open) e.g. "default" and "pulse" pcms. As a test, I took the pcm_list function in aplay and put it in my application. The second (and following) call(s) shows that devices are missing compared to the first call. I use Ubuntu 8.10 (1.0.17) and I see the same behavior on Fedora 7 (1.0.14). Is this a known issue? I haven't found anything searching on this. Thanks, Henrik ________________________________ CONFIDENTIALITY NOTE: This e-mail message, including any attachment(s), contains information that may be confidential, protected by the attorney-client or other legal privileges, and/or proprietary non-public information. If you are not an intended recipient of this message or an authorized assistant to an intended recipient, please do not read, copy, use or disclose this communication and notify the sender immediately. It should be noted that any review, retransmission, dissemination or other use of, or taking action or reliance upon, this information by persons or entities other than the intended recipient is prohibited.

1 0

[alsa-devel] hda-intel failure to install drivers
by Dmitri Seletski 30 Mar '09

30 Mar '09

Hello. Please check out txt attachment. Reason I have decided to go with source code from alsa web page - i have issues with microphone. Basically i can record, but it only records white noise. However, i didn't come across any issues with playing sounds. Even DS 5.1 was working! Anyway, alsalog describes what i did do generally. alsaconf.log describes what has happened when i was using alsaconf 2>>alasaconf.log Any ideas anyone? For time being i shall use kernel modules... Regards Dmitri cd /usr/src/linux make mount /boot cp arch/x86_64/boot/bzImage /boot/vanilla.2.6.28.9 *now i have fresh kernel and all necessary modules, i have soundcore.ko as one of modules, no other alsa module is compiled in or as module /dev/snd is already created * reboot ./configure --with-cards=hda-intel --with-isapnp=no make make install-modules modprobe snd-hda-intel WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hda_codec (/lib/modules/2.6.28.9/kernel/sound/pci/hda/snd-hda-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_hda_intel (/lib/modules/2.6.28.9/kernel/sound/pci/hda/snd-hda-intel.ko): Unknown symbol in module, or unknown parameter (see dmesg) *relevant part of dmesg* [ 26.548089] i2c-adapter i2c-8: adapter [NVIDIA i2c adapter ] registered [ 26.548099] i2c-adapter i2c-9: adapter [NVIDIA i2c adapter ] registered [ 88.000269] Clocksource tsc unstable (delta = -277240031 ns) [ 301.259242] snd: Unknown symbol unregister_sound_special [ 301.260119] snd: Unknown symbol register_sound_special_device [ 301.262749] snd_timer: Unknown symbol snd_info_register [ 301.262896] snd_timer: Unknown symbol snd_info_create_module_entry [ 301.263059] snd_timer: Unknown symbol snd_info_free_entry [ 301.263722] snd_timer: Unknown symbol snd_verbose_printk [ 301.263899] snd_timer: Unknown symbol snd_iprintf [ 301.264124] snd_timer: Unknown symbol snd_ecards_limit [ 301.264623] snd_timer: Unknown symbol snd_oss_info_register [ 301.264768] snd_timer: Unknown symbol snd_unregister_device [ 301.264972] snd_timer: Unknown symbol snd_device_new [ 301.265616] snd_timer: Unknown symbol snd_register_device_for_dev [ 301.266908] snd_pcm: Unknown symbol snd_info_register [ 301.267054] snd_pcm: Unknown symbol snd_info_create_module_entry [ 301.267197] snd_pcm: Unknown symbol snd_ctl_unregister_ioctl_compat [ 301.267981] snd_pcm: Unknown symbol snd_timer_notify [ 301.268160] snd_pcm: Unknown symbol snd_timer_interrupt [ 301.268545] snd_pcm: Unknown symbol snd_info_free_entry [ 301.268689] snd_pcm: Unknown symbol snd_add_device_sysfs_file [ 301.268865] snd_pcm: Unknown symbol snd_info_get_str [ 301.269605] snd_pcm: Unknown symbol snd_verbose_printk [ 301.269924] snd_pcm: Unknown symbol snd_ctl_register_ioctl [ 301.270067] snd_pcm: Unknown symbol snd_card_file_add [ 301.270598] snd_pcm: Unknown symbol snd_iprintf [ 301.270831] snd_pcm: Unknown symbol snd_major [ 301.271211] snd_pcm: Unknown symbol snd_unregister_device [ 301.271800] snd_pcm: Unknown symbol snd_timer_new [ 301.271944] snd_pcm: Unknown symbol snd_device_new [ 301.272228] snd_pcm: Unknown symbol snd_ctl_unregister_ioctl [ 301.272877] snd_pcm: Unknown symbol snd_lookup_minor_data [ 301.273021] snd_pcm: Unknown symbol snd_ctl_register_ioctl_compat [ 301.273215] snd_pcm: Unknown symbol snd_info_create_card_entry [ 301.273706] snd_pcm: Unknown symbol snd_power_wait [ 301.273869] snd_pcm: Unknown symbol snd_device_free [ 301.274150] snd_pcm: Unknown symbol snd_card_file_remove [ 301.274618] snd_pcm: Unknown symbol snd_register_device_for_dev [ 301.274920] snd_pcm: Unknown symbol snd_device_register [ 301.275073] snd_pcm: Unknown symbol snd_info_get_line [ 301.275982] snd_hwdep: Unknown symbol snd_info_register [ 301.276127] snd_hwdep: Unknown symbol snd_info_create_module_entry [ 301.276521] snd_hwdep: Unknown symbol snd_ctl_unregister_ioctl_compat [ 301.276667] snd_hwdep: Unknown symbol snd_info_free_entry [ 301.276835] snd_hwdep: Unknown symbol snd_unregister_oss_device [ 301.276993] snd_hwdep: Unknown symbol snd_verbose_printk [ 301.277137] snd_hwdep: Unknown symbol snd_register_oss_device [ 301.277792] snd_hwdep: Unknown symbol snd_ctl_register_ioctl [ 301.277937] snd_hwdep: Unknown symbol snd_card_file_add [ 301.278109] snd_hwdep: Unknown symbol snd_iprintf [ 301.278579] snd_hwdep: Unknown symbol snd_major [ 301.278769] snd_hwdep: Unknown symbol snd_unregister_device [ 301.278962] snd_hwdep: Unknown symbol snd_device_new [ 301.279146] snd_hwdep: Unknown symbol snd_ctl_unregister_ioctl [ 301.279732] snd_hwdep: Unknown symbol snd_lookup_oss_minor_data [ 301.279887] snd_hwdep: Unknown symbol snd_lookup_minor_data [ 301.280032] snd_hwdep: Unknown symbol snd_ctl_register_ioctl_compat [ 301.280189] snd_hwdep: Unknown symbol snd_card_file_remove [ 301.280748] snd_hwdep: Unknown symbol snd_register_device_for_dev [ 301.281899] snd_hda_codec: Unknown symbol snd_ctl_add [ 301.282114] snd_hda_codec: Unknown symbol snd_card_proc_new [ 301.282530] snd_hda_codec: Unknown symbol snd_add_device_sysfs_file [ 301.282758] snd_hda_codec: Unknown symbol snd_ctl_remove [ 301.282902] snd_hda_codec: Unknown symbol snd_ctl_find_id [ 301.283127] snd_hda_codec: Unknown symbol snd_verbose_printk [ 301.283819] snd_hda_codec: Unknown symbol snd_ctl_new1 [ 301.284027] snd_hda_codec: Unknown symbol snd_component_add [ 301.284173] snd_hda_codec: Unknown symbol snd_ctl_make_virtual_master [ 301.284678] snd_hda_codec: Unknown symbol snd_iprintf [ 301.284837] snd_hda_codec: Unknown symbol snd_ctl_boolean_mono_info [ 301.285165] snd_hda_codec: Unknown symbol snd_hwdep_new [ 301.285861] snd_hda_codec: Unknown symbol snd_device_new [ 301.286044] snd_hda_codec: Unknown symbol _snd_ctl_add_slave [ 301.286576] snd_hda_codec: Unknown symbol snd_pci_quirk_lookup [ 301.286860] snd_hda_codec: Unknown symbol snd_device_free [ 301.287004] snd_hda_codec: Unknown symbol snd_pcm_hw_constraint_step [ 301.287223] snd_hda_codec: Unknown symbol snd_pcm_format_width [ 301.288530] snd_hda_intel: Unknown symbol snd_pcm_new [ 301.288674] snd_hda_intel: Unknown symbol snd_pcm_limit_hw_rates [ 301.288826] snd_hda_intel: Unknown symbol snd_card_register [ 301.288969] snd_hda_intel: Unknown symbol snd_card_free [ 301.289112] snd_hda_intel: Unknown symbol snd_pcm_lib_preallocate_pages_for_all [ 301.289830] snd_hda_intel: Unknown symbol snd_hda_bus_new [ 301.290176] snd_hda_intel: Unknown symbol snd_hda_build_pcms [ 301.290562] snd_hda_intel: Unknown symbol snd_pcm_set_sync [ 301.290750] snd_hda_intel: Unknown symbol snd_verbose_printk [ 301.290894] snd_hda_intel: Unknown symbol snd_hda_codec_new [ 301.291648] snd_hda_intel: Unknown symbol snd_hda_queue_unsol_event [ 301.291837] snd_hda_intel: Unknown symbol snd_hda_power_up [ 301.291997] snd_hda_intel: Unknown symbol snd_pcm_sgbuf_get_chunk_size [ 301.292162] snd_hda_intel: Unknown symbol snd_pcm_lib_malloc_pages [ 301.292788] snd_hda_intel: Unknown symbol snd_pcm_lib_ioctl [ 301.292943] snd_hda_intel: Unknown symbol snd_pcm_lib_free_pages [ 301.293097] snd_hda_intel: Unknown symbol snd_hda_power_down [ 301.293241] snd_hda_intel: Unknown symbol snd_hda_calc_stream_format [ 301.293819] snd_hda_intel: Unknown symbol snd_pcm_set_ops [ 301.294033] snd_hda_intel: Unknown symbol snd_hda_suspend [ 301.294219] snd_hda_intel: Unknown symbol snd_device_new [ 301.294686] snd_hda_intel: Unknown symbol snd_pcm_sgbuf_ops_page [ 301.294887] snd_hda_intel: Unknown symbol snd_pcm_suspend_all [ 301.295036] snd_hda_intel: Unknown symbol snd_card_disconnect [ 301.295180] snd_hda_intel: Unknown symbol snd_hda_resume [ 301.295744] snd_hda_intel: Unknown symbol snd_pcm_hw_constraint_integer [ 301.295933] snd_hda_intel: Unknown symbol snd_hda_build_controls [ 301.296559] snd_hda_intel: Unknown symbol snd_pci_quirk_lookup [ 301.296901] snd_hda_intel: Unknown symbol snd_card_create [ 301.297046] snd_hda_intel: Unknown symbol snd_pcm_period_elapsed [ 301.297190] snd_hda_intel: Unknown symbol snd_pcm_hw_constraint_step *relevant part of "lspci -vv"* 00:0e.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2) Subsystem: ASUSTeK Computer Inc. Device 81f2 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin B routed to IRQ 5 Region 0: Memory at fe024000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Masking: 00000000 Pending: 00000000 Capabilities: [6c] HyperTransport: MSI Mapping Enable+ Fixed+ Kernel modules: snd-hda-intel WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_ad1889 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-ad1889.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_ali5451 (/lib/modules/2.6.28.9/kernel/sound/pci/ali5451/snd-ali5451.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_als300 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-als300.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_opl3_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/opl3/snd-opl3-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_sb_common (/lib/modules/2.6.28.9/kernel/sound/isa/sb/snd-sb-common.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_als4000 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-als4000.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_asihpi (/lib/modules/2.6.28.9/kernel/sound/pci/asihpi/snd-asihpi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_atiixp (/lib/modules/2.6.28.9/kernel/sound/pci/snd-atiixp.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_atiixp_modem (/lib/modules/2.6.28.9/kernel/sound/pci/snd-atiixp-modem.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_au8810 (/lib/modules/2.6.28.9/kernel/sound/pci/au88x0/snd-au8810.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_au8820 (/lib/modules/2.6.28.9/kernel/sound/pci/au88x0/snd-au8820.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_au8830 (/lib/modules/2.6.28.9/kernel/sound/pci/au88x0/snd-au8830.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_aw2 (/lib/modules/2.6.28.9/kernel/sound/pci/aw2/snd-aw2.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_opl3_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/opl3/snd-opl3-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_azt3328 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-azt3328.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_bt87x (/lib/modules/2.6.28.9/kernel/sound/pci/snd-bt87x.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_ca0106 (/lib/modules/2.6.28.9/kernel/sound/pci/ca0106/snd-ca0106.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_opl3_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/opl3/snd-opl3-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_cmipci (/lib/modules/2.6.28.9/kernel/sound/pci/snd-cmipci.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_opl3_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/opl3/snd-opl3-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_cs4281 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-cs4281.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_cs46xx (/lib/modules/2.6.28.9/kernel/sound/pci/cs46xx/snd-cs46xx.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_sb_common (/lib/modules/2.6.28.9/kernel/sound/isa/sb/snd-sb-common.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_sb16_dsp (/lib/modules/2.6.28.9/kernel/sound/isa/sb/snd-sb16-dsp.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_cs5530 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-cs5530.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_darla20 (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-darla20.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_darla24 (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-darla24.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_echo3g (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-echo3g.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_util_mem (/lib/modules/2.6.28.9/kernel/sound/synth/snd-util-mem.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_emu10k1 (/lib/modules/2.6.28.9/kernel/sound/pci/emu10k1/snd-emu10k1.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_emu10k1x (/lib/modules/2.6.28.9/kernel/sound/pci/emu10k1/snd-emu10k1x.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_ens1370 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-ens1370.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_ens1371 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-ens1371.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_opl3_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/opl3/snd-opl3-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_es1938 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-es1938.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_es1968 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-es1968.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_opl3_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/opl3/snd-opl3-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_fm801 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-fm801.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_gina20 (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-gina20.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_gina24 (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-gina24.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hda_codec (/lib/modules/2.6.28.9/kernel/sound/pci/hda/snd-hda-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_hda_intel (/lib/modules/2.6.28.9/kernel/sound/pci/hda/snd-hda-intel.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_hdsp (/lib/modules/2.6.28.9/kernel/sound/pci/rme9652/snd-hdsp.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_hdspm (/lib/modules/2.6.28.9/kernel/sound/pci/rme9652/snd-hdspm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_oxygen_lib (/lib/modules/2.6.28.9/kernel/sound/pci/oxygen/snd-oxygen-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_hifier (/lib/modules/2.6.28.9/kernel/sound/pci/oxygen/snd-hifier.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_i2c (/lib/modules/2.6.28.9/kernel/sound/i2c/snd-i2c.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_cs8427 (/lib/modules/2.6.28.9/kernel/sound/i2c/snd-cs8427.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ak4xxx_adda (/lib/modules/2.6.28.9/kernel/sound/i2c/other/snd-ak4xxx-adda.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ice17xx_ak4xxx (/lib/modules/2.6.28.9/kernel/sound/pci/ice1712/snd-ice17xx-ak4xxx.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_ice1712 (/lib/modules/2.6.28.9/kernel/sound/pci/ice1712/snd-ice1712.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_i2c (/lib/modules/2.6.28.9/kernel/sound/i2c/snd-i2c.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pt2258 (/lib/modules/2.6.28.9/kernel/sound/i2c/other/snd-pt2258.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ak4114 (/lib/modules/2.6.28.9/kernel/sound/i2c/other/snd-ak4114.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ak4xxx_adda (/lib/modules/2.6.28.9/kernel/sound/i2c/other/snd-ak4xxx-adda.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ice17xx_ak4xxx (/lib/modules/2.6.28.9/kernel/sound/pci/ice1712/snd-ice17xx-ak4xxx.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_ice1724 (/lib/modules/2.6.28.9/kernel/sound/pci/ice1712/snd-ice1724.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_indigo (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-indigo.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_indigodj (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-indigodj.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_indigoio (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-indigoio.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_intel8x0 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-intel8x0.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_intel8x0m (/lib/modules/2.6.28.9/kernel/sound/pci/snd-intel8x0m.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_korg1212 (/lib/modules/2.6.28.9/kernel/sound/pci/korg1212/snd-korg1212.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_layla20 (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-layla20.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_layla24 (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-layla24.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_maestro3 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-maestro3.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_mia (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-mia.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_mixart (/lib/modules/2.6.28.9/kernel/sound/pci/mixart/snd-mixart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_mona (/lib/modules/2.6.28.9/kernel/sound/pci/echoaudio/snd-mona.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_nm256 (/lib/modules/2.6.28.9/kernel/sound/pci/nm256/snd-nm256.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_oxygen_lib (/lib/modules/2.6.28.9/kernel/sound/pci/oxygen/snd-oxygen-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_oxygen (/lib/modules/2.6.28.9/kernel/sound/pci/oxygen/snd-oxygen.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_pcxhr (/lib/modules/2.6.28.9/kernel/sound/pci/pcxhr/snd-pcxhr.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_pdplus (/lib/modules/2.6.28.9/kernel/sound/pci/pdplus/snd-pdplus.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_opl3_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/opl3/snd-opl3-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_riptide (/lib/modules/2.6.28.9/kernel/sound/pci/riptide/snd-riptide.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_rme32 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-rme32.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_rme96 (/lib/modules/2.6.28.9/kernel/sound/pci/snd-rme96.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_rme9652 (/lib/modules/2.6.28.9/kernel/sound/pci/rme9652/snd-rme9652.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_opl3_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/opl3/snd-opl3-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_sonicvibes (/lib/modules/2.6.28.9/kernel/sound/pci/snd-sonicvibes.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_util_mem (/lib/modules/2.6.28.9/kernel/sound/synth/snd-util-mem.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_trident (/lib/modules/2.6.28.9/kernel/sound/pci/trident/snd-trident.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_via82xx (/lib/modules/2.6.28.9/kernel/sound/pci/snd-via82xx.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_via82xx_modem (/lib/modules/2.6.28.9/kernel/sound/pci/snd-via82xx-modem.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_oxygen_lib (/lib/modules/2.6.28.9/kernel/sound/pci/oxygen/snd-oxygen-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_virtuoso (/lib/modules/2.6.28.9/kernel/sound/pci/oxygen/snd-virtuoso.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_vx_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/vx/snd-vx-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_vx222 (/lib/modules/2.6.28.9/kernel/sound/pci/vx222/snd-vx222.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_rawmidi (/lib/modules/2.6.28.9/kernel/sound/acore/snd-rawmidi.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mpu401_uart (/lib/modules/2.6.28.9/kernel/sound/drivers/mpu401/snd-mpu401-uart.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_hwdep (/lib/modules/2.6.28.9/kernel/sound/acore/snd-hwdep.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_opl3_lib (/lib/modules/2.6.28.9/kernel/sound/drivers/opl3/snd-opl3-lib.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_ac97_codec (/lib/modules/2.6.28.9/kernel/sound/pci/ac97/snd-ac97-codec.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_ymfpci (/lib/modules/2.6.28.9/kernel/sound/pci/ymfpci/snd-ymfpci.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_midi_event (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-midi-event.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_seq_oss (/lib/modules/2.6.28.9/kernel/sound/acore/seq/oss/snd-seq-oss.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_mixer_oss (/lib/modules/2.6.28.9/kernel/sound/acore/oss/snd-mixer-oss.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_pcm (/lib/modules/2.6.28.9/kernel/sound/acore/snd-pcm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_pcm_oss (/lib/modules/2.6.28.9/kernel/sound/acore/oss/snd-pcm-oss.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_mixer_oss (/lib/modules/2.6.28.9/kernel/sound/acore/oss/snd-mixer-oss.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd (/lib/modules/2.6.28.9/kernel/sound/acore/snd.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_seq_device (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq-device.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting snd_timer (/lib/modules/2.6.28.9/kernel/sound/acore/snd-timer.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting snd_seq (/lib/modules/2.6.28.9/kernel/sound/acore/seq/snd-seq.ko): Unknown symbol in module, or unknown parameter (see dmesg) amixer: Mixer attach default error: No such file or directory /usr/sbin/alsactl: save_state:1501: No soundcards found...

2 1

[alsa-devel] [PATCH] alsa-info.sh: add command-line flag to print alsa info to stdout
by Luke Yelavich 30 Mar '09

30 Mar '09

Add the --stdout command-line flag to allow displaying the collected alsa information directly to standard output. This is useful for bug information collecting systems, such as Ubuntu's apport. Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- utils/alsa-info.sh | 34 +++++++++++++++++++++++++++++----- 1 files changed, 29 insertions(+), 5 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index b62c4d5..8340019 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -250,6 +250,12 @@ case "$1" in REPEAT="" shift ;; + --stdout) + DIALOG="" + NOUPLOAD="yes" + QUESTION="no" + TOSTDOUT="yes" + ;; esac done @@ -596,6 +602,17 @@ then fi fi ;; + --stdout) + NOUPLOAD="yes" + withdevices + withconfigs + withaplay + withamixer + withalsactl + withlsmod + cat $FILE + rm $FILE + ;; --about) echo "Written/Tested by the following users of #alsa on irc.freenode.net:" echo "" @@ -604,6 +621,7 @@ then echo " gnubien - Various script ideas / Testing" echo " GrueMaster - HDA Intel specific items / Testing" echo " olegfink - Script update function" + echo " TheMuso - display to stdout functionality" exit 0 ;; *) @@ -621,6 +639,8 @@ then echo " --no-upload (do not upload contents to remote server)" echo " --pastebin (use http://pastebin.ca) as remote server" echo " instead www.alsa-project.org" + echo " --stdout (print alsa information to standard output" + echo " instead of a file)" echo " --about (show some information about the script)" echo " --debug (will run the script as normal, but will not" echo " delete $FILE)" @@ -637,7 +657,9 @@ fi if [ -n "$NOUPLOAD" ]; then - mv $FILE $NFILE || exit 1 + if [ -z "$TOSTDOUT" ]; then + mv $FILE $NFILE || exit 1 + fi if [[ -n $DIALOG ]] then @@ -655,10 +677,12 @@ if [ -n "$NOUPLOAD" ]; then echo "Your ALSA information can be seen by looking in $NFILE" echo "" else - echo "You requested that your information was NOT automatically uploaded to the $WWWSERVICE" - echo "" - echo "Your ALSA information can be seen by looking in $NFILE" - echo "" + if [ -z "$TOSTDOUT" ]; then + echo "You requested that your information was NOT automatically uploaded to the $WWWSERVICE" + echo "" + echo "Your ALSA information can be seen by looking in $NFILE" + echo "" + fi fi fi -- 1.6.0.4

1 1

[alsa-devel] [PATCH] Staging: line6: convert to snd_card_create()
by Alexander Beregalov 29 Mar '09

29 Mar '09

Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- drivers/staging/line6/audio.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) diff --git a/drivers/staging/line6/audio.c b/drivers/staging/line6/audio.c index 3aa9468..e2ac8d6 100644 --- a/drivers/staging/line6/audio.c +++ b/drivers/staging/line6/audio.c @@ -27,11 +27,12 @@ int line6_init_audio(struct usb_line6 *line6) { static int dev; struct snd_card *card; + int err; - card = snd_card_new(line6_index[dev], line6_id[dev], THIS_MODULE, 0); - - if (card == NULL) - return -ENOMEM; + err = snd_card_create(line6_index[dev], line6_id[dev], THIS_MODULE, 0, + &card); + if (err < 0) + return err; line6->card = card;

1 0

[alsa-devel] Attempts to get a USB device operating.
by Tom Watson 29 Mar '09

29 Mar '09

I've got a (real) cheap usb sound "card" that partially works with ALSA. I can get the microphone input to respond by fiddling with some of the switches, but I have been unable to ge the output functions to work (produce output) at all. The "indicating" (power) light flashes, but I get no sound. Another thing that is weird is that while the litle box has SPDIF connections (both in and out) I don't see them on the mixer panel. Any clues would be appreciated. I have done the upload using 'alsa-info' with the defaults. Some brief details: lsusb yields: Bus 001 Device 034: ID 0d8c:0102 C-Media Electronics, Inc. The output mixer panel shows two outputs (speakers), The input mixer shows three inputs (2xPCM, 1 Mic). The switches mixer shows a PCM capture source and a single PCM switch. Are there any experiments I can do to help out in the investigation? I'm only using aplay/arecord in doing this test, just to make things 'easy' to diagnose. Thanks for the help. -- Tom Watson tsw(a)johana.com

1 0

[alsa-devel] [PATCH] snd-usb-caiaq: clean up header includes
by Daniel Mack 28 Mar '09

28 Mar '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/caiaq-audio.c | 8 +------- sound/usb/caiaq/caiaq-control.c | 6 +----- sound/usb/caiaq/caiaq-device.c | 13 +++---------- sound/usb/caiaq/caiaq-input.c | 6 ------ sound/usb/caiaq/caiaq-midi.c | 9 +-------- 5 files changed, 6 insertions(+), 36 deletions(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index 08d51e0..cf37331 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -16,17 +16,11 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/spinlock.h> #include <linux/init.h> -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/interrupt.h> #include <linux/usb.h> -#include <linux/spinlock.h> #include <sound/core.h> -#include <sound/initval.h> #include <sound/pcm.h> -#include <sound/rawmidi.h> -#include <linux/input.h> #include "caiaq-device.h" #include "caiaq-audio.h" diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index 77f60b4..52685cf 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -18,14 +18,10 @@ */ #include <linux/init.h> -#include <linux/interrupt.h> #include <linux/usb.h> +#include <sound/control.h> #include <sound/core.h> -#include <sound/initval.h> #include <sound/pcm.h> -#include <sound/rawmidi.h> -#include <sound/control.h> -#include <linux/input.h> #include "caiaq-device.h" #include "caiaq-control.h" diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index cf573a9..89f8b68 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -19,27 +19,20 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include <linux/init.h> -#include <linux/module.h> #include <linux/moduleparam.h> #include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/init.h> #include <linux/usb.h> -#include <linux/input.h> -#include <linux/spinlock.h> -#include <sound/core.h> #include <sound/initval.h> +#include <sound/core.h> #include <sound/pcm.h> -#include <sound/rawmidi.h> -#include <sound/control.h> #include "caiaq-device.h" #include "caiaq-audio.h" #include "caiaq-midi.h" #include "caiaq-control.h" - -#ifdef CONFIG_SND_USB_CAIAQ_INPUT #include "caiaq-input.h" -#endif MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13"); diff --git a/sound/usb/caiaq/caiaq-input.c b/sound/usb/caiaq/caiaq-input.c index f743847..4451775 100644 --- a/sound/usb/caiaq/caiaq-input.c +++ b/sound/usb/caiaq/caiaq-input.c @@ -17,14 +17,8 @@ */ #include <linux/init.h> -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/input.h> #include <linux/usb.h> #include <linux/usb/input.h> -#include <linux/spinlock.h> -#include <sound/core.h> -#include <sound/rawmidi.h> #include <sound/pcm.h> #include "caiaq-device.h" #include "caiaq-input.h" diff --git a/sound/usb/caiaq/caiaq-midi.c b/sound/usb/caiaq/caiaq-midi.c index f19fd36..79424c1 100644 --- a/sound/usb/caiaq/caiaq-midi.c +++ b/sound/usb/caiaq/caiaq-midi.c @@ -16,21 +16,14 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include <linux/init.h> -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/interrupt.h> #include <linux/usb.h> -#include <linux/input.h> -#include <linux/spinlock.h> -#include <sound/core.h> #include <sound/rawmidi.h> +#include <sound/core.h> #include <sound/pcm.h> #include "caiaq-device.h" #include "caiaq-midi.h" - static int snd_usb_caiaq_midi_input_open(struct snd_rawmidi_substream *substream) { return 0; -- 1.6.2.1

1 0

[alsa-devel] Regression? Default card no longer shows up
by Susan Cragin 28 Mar '09

28 Mar '09

I have been compiling the development alsa-drivers because I use an X-fi card. They have worked great up until today. I use Ubuntu Studio. I installed it clean this morning, with default options. Then I purged pulseaudio (because pulseaudio does not work with wine) re-booted. compiled and installed alsa-driver re-booted set-default-card to Generic asoundconf list shows the following Names of available sound cards: Intel Generic (this of course is my X-fi card) tried to run Audacity. Audacity only recognizes the Intel card. Installed wine. Wine shows the following: alsa-driver MIDI Out Devices: Midi Through Port-0 MIDI In Devices: Midi Through Port-0 Mixer Devices: HDA Intel HDA-Audio Generic This may be a Ubuntu bug, but I wanted to run it by you first. Susan Cragin

1 0

[alsa-devel] [patch 1/2] alsa: ASoC driver for s6000 I2S interface
by Daniel Glöckner 27 Mar '09

27 Mar '09

This patch adds a driver for the I2S interface found on Stretch s6000 family processors. Signed-off-by: Daniel Glöckner <dg(a)emlix.com> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/s6000/Kconfig | 10 + sound/soc/s6000/Makefile | 6 + sound/soc/s6000/s6000-i2s.c | 421 +++++++++++++++++++++++++++++++++++ sound/soc/s6000/s6000-i2s.h | 167 ++++++++++++++ sound/soc/s6000/s6000-pcm.c | 518 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/s6000/s6000-pcm.h | 41 ++++ 8 files changed, 1165 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s6000/Kconfig create mode 100644 sound/soc/s6000/Makefile create mode 100644 sound/soc/s6000/s6000-i2s.c create mode 100644 sound/soc/s6000/s6000-i2s.h create mode 100644 sound/soc/s6000/s6000-pcm.c create mode 100644 sound/soc/s6000/s6000-pcm.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index ef025c6..56e5198 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -31,6 +31,7 @@ source "sound/soc/fsl/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" +source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 86a9b1f..6b33d80 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,4 +10,5 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ +obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig new file mode 100644 index 0000000..5612ed2 --- /dev/null +++ b/sound/soc/s6000/Kconfig @@ -0,0 +1,10 @@ +config SND_S6000_SOC + tristate "SoC Audio for the Stretch s6000 family" + depends on XTENSA_VARIANT_S6000 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + s6000 family chips. You will also need to select the platform + to support below. + +config SND_S6000_SOC_I2S + tristate diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile new file mode 100644 index 0000000..df15f87 --- /dev/null +++ b/sound/soc/s6000/Makefile @@ -0,0 +1,6 @@ +# s6000 Platform Support +snd-soc-s6000-objs := s6000-pcm.o +snd-soc-s6000-i2s-objs := s6000-i2s.o + +obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o +obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c new file mode 100644 index 0000000..719a3c4 --- /dev/null +++ b/sound/soc/s6000/s6000-i2s.c @@ -0,0 +1,421 @@ +/* + * ALSA SoC I2S Audio Layer for the Stretch S6000 family + * + * Author: Daniel Gloeckner, <dg(a)emlix.com> + * Copyright: (C) 2009 emlix GmbH <info(a)emlix.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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/interrupt.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "s6000-i2s.h" +#include "s6000-pcm.h" + +static void s6000_i2s_wait_disabled(struct s6000_i2s_dev *dev) +{ + int channel; + int n = 50; + for (channel = 0; channel < 2; channel++) { + while (--n >= 0) { + int v = s6_i2s_read_reg(dev, S6_I2S_ENABLE(channel)); + if ((v & S6_I2S_IS_ENABLED) + || !(v & (S6_I2S_DMA_ACTIVE | S6_I2S_IS_BUSY))) + break; + udelay(20); + } + } + if (n < 0) + printk(KERN_WARNING "s6000-i2s: timeout disabling interfaces"); +} + +static int s6000_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s6000_i2s_dev *dev = cpu_dai->private_data; + u32 w; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + w = S6_I2S_SCK_IN | S6_I2S_WS_IN; + break; + case SND_SOC_DAIFMT_CBS_CFM: + w = S6_I2S_SCK_OUT | S6_I2S_WS_IN; + break; + case SND_SOC_DAIFMT_CBM_CFS: + w = S6_I2S_SCK_IN | S6_I2S_WS_OUT; + break; + case SND_SOC_DAIFMT_CBS_CFS: + w = S6_I2S_SCK_OUT | S6_I2S_WS_OUT; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + w |= S6_I2S_LEFT_FIRST; + break; + case SND_SOC_DAIFMT_IB_NF: + w |= S6_I2S_RIGHT_FIRST; + break; + default: + return -EINVAL; + } + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(0), + S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(1), + S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); + + return 0; +} + +static int s6000_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct s6000_i2s_dev *dev = dai->private_data; + + if (!div || (div & 1) || div > (S6_I2S_DIV_MASK + 1) * 2) + return -EINVAL; + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(div_id), + S6_I2S_DIV_MASK, div / 2 - 1); + return 0; +} + +static int s6000_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct s6000_i2s_dev *dev = dai->private_data; + int interf; + u32 w = 0; + + if (dev->wide) + interf = 0; + else { + w |= (((params_channels(params) - 2) / 2) + << S6_I2S_CHANNELS_SHIFT) & S6_I2S_CHANNELS_MASK; + interf = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ? dev->channel_out : dev->channel_in; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + w |= S6_I2S_16BIT | S6_I2S_MEM_16BIT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + w |= S6_I2S_32BIT | S6_I2S_MEM_32BIT; + break; + default: + printk(KERN_WARNING "s6000-i2s: unsupported PCM format %x\n", + params_format(params)); + return -EINVAL; + } + + if (s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(interf)) + & S6_I2S_IS_ENABLED) { + printk(KERN_ERR "s6000-i2s: interface already enabled\n"); + return -EBUSY; + } + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(interf), + S6_I2S_CHANNELS_MASK|S6_I2S_MEM_MASK|S6_I2S_BITS_MASK, + w); + + return 0; +} + +static int __devinit s6000_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct s6000_i2s_dev *dev; + struct resource *scbmem, *sifmem, *region, *dma_in, *dma_out; + struct s6000_snd_platform_data *pdata; + u8 __iomem *mmio; + int ret; + + pdata = pdev->dev.platform_data; + + scbmem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!scbmem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_none; + } + + region = request_mem_region(scbmem->start, + scbmem->end - scbmem->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S SCB region already claimed\n"); + ret = -EBUSY; + goto err_release_none; + } + + mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1); + if (!mmio) { + dev_err(&pdev->dev, "can't ioremap SCB region\n"); + ret = -ENOMEM; + goto err_release_scb; + } + + sifmem = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!sifmem) { + dev_err(&pdev->dev, "no second mem resource?\n"); + ret = -ENODEV; + goto err_release_map; + } + + region = request_mem_region(sifmem->start, + sifmem->end - sifmem->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S SIF region already claimed\n"); + ret = -EBUSY; + goto err_release_map; + } + + dma_in = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dma_in) { + dev_err(&pdev->dev, "no dma resource?\n"); + ret = -ENODEV; + goto err_release_sif; + } + + region = request_mem_region(dma_in->start, + dma_in->end - dma_in->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S DMA region already claimed\n"); + ret = -EBUSY; + goto err_release_sif; + } + + dma_out = 0; + if (pdata->lines_out) { + if (pdata->lines_in) { + dma_out = platform_get_resource(pdev, IORESOURCE_DMA, + 1); + if (!dma_out) { + dev_err(&pdev->dev, + "no second dma resource?\n"); + ret = -ENODEV; + goto err_release_dma1; + } + + region = request_mem_region(dma_out->start, + dma_out->end + - dma_out->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, + "I2S DMA region already claimed\n"); + ret = -EBUSY; + goto err_release_dma1; + } + } else { + dma_out = dma_in; + dma_in = 0; + } + } + + dev = kzalloc(sizeof(struct s6000_i2s_dev), GFP_KERNEL); + if (!dev) { + ret = -ENOMEM; + goto err_release_dma2; + } + + dai->private_data = dev; + dai->dma_data = &dev->dma_params; + + dev->sifbase = sifmem->start; + dev->scbbase = mmio; + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, + S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN); + + s6000_i2s_stop_channel(dev, 0); + s6000_i2s_stop_channel(dev, 1); + s6000_i2s_wait_disabled(dev); + + dev->wide = pdata->wide; + dev->channel_in = pdata->channel_in; + dev->channel_out = pdata->channel_out; + dev->lines_in = pdata->lines_in; + dev->lines_out = pdata->lines_out; + + s6_i2s_write_reg(dev, S6_I2S_MODE, + dev->wide ? S6_I2S_WIDE : S6_I2S_DUAL); + + if (dev->wide) { + int i; + dev->channel_in = 0; + dev->channel_out = 1; + dai->capture.channels_min = 2 * dev->lines_in; + dai->capture.channels_max = dai->capture.channels_min; + dai->playback.channels_min = 2 * dev->lines_out; + dai->playback.channels_max = dai->playback.channels_min; + + for (i = 0; i < dev->lines_out; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_OUT); + + for (; i < S6_I2S_NUM_LINES - dev->lines_in; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), + S6_I2S_UNUSED); + + for (; i < S6_I2S_NUM_LINES; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_IN); + } else { + unsigned int cfg[2] = {S6_I2S_UNUSED, S6_I2S_UNUSED}; + + dev->lines_in = (dev->lines_in != 0); + dev->lines_out = (dev->lines_out != 0); + dai->capture.channels_min = 2 * dev->lines_in; + dai->capture.channels_max = 8 * dev->lines_in; + dai->playback.channels_min = 2 * dev->lines_out; + dai->playback.channels_max = 8 * dev->lines_out; + + cfg[dev->channel_in] = S6_I2S_IN; + cfg[dev->channel_out] = S6_I2S_OUT; + + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(0), cfg[0]); + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(1), cfg[1]); + } + + dev->dma_params.dma_in = dma_in ? dma_in->start : 0; + dev->dma_params.dma_out = dma_out ? dma_out->start : 0; + dev->dma_params.sif_in = dev->sifbase + (dev->channel_in ? + S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); + dev->dma_params.sif_out = dev->sifbase + (dev->channel_out ? + S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); + dev->dma_params.same_rate = pdata->same_rate | pdata->wide; + + dev->dma_params.irq = platform_get_irq(pdev, 0); + if (dev->dma_params.irq < 0) { + dev_err(&pdev->dev, "no irq resource?\n"); + ret = -ENODEV; + goto err_release_dev; + } + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, + S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN); + + return 0; + +err_release_dev: + kfree(dev); +err_release_dma2: + if (dma_out) + release_mem_region(dma_out->start, + dma_out->end - dma_out->start + 1); +err_release_dma1: + if (dma_in) + release_mem_region(dma_in->start, + dma_in->end - dma_in->start + 1); +err_release_sif: + release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1); +err_release_map: + iounmap(mmio); +err_release_scb: + release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1); +err_release_none: + return ret; +} + +static void __devexit s6000_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct s6000_i2s_dev *dev = dai->private_data; + struct resource *region; + void __iomem *mmio = dev->scbbase; + + s6000_i2s_stop_channel(dev, 0); + s6000_i2s_stop_channel(dev, 1); + kfree(dev); + dai->private_data = 0; + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); + + region = platform_get_resource(pdev, IORESOURCE_DMA, 0); + release_region(region->start, region->end - region->start + 1); + + if (dev->lines_in && dev->lines_out) { + region = platform_get_resource(pdev, IORESOURCE_DMA, 1); + release_region(region->start, region->end - region->start + 1); + } + + region = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_region(region->start, (region->end - region->start) + 1); + + iounmap(mmio); + region = platform_get_resource(pdev, IORESOURCE_IO, 0); + release_mem_region(region->start, (region->end - region->start) + 1); +} + +#define S6000_I2S_RATES (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT | \ + SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000) +#define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai s6000_i2s_dai = { + .name = "s6000-i2s", + .id = 0, + .probe = s6000_i2s_probe, + .remove = __devexit_p(s6000_i2s_remove), + .playback = { + .channels_min = 2, + .channels_max = 8, + .formats = S6000_I2S_FORMATS, + .rates = S6000_I2S_RATES, + .rate_min = 0, + .rate_max = 1562500, + }, + .capture = { + .channels_min = 2, + .channels_max = 8, + .formats = S6000_I2S_FORMATS, + .rates = S6000_I2S_RATES, + .rate_min = 0, + .rate_max = 1562500, + }, + .ops = { + .set_fmt = s6000_i2s_set_dai_fmt, + .set_clkdiv = s6000_i2s_set_clkdiv, + .hw_params = s6000_i2s_hw_params, + } +} +EXPORT_SYMBOL_GPL(s6000_i2s_dai); + +static int __init s6000_i2s_init(void) +{ + return snd_soc_register_dai(&s6000_i2s_dai); +} +module_init(s6000_i2s_init); + +static void __exit s6000_i2s_exit(void) +{ + snd_soc_unregister_dai(&s6000_i2s_dai); +} +module_exit(s6000_i2s_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-i2s.h b/sound/soc/s6000/s6000-i2s.h new file mode 100644 index 0000000..4918f95 --- /dev/null +++ b/sound/soc/s6000/s6000-i2s.h @@ -0,0 +1,167 @@ +/* + * ALSA SoC I2S Audio Layer for the Stretch s6000 family + * + * Author: Daniel Gloeckner, <dg(a)emlix.com> + * Copyright: (C) 2009 emlix GmbH <info(a)emlix.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 _S6000_I2S_H +#define _S6000_I2S_H + +#include <linux/io.h> +#include "s6000-pcm.h" + +extern struct snd_soc_dai s6000_i2s_dai; + +struct s6000_i2s_dev { + dma_addr_t sifbase; + u8 __iomem *scbbase; + int wide; + int channel_in; + int channel_out; + int lines_in; + int lines_out; + struct s6000_pcm_dma_params dma_params; +}; + +#define S6_I2S_INTERRUPT_STATUS 0x00 +#define S6_I2S_INT_OVERRUN 1 +#define S6_I2S_INT_UNDERRUN 2 +#define S6_I2S_INT_ALIGNMENT 4 +#define S6_I2S_INTERRUPT_ENABLE 0x04 +#define S6_I2S_INTERRUPT_RAW 0x08 +#define S6_I2S_INTERRUPT_CLEAR 0x0C +#define S6_I2S_INTERRUPT_SET 0x10 +#define S6_I2S_MODE 0x20 +#define S6_I2S_DUAL 0 +#define S6_I2S_WIDE 1 +#define S6_I2S_TX_DEFAULT 0x24 +#define S6_I2S_DATA_CFG(c) (0x40 + 0x10 * (c)) +#define S6_I2S_IN 0 +#define S6_I2S_OUT 1 +#define S6_I2S_UNUSED 2 +#define S6_I2S_INTERFACE_CFG(c) (0x44 + 0x10 * (c)) +#define S6_I2S_DIV_MASK 0x001fff +#define S6_I2S_16BIT 0x000000 +#define S6_I2S_20BIT 0x002000 +#define S6_I2S_24BIT 0x004000 +#define S6_I2S_32BIT 0x006000 +#define S6_I2S_BITS_MASK 0x006000 +#define S6_I2S_MEM_16BIT 0x000000 +#define S6_I2S_MEM_32BIT 0x008000 +#define S6_I2S_MEM_MASK 0x008000 +#define S6_I2S_CHANNELS_SHIFT 16 +#define S6_I2S_CHANNELS_MASK 0x030000 +#define S6_I2S_SCK_IN 0x000000 +#define S6_I2S_SCK_OUT 0x040000 +#define S6_I2S_SCK_DIR 0x040000 +#define S6_I2S_WS_IN 0x000000 +#define S6_I2S_WS_OUT 0x080000 +#define S6_I2S_WS_DIR 0x080000 +#define S6_I2S_LEFT_FIRST 0x000000 +#define S6_I2S_RIGHT_FIRST 0x100000 +#define S6_I2S_FIRST 0x100000 +#define S6_I2S_CUR_SCK 0x200000 +#define S6_I2S_CUR_WS 0x400000 +#define S6_I2S_ENABLE(c) (0x48 + 0x10 * (c)) +#define S6_I2S_DISABLE_IF 0x02 +#define S6_I2S_ENABLE_IF 0x03 +#define S6_I2S_IS_BUSY 0x04 +#define S6_I2S_DMA_ACTIVE 0x08 +#define S6_I2S_IS_ENABLED 0x10 + +#define S6_I2S_NUM_LINES 4 + +#define S6_I2S_SIF_PORT0 0x0000000 +#define S6_I2S_SIF_PORT1 0x0000080 /* docs say 0x0000010 */ + +static inline void s6_i2s_write_reg(struct s6000_i2s_dev *dev, int reg, u32 val) +{ + writel(val, dev->scbbase + reg); +} + +static inline u32 s6_i2s_read_reg(struct s6000_i2s_dev *dev, int reg) +{ + return readl(dev->scbbase + reg); +} + +static inline void s6_i2s_mod_reg(struct s6000_i2s_dev *dev, int reg, + u32 mask, u32 val) +{ + val ^= s6_i2s_read_reg(dev, reg) & ~mask; + s6_i2s_write_reg(dev, reg, val); +} + +static inline void s6000_i2s_start_channel(struct s6000_i2s_dev *dev, + int channel) +{ + int i, j, cur, prev; + + /* + * Wait for WCLK to toggle 5 times before enabling the channel + * s6000 Family Datasheet 3.6.4: + * "At least two cycles of WS must occur between commands + * to disable or enable the interface" + */ + j = 0; + prev = ~S6_I2S_CUR_WS; + for (i = 1000000; --i && j < 6; ) { + cur = s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(channel)) + & S6_I2S_CUR_WS; + if (prev != cur) { + prev = cur; + j++; + } + } + if (j < 6) + printk(KERN_WARNING "s6000-i2s: timeout waiting for WCLK\n"); + + s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_ENABLE_IF); +} + +static inline void s6000_i2s_stop_channel(struct s6000_i2s_dev *dev, + int channel) +{ + s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_DISABLE_IF); +} + +static inline void s6000_i2s_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data; + int channel; + + channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dev->channel_out : dev->channel_in; + + s6000_i2s_start_channel(dev, channel); +} + +static inline void s6000_i2s_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data; + int channel; + + channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dev->channel_out : dev->channel_in; + + s6000_i2s_stop_channel(dev, channel); +} + +static inline unsigned int s6000_i2s_int_sources(struct s6000_i2s_dev *dev) +{ + unsigned int pending; + pending = s6_i2s_read_reg(dev, S6_I2S_INTERRUPT_RAW); + pending &= S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN; + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, pending); + + return pending; +} +#endif diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c new file mode 100644 index 0000000..90b546c --- /dev/null +++ b/sound/soc/s6000/s6000-pcm.c @@ -0,0 +1,518 @@ +/* + * ALSA PCM interface for the Stetch s6000 family + * + * Author: Daniel Gloeckner, <dg(a)emlix.com> + * Copyright: (C) 2009 emlix GmbH <info(a)emlix.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. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <variant/dmac.h> + +#include "s6000-i2s.h" +#include "s6000-pcm.h" + +#define S6000_PCM_DEBUG 0 +#if S6000_PCM_DEBUG +#define DPRINTK(x...) printk(KERN_DEBUG x) +#else +#define DPRINTK(x...) +#endif + +#define S6_PCM_PREALLOCATE_SIZE (96 * 1024) +#define S6_PCM_PREALLOCATE_MAX (2048 * 1024) + +static struct snd_pcm_hardware s6000_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT | + SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000), + .rate_min = 0, + .rate_max = 1562500, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = 0x7ffffff0, + .period_bytes_min = 16, + .period_bytes_max = 0xfffff0, + .periods_min = 2, + .periods_max = 1024, /* no limit */ + .fifo_size = 0, +}; + +struct s6000_runtime_data { + spinlock_t lock; + int period; /* current DMA period */ +}; + +static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int channel; + unsigned int period_size; + unsigned int dma_offset; + dma_addr_t dma_pos; + dma_addr_t src, dst; + + period_size = snd_pcm_lib_period_bytes(substream); + dma_offset = prtd->period * period_size; + dma_pos = runtime->dma_addr + dma_offset; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = dma_pos; + dst = par->sif_out; + channel = par->dma_out; + } else { + src = par->sif_in; + dst = dma_pos; + channel = par->dma_in; + } + + if (!s6dmac_channel_enabled(DMA_MASK_DMAC(channel), + DMA_INDEX_CHNL(channel))) + return; + + if (s6dmac_fifo_full(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel))) { + printk(KERN_ERR "s6000-pcm: fifo full\n"); + return; + } + + BUG_ON(period_size & 15); + s6dmac_put_fifo(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel), + src, dst, period_size); + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +static irqreturn_t s6000_pcm_irq(int irq, void *data) +{ + struct snd_pcm *pcm = data; + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct snd_pcm_substream *playback, *capture; + struct s6000_runtime_data *prtd; + unsigned int i2s_errors; + int i, ret = IRQ_NONE; + u32 channel[2] = { + [SNDRV_PCM_STREAM_PLAYBACK] = params->dma_out, + [SNDRV_PCM_STREAM_CAPTURE] = params->dma_in + }; + + i2s_errors = s6000_i2s_int_sources(runtime->dai->cpu_dai->private_data); + playback = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + capture = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + + if (unlikely(i2s_errors & S6_I2S_INT_ALIGNMENT)) { + printk(KERN_ERR "s6000-i2s: Alignment Error\n"); + ret = IRQ_HANDLED; + } + + if (unlikely(i2s_errors & S6_I2S_INT_UNDERRUN) && + playback->runtime && + snd_pcm_running(playback)) { + printk(KERN_WARNING "s6000-i2s: Tx Underrun\n"); + s6000_i2s_start(playback); + + prtd = playback->runtime->private_data; + spin_lock(&prtd->lock); + s6000_pcm_enqueue_dma(playback); + /* a second descriptor will be queued below */ + spin_unlock(&prtd->lock); + + ret = IRQ_HANDLED; + } + + for (i = 0; i < 2; ++i) { + struct snd_pcm_substream *substream = pcm->streams[i].substream; + unsigned int pending; + + if (!channel[i]) + continue; + + pending = s6dmac_int_sources(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i])); + + if (pending & 1) { + ret = IRQ_HANDLED; + if (likely(substream->runtime && + snd_pcm_running(substream))) { + snd_pcm_period_elapsed(substream); + DPRINTK("period elapsed %x %x\n", + s6dmac_cur_src(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i])), + s6dmac_cur_dst(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i]))); + prtd = substream->runtime->private_data; + spin_lock(&prtd->lock); + s6000_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } + } + + if (unlikely(pending & ~7)) { + if (pending & (1 << 3)) + printk(KERN_WARNING + "s6000-pcm: DMA %x Underflow\n", + channel[i]); + if (pending & (1 << 4)) + printk(KERN_WARNING + "s6000-pcm: DMA %x Overflow\n", + channel[i]); + if (pending & 0x1e0) + printk(KERN_WARNING + "s6000-pcm: DMA %x Master Error " + "(mask %x)\n", + channel[i], pending >> 5); + + } + } + + if (unlikely(i2s_errors & S6_I2S_INT_OVERRUN) && + capture->runtime && + snd_pcm_running(capture)) { + printk(KERN_WARNING "s6000-i2s: Rx Overrun\n"); + + prtd = capture->runtime->private_data; + spin_lock(&prtd->lock); + s6000_pcm_enqueue_dma(capture); + spin_unlock(&prtd->lock); + + s6000_i2s_start(capture); + ret = IRQ_HANDLED; + } + + return ret; +} + +int s6000_pcm_start(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + unsigned long flags; + + spin_lock_irqsave(&prtd->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + s6dmac_enable_chan(DMA_MASK_DMAC(par->dma_out), + DMA_INDEX_CHNL(par->dma_out), + 1, 0, 1, 0, 0, 0, 0, 4, -1, 0, 0, 1); + else + s6dmac_enable_chan(DMA_MASK_DMAC(par->dma_in), + DMA_INDEX_CHNL(par->dma_in), + 1, 0, 0, 1, 0, 0, 0, 4, -1, 0, 0, 1); + + s6000_pcm_enqueue_dma(substream); + s6000_pcm_enqueue_dma(substream); + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +int s6000_pcm_stop(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + unsigned long flags; + u32 channel; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + channel = par->dma_out; + else + channel = par->dma_in; + + s6dmac_set_terminal_count(DMA_MASK_DMAC(channel), + DMA_INDEX_CHNL(channel), 0); + + spin_lock_irqsave(&prtd->lock, flags); + + s6dmac_disable_chan(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel)); + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + s6000_i2s_start(substream); + ret = s6000_pcm_start(substream); + } else { + ret = s6000_pcm_start(substream); + s6000_i2s_start(substream); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + s6000_i2s_stop(substream); + ret = s6000_pcm_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int s6000_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + + prtd->period = 0; + + return 0; +} + +static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + unsigned long flags; + unsigned int offset; + dma_addr_t count; + + spin_lock_irqsave(&prtd->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + count = s6dmac_cur_src(DMA_MASK_DMAC(par->dma_out), + DMA_INDEX_CHNL(par->dma_out)); + else + count = s6dmac_cur_dst(DMA_MASK_DMAC(par->dma_in), + DMA_INDEX_CHNL(par->dma_in)); + + count -= runtime->dma_addr; + + spin_unlock_irqrestore(&prtd->lock, flags); + + offset = bytes_to_frames(runtime, count); + if (unlikely(offset >= runtime->buffer_size)) + offset = 0; + + return offset; +} + +static int s6000_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware); + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 16); + if (ret < 0) + return ret; + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16); + if (ret < 0) + return ret; + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + if (par->same_rate) { + int rate; + spin_lock(&par->lock); /* needed? */ + rate = par->rate; + spin_unlock(&par->lock); + if (rate != -1) { + ret = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_RATE, + rate, rate); + if (ret < 0) + return ret; + } + } + + prtd = kzalloc(sizeof(struct s6000_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int s6000_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int ret; + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) { + printk(KERN_WARNING "s6000-pcm: allocation of memory failed\n"); + return ret; + } + + if (par->same_rate) { + spin_lock(&par->lock); + if (par->rate == -1 || + !(par->in_use & ~(1 << substream->stream))) { + par->rate = params_rate(hw_params); + par->in_use |= 1 << substream->stream; + } else if (params_rate(hw_params) != par->rate) { + snd_pcm_lib_free_pages(substream); + par->in_use &= ~(1 << substream->stream); + ret = -EBUSY; + } + spin_unlock(&par->lock); + } + return ret; +} + +static int s6000_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + + spin_lock(&par->lock); + par->in_use &= ~(1 << substream->stream); + if (!par->in_use) + par->rate = -1; + spin_unlock(&par->lock); + + return snd_pcm_lib_free_pages(substream); +} + +struct snd_pcm_ops s6000_pcm_ops = { + .open = s6000_pcm_open, + .close = s6000_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = s6000_pcm_hw_params, + .hw_free = s6000_pcm_hw_free, + .trigger = s6000_pcm_trigger, + .prepare = s6000_pcm_prepare, + .pointer = s6000_pcm_pointer, +}; + +static void s6000_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + + free_irq(params->irq, pcm); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static u64 s6000_pcm_dmamask = DMA_32BIT_MASK; + +static int s6000_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + int res; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s6000_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_32BIT_MASK; + + if (params->dma_in) { + s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), + DMA_INDEX_CHNL(params->dma_in)); + s6dmac_int_sources(DMA_MASK_DMAC(params->dma_in), + DMA_INDEX_CHNL(params->dma_in)); + } + + if (params->dma_out) { + s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_out), + DMA_INDEX_CHNL(params->dma_out)); + s6dmac_int_sources(DMA_MASK_DMAC(params->dma_out), + DMA_INDEX_CHNL(params->dma_out)); + } + + res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED, + s6000_soc_platform.name, pcm); + if (res) { + printk(KERN_ERR "s6000-pcm couldn't get IRQ\n"); + return res; + } + + res = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, + S6_PCM_PREALLOCATE_SIZE, + S6_PCM_PREALLOCATE_MAX); + if (res) + printk(KERN_WARNING "s6000-pcm: preallocation failed\n"); + + spin_lock_init(&params->lock); + params->in_use = 0; + params->rate = -1; + return 0; +} + +struct snd_soc_platform s6000_soc_platform = { + .name = "s6000-audio", + .pcm_ops = &s6000_pcm_ops, + .pcm_new = s6000_pcm_new, + .pcm_free = s6000_pcm_free, +}; +EXPORT_SYMBOL_GPL(s6000_soc_platform); + +static int __init s6000_pcm_init(void) +{ + return snd_soc_register_platform(&s6000_soc_platform); +} +module_init(s6000_pcm_init); + +static void __exit s6000_pcm_exit(void) +{ + snd_soc_unregister_platform(&s6000_soc_platform); +} +module_exit(s6000_pcm_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-pcm.h b/sound/soc/s6000/s6000-pcm.h new file mode 100644 index 0000000..8d7305f --- /dev/null +++ b/sound/soc/s6000/s6000-pcm.h @@ -0,0 +1,41 @@ +/* + * ALSA PCM interface for the Stretch s6000 family + * + * Author: Daniel Gloeckner, <dg(a)emlix.com> + * Copyright: (C) 2009 emlix GmbH <info(a)emlix.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 _S6000_PCM_H +#define _S6000_PCM_H + +struct s6000_pcm_dma_params { + dma_addr_t sif_in; + dma_addr_t sif_out; + u32 dma_in; + u32 dma_out; + int irq; + int same_rate; + + spinlock_t lock; + int in_use; + int rate; +}; + +struct s6000_snd_platform_data { + int lines_in; + int lines_out; + int channel_in; + int channel_out; + int wide; + int same_rate; +}; + +int s6000_pcm_start(struct snd_pcm_substream *substream); +int s6000_pcm_stop(struct snd_pcm_substream *substream); +extern struct snd_soc_platform s6000_soc_platform; + +#endif -- 1.6.2.107.ge47ee

2 9

[alsa-devel] xrun on sdp340 with small buffers
by Lopez Cruz, Misael 27 Mar '09

27 Mar '09

I use the sdp3430 machine driver for my board, which connects twl4030 codec and OMAP McBSP/DMA drivers. Everything works fine, I can play audio files (using aplay). However, I tried to force 'aplay' to use a different buffer size and period count, and it misbehaves. My experiment was setting buffer time to 10ms and periods to 4, and play a 8kHz sample file; this only generates underrun messages. Under that configuration, for 8kHz, it means a buffer size of 320B (16-bits, stereo) divided into 4 periods of 80B each. omap-pcm driver will report interrupts each 80B, as it uses a single DMA transfer for the buffer and generates interrupts each frame. I enabled the SND_PCM_XRUN_DEBUG option in the kernel and set xrun_debug proc-entry to 1. And this is what I get: # aplay audio_8000.wav --buffer-time=10000 --period-time=2500 Playing WAVE 'audio_8000.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo Unexpected hw_pointer value [1] (stream = 0, delta: -67, max jitter = 80): wrong interrupt acknowledge? Unexpected hw_pointer value [1] (stream = 0, delta: -6, max jitter = 80): wrong interrupt acknowledge? What I understand it reports is that when an interrupt happens, the position of the transfer queried (using a dma get_pos) is lower than the position it (at least) should be. Let's say if in the third period (80B/period) of the buffer I should get at least a position greater than 240, and the position returned by query is -67B behind. Is my understanding correct? If so, does that mean the issue is with the DMA mechanism? Don't know if it's worth to say that adding the xrun_debung only prints some debug messages and then playback works. But disabling the debug shows only underrun messages. -Misa

3 4

[alsa-devel] Closing alsa devices after forking off children
by Werner Van Belle 27 Mar '09

27 Mar '09

Hello, I'm currently trying to figure out why a close of the alsa device is not realized 'quickly'. That is it seems to take 2 to 3 seconds before I can reopen the device in anopther program. After observing this behavior somewhat I figured out that opening an alsa device and then forking off a process will keep the alsa device open until the child dies, even when the parent process closes the alsa device. Is this behavior known and if so how can I work around this ? I tried to use pthread_atfork to close the device in the forked off child, but that closes the device in the parent process as well, so I would rather want to unlink the device in such a way that the close call does not need to wait until all children have given up their handle. With kind regards, -- Dr. Werner Van Belle http://werner.yellowcouch.org/

1 0

[alsa-devel] [PATCH 2.6.30] ASoC: trim SSI sysfs statistics in Freescale MPC8610 sound drivers
by Timur Tabi 27 Mar '09

27 Mar '09

Optimize the display of SSI statistics in the Freescale MPC8610 sound driver to display the status count only of the interrupts that were actually enabled. Previously, it would display the counts of all SISR status bits, even those that were not enabled. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 79 ++++++++++++++++++++++++++++------------------- 1 files changed, 47 insertions(+), 32 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 72823a2..3711d84 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -60,6 +60,13 @@ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) #endif +/* SIER bitflag of interrupts to enable */ +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ + CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ + CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ + CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ + CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) + /** * fsl_ssi_private: per-SSI private data * @@ -140,7 +147,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) were interrupted for. We mask it with the Interrupt Enable register so that we only check for events that we're interested in. */ - sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier); + sisr = in_be32(&ssi->sisr) & SIER_FLAGS; if (sisr & CCSR_SSI_SISR_RFRC) { ssi_private->stats.rfrc++; @@ -324,12 +331,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ /* 4. Enable the interrupts and DMA requests */ - out_be32(&ssi->sier, - CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | - CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | - CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN); + out_be32(&ssi->sier, SIER_FLAGS); /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We @@ -590,39 +592,52 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .ops = &fsl_ssi_dai_ops, }; +/* Show the statistics of a flag only if its interrupt is enabled. The + * compiler will optimze this code to a no-op if the interrupt is not + * enabled. + */ +#define SIER_SHOW(flag, name) \ + do { \ + if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \ + length += sprintf(buf + length, #name "=%u\n", \ + ssi_private->stats.name); \ + } while (0) + + /** * fsl_sysfs_ssi_show: display SSI statistics * - * Display the statistics for the current SSI device. + * Display the statistics for the current SSI device. To avoid confusion, + * we only show those counts that are enabled. */ static ssize_t fsl_sysfs_ssi_show(struct device *dev, struct device_attribute *attr, char *buf) { struct fsl_ssi_private *ssi_private = - container_of(attr, struct fsl_ssi_private, dev_attr); - ssize_t length; - - length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc); - length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc); - length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau); - length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu); - length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt); - length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1); - length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0); - length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1); - length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0); - length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1); - length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0); - length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1); - length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0); - length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs); - length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs); - length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls); - length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls); - length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1); - length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0); - length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1); - length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0); + container_of(attr, struct fsl_ssi_private, dev_attr); + ssize_t length = 0; + + SIER_SHOW(RFRC_EN, rfrc); + SIER_SHOW(TFRC_EN, tfrc); + SIER_SHOW(CMDAU_EN, cmdau); + SIER_SHOW(CMDDU_EN, cmddu); + SIER_SHOW(RXT_EN, rxt); + SIER_SHOW(RDR1_EN, rdr1); + SIER_SHOW(RDR0_EN, rdr0); + SIER_SHOW(TDE1_EN, tde1); + SIER_SHOW(TDE0_EN, tde0); + SIER_SHOW(ROE1_EN, roe1); + SIER_SHOW(ROE0_EN, roe0); + SIER_SHOW(TUE1_EN, tue1); + SIER_SHOW(TUE0_EN, tue0); + SIER_SHOW(TFS_EN, tfs); + SIER_SHOW(RFS_EN, rfs); + SIER_SHOW(TLS_EN, tls); + SIER_SHOW(RLS_EN, rls); + SIER_SHOW(RFF1_EN, rff1); + SIER_SHOW(RFF0_EN, rff0); + SIER_SHOW(TFE1_EN, tfe1); + SIER_SHOW(TFE0_EN, tfe0); return length; } -- 1.6.0.6

2 1

[alsa-devel] Crackling in USB audio capture
by Ben Hutchings 27 Mar '09

27 Mar '09

I'm writing an ALSA application that captures audio. It worked OK with PCI sound cards but is somehow getting regular faint clicking/crackling with a USB audio device (TI/Burr-Brown PCM2902). Source code is below. What am I doing wrong? This is trying to use a period size near 1602 frames, but choosing other period sizes makes no difference. Also, JACK doesn't seem to have the same problem, so this does appear to be an application bug. Ben. /* Copyright 2007-2009 Ben Hutchings. * Licenced under GNU GPL version 2 or later. */ #include <assert.h> #include <stdbool.h> #include <stddef.h> #include <stdio.h> #include <stdlib.h> #include <string.h> #include <fcntl.h> #include <getopt.h> #include <sys/types.h> #include <unistd.h> #include <netinet/in.h> #include <asoundlib.h> #include "config.h" #include "dif.h" #include "protocol.h" #include "socket.h" static struct option options[] = { {"host", 1, NULL, 'h'}, {"port", 1, NULL, 'p'}, {"system", 1, NULL, 's'}, {"rate", 1, NULL, 'r'}, {"delay", 1, NULL, 'd'}, {"help", 0, NULL, 'H'}, {NULL, 0, NULL, 0} }; static char * mixer_host = NULL; static char * mixer_port = NULL; static void handle_config(const char * name, const char * value) { if (strcmp(name, "MIXER_HOST") == 0) { free(mixer_host); mixer_host = strdup(value); } else if (strcmp(name, "MIXER_PORT") == 0) { free(mixer_port); mixer_port = strdup(value); } } static void usage(const char * progname) { fprintf(stderr, "\ Usage: %s [-h HOST] [-p PORT] [-s ntsc|pal] \\\n\ [-r 48000|32000|44100] [-d DELAY] [DEVICE]\n", progname); } struct transfer_params { snd_pcm_t * pcm; snd_pcm_uframes_t hw_sample_count; const struct dv_system * system; enum dv_sample_rate sample_rate_code; snd_pcm_uframes_t delay_size; int sock; }; void dv_buffer_fill_dummy(uint8_t * buf, const struct dv_system * system); static void transfer_frames(struct transfer_params * params) { static uint8_t buf[DIF_MAX_FRAME_SIZE]; static const unsigned channel_count = 2; unsigned avail_count = 0; unsigned serial_num = 0; const snd_pcm_uframes_t buffer_size = (params->delay_size >= 2000 ? params->delay_size : 2000) + params->hw_sample_count - 1; int16_t * samples = malloc(sizeof(int16_t) * channel_count * buffer_size); dv_buffer_fill_dummy(buf, params->system); for (;;) { unsigned sample_count = params->system->sample_counts[params->sample_rate_code].std_cycle[ serial_num % params->system->sample_counts[params->sample_rate_code].std_cycle_len]; while (avail_count < params->delay_size || avail_count < sample_count) { snd_pcm_sframes_t rc = snd_pcm_readi(params->pcm, samples + channel_count * avail_count, params->hw_sample_count); if (rc < 0) { // Recover from buffer underrun if (rc == -EPIPE && snd_pcm_prepare(params->pcm) == 0) { fprintf(stderr, "WARN: Failing to keep up with audio source\n"); continue; } else { fprintf(stderr, "ERROR: snd_pcm_readi: %s\n", snd_strerror(rc)); exit(1); } } avail_count += rc; } dv_buffer_set_audio(buf, params->sample_rate_code, sample_count, samples); if (write(params->sock, buf, params->system->size) != (ssize_t)params->system->size) { perror("ERROR: write"); exit(1); } memmove(samples, samples + channel_count * sample_count, sizeof(int16_t) * channel_count * (avail_count - sample_count)); avail_count -= sample_count; ++serial_num; } } int main(int argc, char ** argv) { /* Initialise settings from configuration files. */ dvswitch_read_config(handle_config); struct transfer_params params; char * system_name = NULL; long sample_rate = 48000; double delay = 0.2; /* Parse arguments. */ int opt; while ((opt = getopt_long(argc, argv, "h:p:s:r:d:", options, NULL)) != -1) { switch (opt) { case 'h': free(mixer_host); mixer_host = strdup(optarg); break; case 'p': free(mixer_port); mixer_port = strdup(optarg); break; case 's': free(system_name); system_name = strdup(optarg); break; case 'r': sample_rate = strtol(optarg, NULL, 10); break; case 'd': delay = strtod(optarg, NULL); break; case 'H': /* --help */ usage(argv[0]); return 0; default: usage(argv[0]); return 2; } } if (!mixer_host || !mixer_port) { fprintf(stderr, "%s: mixer hostname and port not defined\n", argv[0]); return 2; } if (!system_name || !strcasecmp(system_name, "pal")) { params.system = &dv_system_625_50; } else if (!strcasecmp(system_name, "ntsc")) { params.system = &dv_system_525_60; } else { fprintf(stderr, "%s: invalid system name \"%s\"\n", argv[0], system_name); return 2; } if (sample_rate == 32000) { params.sample_rate_code = dv_sample_rate_32k; } else if (sample_rate == 44100) { params.sample_rate_code = dv_sample_rate_44k1; } else if (sample_rate == 48000) { params.sample_rate_code = dv_sample_rate_48k; } else { fprintf(stderr, "%s: invalid sample rate %ld\n", argv[0], sample_rate); return 2; } if (delay >= 0.0) { params.delay_size = delay * sample_rate; } else { fprintf(stderr, "%s: delays do not work that way!\n", argv[0]); return 2; } if (argc > optind + 1) { fprintf(stderr, "%s: excess argument \"%s\"\n", argv[0], argv[optind + 1]); usage(argv[0]); return 2; } const char * device = (argc == optind) ? "default" : argv[optind]; int rc; /* Prepare to capture and connect a socket to the mixer. */ printf("INFO: Capturing from %s\n", device); rc = snd_pcm_open(&params.pcm, device, SND_PCM_STREAM_CAPTURE, 0); if (rc < 0) { fprintf(stderr, "ERROR: snd_pcm_open: %s\n", snd_strerror(rc)); return 1; } snd_pcm_hw_params_t * hw_params; snd_pcm_hw_params_alloca(&hw_params); rc = snd_pcm_hw_params_any(params.pcm, hw_params); if (rc < 0) { fprintf(stderr, "ERROR: snd_pcm_hw_params_any: %s\n", snd_strerror(rc)); return 1; } rc = snd_pcm_hw_params_set_access(params.pcm, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); if (rc >= 0) rc = snd_pcm_hw_params_set_format(params.pcm, hw_params, SND_PCM_FORMAT_S16); if (rc >= 0) snd_pcm_hw_params_set_channels(params.pcm, hw_params, 2); if (rc >= 0) snd_pcm_hw_params_set_rate_resample(params.pcm, hw_params, 1); if (rc >= 0) snd_pcm_hw_params_set_rate(params.pcm, hw_params, sample_rate, 0); if (rc >= 0) { params.hw_sample_count = params.system->sample_counts[params.sample_rate_code].std_cycle[0]; rc = snd_pcm_hw_params_set_period_size_near(params.pcm, hw_params, &params.hw_sample_count, 0); } if (rc >= 0) { unsigned buffer_time = 250000; rc = snd_pcm_hw_params_set_buffer_time_near(params.pcm, hw_params, &buffer_time, 0); } if (rc >= 0) rc = snd_pcm_hw_params(params.pcm, hw_params); if (rc < 0) { fprintf(stderr, "ERROR: snd_pcm_hw_params: %s\n", snd_strerror(rc)); return 1; } printf("INFO: Connecting to %s:%s\n", mixer_host, mixer_port); params.sock = create_connected_socket(mixer_host, mixer_port); assert(params.sock >= 0); /* create_connected_socket() should handle errors */ if (write(params.sock, GREETING_SOURCE, GREETING_SIZE) != GREETING_SIZE) { perror("ERROR: write"); exit(1); } transfer_frames(&params); close(params.sock); snd_pcm_close(params.pcm); return 0; } -- Ben Hutchings Editing code like this is akin to sticking plasters on the bleeding stump of a severed limb. - me, 29 June 1999

1 1

[alsa-devel] [PATCH 2.6.30] ASoC: remove trigger delay in Freescale MPC8610 sound driver
by Timur Tabi 26 Mar '09

26 Mar '09

Remove the delay from the trigger function in the Freescale MPC8610 sound driver when capture is started. This delay was used to ensure that the DMA controller was active when ALSA call the .pointer function to request a DMA transfer status. A better approach is for the .pointer function to detect that DMA has not started, and return zero instead. This change eliminates the need for the delay. Also add some related code to check for a DMA programming error, and report XRUN if it occurs. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 17 +++++++++++++++++ sound/soc/fsl/fsl_ssi.c | 20 ++------------------ 2 files changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index b3eb857..2c4892c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -697,6 +697,23 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) else position = in_be32(&dma_channel->dar); + /* + * When capture is started, the SSI immediately starts to fill its FIFO. + * This means that the DMA controller is not started until the FIFO is + * full. However, ALSA calls this function before that happens, when + * MR.DAR is still zero. In this case, just return zero to indicate + * that nothing has been received yet. + */ + if (!position) + return 0; + + if ((position < dma_private->dma_buf_phys) || + (position > dma_private->dma_buf_end)) { + dev_err(substream->pcm->card->dev, + "dma pointer is out of range, halting stream\n"); + return SNDRV_PCM_POS_XRUN; + } + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); /* diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 169bca2..72823a2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -466,28 +466,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); - } else { - long timeout = jiffies + 10; - + else setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - - /* Wait until the SSI has filled its FIFO. Without this - * delay, ALSA complains about overruns. When the FIFO - * is full, the DMA controller initiates its first - * transfer. Until then, however, the DMA's DAR - * register is zero, which translates to an - * out-of-bounds pointer. This makes ALSA think an - * overrun has occurred. - */ - while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) && - (jiffies < timeout)); - if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)) - return -EIO; - } break; case SNDRV_PCM_TRIGGER_STOP: -- 1.6.0.6

2 1

[alsa-devel] [RESEND] [PATCH] pxa2xx-ac97: fix displaying GSR after reset timeout
by Luotao Fu 26 Mar '09

26 Mar '09

the variable gsr_bit is set in isr. It is however set to 0 and interrupts are disabled prior to reset on pxa27x and pxa3xx. Hence it doesn't make a lot of sense to show the content of gsr_bit in case of a reset timeout. Signed-off-by: Luotao Fu <l.fu(a)pengutronix.de> --- sound/arm/pxa2xx-ac97-lib.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 35afd0c..25933ba 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -218,7 +218,7 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, GSR | gsr_bits); return false; } @@ -248,7 +248,7 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, GSR | gsr_bits); return false; } -- 1.5.6.5

3 4

[alsa-devel] [patch] alsa: provide a more useful get_unmapped_area handler for pcm
by Daniel Glöckner 26 Mar '09

26 Mar '09

Shared memory mappings on nommu machines require a get_unmapped_area file operation that suggests an address for the mapping. The current implementation returns 0 and thus forces the driver to implement an mmap handler that fixes up the start and end address of the vma. This patch returns the address of the dma buffer, so it should work out of the box for all drivers that use the snd_pcm_runtime->dma_area pointer. Addresses for mapping the status and control pages are returned as well, but to make those work the conditional compilation of snd_pcm_mmap_{status,control} would need to be revised. Signed-off-by: Daniel Glöckner <dg(a)emlix.com> --- sound/core/pcm_native.c | 30 ++++++++++++++++++++++-------- 1 files changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a789efc..09ec8d7 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3377,14 +3377,28 @@ out: #endif /* CONFIG_SND_SUPPORT_OLD_API */ #ifndef CONFIG_MMU -unsigned long dummy_get_unmapped_area(struct file *file, unsigned long addr, - unsigned long len, unsigned long pgoff, - unsigned long flags) -{ - return 0; +static unsigned long snd_pcm_get_unmapped_area(struct file *file, + unsigned long addr, + unsigned long len, + unsigned long pgoff, + unsigned long flags) +{ + struct snd_pcm_file *pcm_file = file->private_data; + struct snd_pcm_substream *substream = pcm_file->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long offset = pgoff << PAGE_SHIFT; + + switch (offset) { + case SNDRV_PCM_MMAP_OFFSET_STATUS: + return (unsigned long)runtime->status; + case SNDRV_PCM_MMAP_OFFSET_CONTROL: + return (unsigned long)runtime->control; + default: + return (unsigned long)runtime->dma_area + offset; + } } #else -# define dummy_get_unmapped_area NULL +# define snd_pcm_get_unmapped_area NULL #endif /* @@ -3403,7 +3417,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, - .get_unmapped_area = dummy_get_unmapped_area, + .get_unmapped_area = snd_pcm_get_unmapped_area, }, { .owner = THIS_MODULE, @@ -3416,6 +3430,6 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, - .get_unmapped_area = dummy_get_unmapped_area, + .get_unmapped_area = snd_pcm_get_unmapped_area, } }; -- 1.6.2.107.ge47ee

1 0

[alsa-devel] CA0106 mic not working, logs attached
by Steven Edwards 26 Mar '09

26 Mar '09

Dear ALSA, I installed Ubuntu 8.10 yesterday, updating ALSA to 1.0.19 to mute my speakers when the headset is plugged in. I unchecked the mute below the Microphone slider, maximized the volume, clicked Close. Sound Recorder detected nothing, so I returned to Volume Control->Recording and noticed that Microphone stayed checked. Repeat x3, same results. I then tried the PulseAudio setup outlined by markbuntu at: http://ubuntuforums.org/showthread.php?t=843012 That failed, so I tried soundrecorder's ALSA upgrade script at: http://ubuntuforums.org/showthread.php?t=1046137 This (sudo bash ./AlsaUpgrade-1.0.x-rev-1.16.sh -di) killed all sound and aplay -l detected no sound cards. Restore (sudo bash ./ AlsaUpgrade-1.0.x-rev-1.16.sh -r) didn't restore my speakers, so I ran alsa-info.sh: http://www.alsa-project.org/db/?f=5a408b2a5328279dbac25a54ce34f6ae707e7cf6 Next, retrying the update script didn't restore my speakers either. alsa-info.sh: http://www.alsa-project.org/db/?f=bd33dc839806cd151e21bdfe4c32bf02a0a9aa57 Finally, (sudo ./configure && sudo make && sudo make install) restored my speakers' ability to output sound, but the mic still isn't recording sound. The original problem--Volume Recorder not accepting/remembering/allowing my unmuted Microphone--remains. alsa-info.sh: http://www.alsa-project.org/db/?f=5e4bcbc453960ece52e3bf21fb475c768b50f979 I don't know where to go next. I hope the reports help and that you will be able to help me. Best, Steven

1 0

[alsa-devel] [PATCH 1/2] ASoC: dapm - Add DAPM for always active DAC or ADC
by Joonyoung Shim 25 Mar '09

25 Mar '09

This patch supports DAPM widget for always active DAC or ADC regardless of stream. This widgets will be used for voice call at TWL4030 codec. TWL4030 codec use the digital signal for voice call, so it needs DAC and ADC to change signal to digital or analog, however currently DAC and ADC widgets cannot support voice call because they are activated only when playback or capture is executed. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- Documentation/sound/alsa/soc/dapm.txt | 3 +++ include/sound/soc-dapm.h | 6 ++++++ 2 files changed, 9 insertions(+), 0 deletions(-) diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 9e67632..50a1e21 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -93,6 +93,9 @@ e.g. stream widgets for HiFi playback and capture SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1), SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), +If you want always active DAC or ADC widget regardless of stream, you can use +SND_SOC_DAPM_DAC_ACTIVE or SND_SOC_DAPM_ADC_ACTIVE instead, and stream name +parameter isn't used. 2.2 Path Domain Widgets ----------------------- diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a7def6a..5e26f81 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -143,6 +143,12 @@ #define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} +#define SND_SOC_DAPM_DAC_ACTIVE(wname, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_dac, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .active = 1} +#define SND_SOC_DAPM_ADC_ACTIVE(wname, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_adc, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .active = 1} /* generic register modifier widget */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ -- 1.5.6.3

2 3

[alsa-devel] [PATCH 2/2] ASoC: twl4030 - Add VDL path support
by Joonyoung Shim 25 Mar '09

25 Mar '09

Adds DAPMs for VDL(voice down link) path. To support VDL path, we have to change DAPM of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer, also codec mode from option1 to option2. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 259 +++++++++++++++++++++----------------------- 1 files changed, 126 insertions(+), 133 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 97738e2..967048a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -42,8 +42,8 @@ */ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* this register not used */ - 0x91, /* REG_CODEC_MODE (0x1) */ - 0xc3, /* REG_OPTION (0x2) */ + 0x90, /* REG_CODEC_MODE (0x1) */ + 0xd3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ 0x20, /* REG_ANAMICL (0x5) */ @@ -56,7 +56,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_AVTXL2PGA (0xC) */ 0x00, /* REG_AVTXR2PGA (0xD) */ 0x01, /* REG_AUDIO_IF (0xE) */ - 0x00, /* REG_VOICE_IF (0xF) */ + 0xe3, /* REG_VOICE_IF (0xF) */ 0x00, /* REG_ARXR1PGA (0x10) */ 0x00, /* REG_ARXL1PGA (0x11) */ 0x6c, /* REG_ARXR2PGA (0x12) */ @@ -313,104 +313,60 @@ static void twl4030_power_down(struct snd_soc_codec *codec) } /* Earpiece */ -static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "DACR1"}; - -static const unsigned int twl4030_earpiece_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_earpiece_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_earpiece_texts), - twl4030_earpiece_texts, - twl4030_earpiece_values); - -static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum); +static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0), +}; /* PreDrive Left */ -static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "DACR2"}; - -static const unsigned int twl4030_predrivel_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predrivel_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predrivel_texts), - twl4030_predrivel_texts, - twl4030_predrivel_values); - -static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum); +static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0), +}; /* PreDrive Right */ -static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "DACL2"}; - -static const unsigned int twl4030_predriver_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predriver_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predriver_texts), - twl4030_predriver_texts, - twl4030_predriver_values); - -static const struct snd_kcontrol_new twl4030_dapm_predriver_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum); +static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0), +}; /* Headset Left */ -static const char *twl4030_hsol_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_hsol_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1, - ARRAY_SIZE(twl4030_hsol_texts), - twl4030_hsol_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsol_control = -SOC_DAPM_ENUM("Route", twl4030_hsol_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0), +}; /* Headset Right */ -static const char *twl4030_hsor_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_hsor_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4, - ARRAY_SIZE(twl4030_hsor_texts), - twl4030_hsor_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsor_control = -SOC_DAPM_ENUM("Route", twl4030_hsor_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0), +}; /* Carkit Left */ -static const char *twl4030_carkitl_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_carkitl_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1, - ARRAY_SIZE(twl4030_carkitl_texts), - twl4030_carkitl_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitl_control = -SOC_DAPM_ENUM("Route", twl4030_carkitl_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0), +}; /* Carkit Right */ -static const char *twl4030_carkitr_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_carkitr_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1, - ARRAY_SIZE(twl4030_carkitr_texts), - twl4030_carkitr_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = -SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0), +}; /* Handsfree Left */ static const char *twl4030_handsfreel_texts[] = - {"Voice", "DACL1", "DACL2", "DACR2"}; + {"Voice", "AudioL1", "AudioL2", "AudioR2"}; static const struct soc_enum twl4030_handsfreel_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, @@ -422,7 +378,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = - {"Voice", "DACR1", "DACR2", "DACL2"}; + {"Voice", "AudioR1", "AudioR2", "AudioL2"}; static const struct soc_enum twl4030_handsfreer_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, @@ -808,6 +764,12 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, } /* + * Voice RX GAIN volume control: + * from -37 to 12 dB in 1 dB steps (mute instead of -37 dB) + */ +static DECLARE_TLV_DB_SCALE(voice_rx_tlv, -3700, 100, 1); + +/* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) */ @@ -856,6 +818,9 @@ static const struct soc_enum twl4030_rampdelay_enum = twl4030_rampdelay_texts); static const struct snd_kcontrol_new twl4030_snd_controls[] = { + SOC_SINGLE_TLV("DAC Voice RX Volume", + TWL4030_REG_VRXPGA, 0, 0x31, 0, voice_rx_tlv), + /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, @@ -871,6 +836,11 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, 6, 0x2, 0, digital_coarse_tlv), + SOC_SINGLE_TLV("DAC Voice Down Link Analog Volume", + TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv), + SOC_SINGLE("DAC Voice Down Link Switch", + TWL4030_REG_VDL_APGA_CTL, 1, 1, 0), + SOC_DOUBLE_R_TLV("DAC1 Analog Playback Volume", TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL, 3, 0x12, 1, analog_tlv), @@ -940,6 +910,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), /* DACs */ + SND_SOC_DAPM_DAC_ACTIVE("DAC Voice", TWL4030_REG_AVDAC_CTL, 4, 0), SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", @@ -950,6 +921,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), /* Analog PGAs */ + SND_SOC_DAPM_PGA("VDL_APGA", TWL4030_REG_VDL_APGA_CTL, + 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL, @@ -990,26 +963,35 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), - /* Output MUX controls */ + /* Output MIXER controls */ /* Earpiece */ - SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_earpiece_control), + SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_controls[0], + ARRAY_SIZE(twl4030_dapm_earpiece_controls)), /* PreDrivL/R */ - SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predrivel_control), - SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predriver_control), + SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_controls[0], + ARRAY_SIZE(twl4030_dapm_predrivel_controls)), + SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_controls[0], + ARRAY_SIZE(twl4030_dapm_predriver_controls)), /* HeadsetL/R */ - SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsol_control, headsetl_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsor_control), + SND_SOC_DAPM_MIXER_E("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_controls[0], + ARRAY_SIZE(twl4030_dapm_hsol_controls), headsetl_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsor_controls[0], + ARRAY_SIZE(twl4030_dapm_hsor_controls)), /* CarkitL/R */ - SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitl_control), - SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitr_control), + SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitl_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitl_controls)), + SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitr_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitr_controls)), + + /* Output MUX controls */ /* HandsfreeL/R */ SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, &twl4030_dapm_handsfreel_control, handsfree_event, @@ -1064,6 +1046,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + {"VDL_APGA", NULL, "DAC Voice"}, + {"Analog L1 Playback Mixer", NULL, "DAC Left1"}, {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, @@ -1076,48 +1060,57 @@ static const struct snd_soc_dapm_route intercon[] = { /* Internal playback routings */ /* Earpiece */ - {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, - {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, - {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + {"Earpiece Mixer", "Voice", "VDL_APGA"}, + {"Earpiece Mixer", "AudioL1", "ARXL1_APGA"}, + {"Earpiece Mixer", "AudioL2", "ARXL2_APGA"}, + {"Earpiece Mixer", "AudioR1", "ARXR1_APGA"}, /* PreDrivL */ - {"PredriveL Mux", "DACL1", "ARXL1_APGA"}, - {"PredriveL Mux", "DACL2", "ARXL2_APGA"}, - {"PredriveL Mux", "DACR2", "ARXR2_APGA"}, + {"PredriveL Mixer", "Voice", "VDL_APGA"}, + {"PredriveL Mixer", "AudioL1", "ARXL1_APGA"}, + {"PredriveL Mixer", "AudioL2", "ARXL2_APGA"}, + {"PredriveL Mixer", "AudioR2", "ARXR2_APGA"}, /* PreDrivR */ - {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, - {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, - {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, + {"PredriveR Mixer", "Voice", "VDL_APGA"}, + {"PredriveR Mixer", "AudioR1", "ARXR1_APGA"}, + {"PredriveR Mixer", "AudioR2", "ARXR2_APGA"}, + {"PredriveR Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetL */ - {"HeadsetL Mux", "DACL1", "ARXL1_APGA"}, - {"HeadsetL Mux", "DACL2", "ARXL2_APGA"}, + {"HeadsetL Mixer", "Voice", "VDL_APGA"}, + {"HeadsetL Mixer", "AudioL1", "ARXL1_APGA"}, + {"HeadsetL Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetR */ - {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, - {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, + {"HeadsetR Mixer", "Voice", "VDL_APGA"}, + {"HeadsetR Mixer", "AudioR1", "ARXR1_APGA"}, + {"HeadsetR Mixer", "AudioR2", "ARXR2_APGA"}, /* CarkitL */ - {"CarkitL Mux", "DACL1", "ARXL1_APGA"}, - {"CarkitL Mux", "DACL2", "ARXL2_APGA"}, + {"CarkitL Mixer", "Voice", "VDL_APGA"}, + {"CarkitL Mixer", "AudioL1", "ARXL1_APGA"}, + {"CarkitL Mixer", "AudioL2", "ARXL2_APGA"}, /* CarkitR */ - {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, - {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, + {"CarkitR Mixer", "Voice", "VDL_APGA"}, + {"CarkitR Mixer", "AudioR1", "ARXR1_APGA"}, + {"CarkitR Mixer", "AudioR2", "ARXR2_APGA"}, /* HandsfreeL */ - {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"}, - {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"}, - {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"}, + {"HandsfreeL Mux", "Voice", "VDL_APGA"}, + {"HandsfreeL Mux", "AudioL1", "ARXL1_APGA"}, + {"HandsfreeL Mux", "AudioL2", "ARXL2_APGA"}, + {"HandsfreeL Mux", "AudioR2", "ARXR2_APGA"}, /* HandsfreeR */ - {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"}, - {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"}, - {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"}, + {"HandsfreeR Mux", "Voice", "VDL_APGA"}, + {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, + {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, + {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, {"OUTR", NULL, "ARXR2_APGA"}, - {"EARPIECE", NULL, "Earpiece Mux"}, - {"PREDRIVEL", NULL, "PredriveL Mux"}, - {"PREDRIVER", NULL, "PredriveR Mux"}, - {"HSOL", NULL, "HeadsetL Mux"}, - {"HSOR", NULL, "HeadsetR Mux"}, - {"CARKITL", NULL, "CarkitL Mux"}, - {"CARKITR", NULL, "CarkitR Mux"}, + {"EARPIECE", NULL, "Earpiece Mixer"}, + {"PREDRIVEL", NULL, "PredriveL Mixer"}, + {"PREDRIVER", NULL, "PredriveR Mixer"}, + {"HSOL", NULL, "HeadsetL Mixer"}, + {"HSOR", NULL, "HeadsetR Mixer"}, + {"CARKITL", NULL, "CarkitL Mixer"}, + {"CARKITR", NULL, "CarkitR Mixer"}, {"HFL", NULL, "HandsfreeL Mux"}, {"HFR", NULL, "HandsfreeR Mux"}, -- 1.5.6.3

3 4

Re: [alsa-devel] Acer Ferrari 5000
by Takashi Iwai 25 Mar '09

25 Mar '09

At Wed, 25 Mar 2009 01:47:16 -0700, Russ Dill wrote: > > The patch is causing breakage for me (no audio): > > [37998.770034] HDA Intel 0000:00:14.2: PCI INT A disabled > [38019.228311] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, > low) -> IRQ 16 > [38019.360994] input: HDA Digital PCBeep as > /devices/pci0000:00/0000:00:14.2/input/input9 > [38019.389307] hda_codec: formats == 0 (nid=0x2, val=0xe0560, ovrd=1, > streams=0xe0560) > [38019.389318] hda_codec: cannot attach PCM stream 0 for codec #1 > [38019.389367] hda_codec: formats == 0 (nid=0x6, val=0x1e0560, ovrd=1, > streams=0x1e0560) > [38019.389373] hda_codec: cannot attach PCM stream 1 for codec #1 > [38019.389435] hda_codec: formats == 0 (nid=0x9, val=0x60160, ovrd=1, > streams=0x60160) > [38019.389440] hda_codec: cannot attach PCM stream 2 for codec #1 Looks like something wrong in my caching code, likely a wrong hash key generation or so. Will take a look after returning from the vacation. thanks, Takashi > > > On Mon, Mar 23, 2009 at 4:27 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 23 Mar 2009 16:12:38 -0700, > > Russ Dill wrote: > >> > >> On Mon, Mar 23, 2009 at 3:55 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Mon, 23 Mar 2009 15:48:19 -0700, > >> > Russ Dill wrote: > >> >> > >> >> On Mon, Mar 23, 2009 at 3:45 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> > At Mon, 23 Mar 2009 15:28:57 -0700, > >> >> > Russ Dill wrote: > >> >> >> > >> >> >> >> > BTW, I found why model=auto doesn't work well on your machine. > >> >> >> >> > It's because (again) of BIOS. It doesn't set the codec SSID properly > >> >> >> >> > so the driver doesn't accept the HP auto-toggle and other features. > >> >> >> >> > > >> >> >> >> > So, in your case, using model=acer-aspire would be likely the best > >> >> >> >> > choice (supposing it matches with most functionality). If you have no > >> >> >> >> > problem (at least regressions) with this model, I'll add the entry to > >> >> >> >> > point to model=acer-aspire > >> >> >> >> > >> >> >> >> Add Acer Ferrar 5000 Quirk to Intel HDA driver > >> >> >> > > >> >> >> > Thanks, a similar fix was already merged now. > >> >> >> > >> >> >> It all works for a while, but eventually I'll get a "azx_get_response > >> >> >> timeout, switching to polling mode: last cmd=0x10bb2001" and it no > >> >> >> longer responds to headphone plug events without an unload/reload of > >> >> >> the module. > >> >> > > >> >> > Did you try my test patch? > >> >> > >> >> It is with the patch: > >> >> > >> >> [16877.337044] hda_intel: azx_get_response timeout, switching to > >> >> polling mode: last cmd=0x106f000a > >> >> [16878.341056] hda_intel: azx_get_response timeout (ERROR): last cmd=0x106f000a > >> >> [16879.345036] hda_intel: azx_get_response timeout (ERROR): last cmd=0x106f000a > >> >> [16880.380016] hda_intel: azx_get_response timeout (ERROR): last cmd=0x106f000a > >> >> [16881.384022] hda_intel: azx_get_response timeout (ERROR): last cmd=0x106f000a > >> > > >> > Interesting, this is again the same verb. > >> > So, there seems something unstable regarding the digital PCM... > >> > > >> > And do you still have the problem with HP plugging? > >> > > >> > >> There seems to be some confusion. All I'm doing right now is playing > >> audio through the analog PCM. After a while, I get that dmesg output > >> and HP plugging stops working. > > > > Yes, that's what I understood, too. The driver still queries the PCM > > parameter for the digital output widget because you turn on "IEC958 > > Default PCM" switch, thus the signal is still routed to the digital, > > too. > > > > > > The below is another test patch to reduce this problem (hopefully). > > It basically caches the parameter values to avoid superfluous > > queries. It's against the very latest alsa-driver snapshot. > > Apply it in addition to the previous test patch. > > > > > > thanks, > > > > Takashi > > > > --- > > > > diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c > > index a4e5e59..1d88e2f 100644 > > --- a/sound/pci/hda/hda_codec.c > > +++ b/sound/pci/hda/hda_codec.c > > @@ -1053,6 +1053,8 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); > > /* FIXME: more better hash key? */ > > #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) > > #define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) > > +#define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24)) > > +#define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24)) > > #define INFO_AMP_CAPS (1<<0) > > #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) > > > > @@ -1143,7 +1145,9 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, > > } > > EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); > > > > -u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) > > +static unsigned int > > +query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key, > > + unsigned int (*func)(struct hda_codec *, hda_nid_t)) > > { > > struct hda_amp_info *info; > > > > @@ -1151,11 +1155,22 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) > > if (!info) > > return 0; > > if (!info->head.val) { > > - info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); > > info->head.val |= INFO_AMP_CAPS; > > + info->amp_caps = func(codec, nid); > > } > > return info->amp_caps; > > } > > + > > +static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); > > +} > > + > > +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid), > > + read_pin_cap); > > +} > > EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); > > > > /* > > @@ -2538,6 +2553,41 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, > > } > > EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); > > > > +static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + unsigned int val = 0; > > + if (nid != codec->afg && > > + (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) > > + val = snd_hda_param_read(codec, nid, AC_PAR_PCM); > > + if (!val || val == -1) > > + val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); > > + if (!val || val == -1) > > + return 0; > > + return val; > > +} > > + > > +static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid), > > + get_pcm_param); > > +} > > + > > +static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); > > + if (!streams || streams == -1) > > + streams = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); > > + if (!streams || streams == -1) > > + return 0; > > + return streams; > > +} > > + > > +static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid), > > + get_stream_param); > > +} > > + > > /** > > * snd_hda_query_supported_pcm - query the supported PCM rates and formats > > * @codec: the HDA codec > > @@ -2556,15 +2606,8 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, > > { > > unsigned int i, val, wcaps; > > > > - val = 0; > > wcaps = get_wcaps(codec, nid); > > - if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) { > > - val = snd_hda_param_read(codec, nid, AC_PAR_PCM); > > - if (val == -1) > > - return -EIO; > > - } > > - if (!val) > > - val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); > > + val = query_pcm_param(codec, nid); > > > > if (ratesp) { > > u32 rates = 0; > > @@ -2586,15 +2629,9 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, > > u64 formats = 0; > > unsigned int streams, bps; > > > > - streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); > > - if (streams == -1) > > + streams = query_stream_param(codec, nid); > > + if (!streams) > > return -EIO; > > - if (!streams) { > > - streams = snd_hda_param_read(codec, codec->afg, > > - AC_PAR_STREAM); > > - if (streams == -1) > > - return -EIO; > > - } > > > > bps = 0; > > if (streams & AC_SUPFMT_PCM) { > > @@ -2668,17 +2705,9 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, > > int i; > > unsigned int val = 0, rate, stream; > > > > - if (nid != codec->afg && > > - (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { > > - val = snd_hda_param_read(codec, nid, AC_PAR_PCM); > > - if (val == -1) > > - return 0; > > - } > > - if (!val) { > > - val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); > > - if (val == -1) > > - return 0; > > - } > > + val = query_pcm_param(codec, nid); > > + if (!val) > > + return 0; > > > > rate = format & 0xff00; > > for (i = 0; i < AC_PAR_PCM_RATE_BITS; i++) > > @@ -2690,12 +2719,8 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, > > if (i >= AC_PAR_PCM_RATE_BITS) > > return 0; > > > > - stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM); > > - if (stream == -1) > > - return 0; > > - if (!stream && nid != codec->afg) > > - stream = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); > > - if (!stream || stream == -1) > > + stream = query_stream_param(codec, nid); > > + if (!stream) > > return 0; > > > > if (stream & AC_SUPFMT_PCM) { > > >

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[alsa-devel] building soc module drivers
by Geiger Ho 25 Mar '09

25 Mar '09

Hi all, I would like to build the WM8510 SOC sound driver for my ARM platform. After downloading alsa-driver-1.0.19, "./configure --with-cross=arm-linux- --with-kernel=/home/source/linux-2.6.22.19 --with-build=/home/source/linux-2.6.22.19 --with-cards=all", and then make. I get no wm8510.c compiled. What other stuff should I do to make it compiled? Many thanks. Best regards, Geiger -- Geiger Ho <geiger.ho(a)fmp.fujitsu.com>, armed with Agile and Lean. Mission: To deliver quality of work. Phone : (852) 2736-3232 Direct : (852) 2737-9681 Fax : (852) 2314-4207 Fujitsu Microelectronics Pacific Asia Ltd, Design Centre

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Re: [alsa-devel] How to find sampling rate supported
by reetesh chandran 24 Mar '09

24 Mar '09

Thanks Takashi. If the lib does not provide the API to read the device capabilitiess, is there a way for applications to directly read it from the Alsa kernel driver ? --- On Mon, 3/23/09, Takashi Iwai <tiwai(a)suse.de> wrote: From: Takashi Iwai <tiwai(a)suse.de> Subject: Re: [alsa-devel] How to find sampling rate supported To: "reetesh chandran" <reeteshchandran(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Date: Monday, March 23, 2009, 3:03 AM At Sun, 22 Mar 2009 09:00:00 -0700 (PDT), reetesh chandran wrote: > > Hi all, > > I am new to the group and to Alsa. > > For our project, we have to find the max sampling rate and other pcm hardware parameters that the connected USB headset supports. I looked at the alsa library, but could not find an API which returned this information. > > I found snd_pcm_hw_params_test_* function to test a specific value is supported, but I was wondering if there was a way to query the device to get the values. The min / max can be obtained via *_get_rate_min() and *_get_rate_max(), at least. The other detailed information can't be get except for testing the explicit value via *_test_rate(), so far. HTH, Takashi _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] ASoC: twl4030 - Add VDL path support
by Joonyoung Shim 24 Mar '09

24 Mar '09

Adds DAPMs for VDL(voice down link) path. To support VDL path, we have to change DAPM of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 226 ++++++++++++++++++++------------------------ 1 files changed, 103 insertions(+), 123 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 5ed5f1b..317d710 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -313,104 +313,60 @@ static void twl4030_power_down(struct snd_soc_codec *codec) } /* Earpiece */ -static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "DACR1"}; - -static const unsigned int twl4030_earpiece_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_earpiece_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_earpiece_texts), - twl4030_earpiece_texts, - twl4030_earpiece_values); - -static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum); +static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0), +}; /* PreDrive Left */ -static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "DACR2"}; - -static const unsigned int twl4030_predrivel_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predrivel_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predrivel_texts), - twl4030_predrivel_texts, - twl4030_predrivel_values); - -static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum); +static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0), +}; /* PreDrive Right */ -static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "DACL2"}; - -static const unsigned int twl4030_predriver_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predriver_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predriver_texts), - twl4030_predriver_texts, - twl4030_predriver_values); - -static const struct snd_kcontrol_new twl4030_dapm_predriver_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum); +static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0), +}; /* Headset Left */ -static const char *twl4030_hsol_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_hsol_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1, - ARRAY_SIZE(twl4030_hsol_texts), - twl4030_hsol_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsol_control = -SOC_DAPM_ENUM("Route", twl4030_hsol_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0), +}; /* Headset Right */ -static const char *twl4030_hsor_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_hsor_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4, - ARRAY_SIZE(twl4030_hsor_texts), - twl4030_hsor_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsor_control = -SOC_DAPM_ENUM("Route", twl4030_hsor_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0), +}; /* Carkit Left */ -static const char *twl4030_carkitl_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_carkitl_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1, - ARRAY_SIZE(twl4030_carkitl_texts), - twl4030_carkitl_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitl_control = -SOC_DAPM_ENUM("Route", twl4030_carkitl_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0), +}; /* Carkit Right */ -static const char *twl4030_carkitr_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_carkitr_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1, - ARRAY_SIZE(twl4030_carkitr_texts), - twl4030_carkitr_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = -SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0), +}; /* Handsfree Left */ static const char *twl4030_handsfreel_texts[] = - {"Voice", "DACL1", "DACL2", "DACR2"}; + {"Voice", "AudioL1", "AudioL2", "AudioR2"}; static const struct soc_enum twl4030_handsfreel_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, @@ -422,7 +378,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = - {"Voice", "DACR1", "DACR2", "DACL2"}; + {"Voice", "AudioR1", "AudioR2", "AudioL2"}; static const struct soc_enum twl4030_handsfreer_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, @@ -940,6 +896,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), /* DACs */ + SND_SOC_DAPM_DAC("DAC Voice", "Voice Down Link", + TWL4030_REG_AVDAC_CTL, 4, 0), SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", @@ -950,6 +908,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), /* Analog PGAs */ + SND_SOC_DAPM_PGA("VDL_APGA", TWL4030_REG_VDL_APGA_CTL, + 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL, @@ -990,26 +950,35 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), - /* Output MUX controls */ + /* Output MIXER controls */ /* Earpiece */ - SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_earpiece_control), + SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_controls[0], + ARRAY_SIZE(twl4030_dapm_earpiece_controls)), /* PreDrivL/R */ - SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predrivel_control), - SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predriver_control), + SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_controls[0], + ARRAY_SIZE(twl4030_dapm_predrivel_controls)), + SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_controls[0], + ARRAY_SIZE(twl4030_dapm_predriver_controls)), /* HeadsetL/R */ - SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsol_control, headsetl_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsor_control), + SND_SOC_DAPM_MIXER_E("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_controls[0], + ARRAY_SIZE(twl4030_dapm_hsol_controls), headsetl_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsor_controls[0], + ARRAY_SIZE(twl4030_dapm_hsor_controls)), /* CarkitL/R */ - SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitl_control), - SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitr_control), + SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitl_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitl_controls)), + SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitr_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitr_controls)), + + /* Output MUX controls */ /* HandsfreeL/R */ SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, &twl4030_dapm_handsfreel_control, handsfree_event, @@ -1064,6 +1033,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + {"VDL_APGA", NULL, "DAC Voice"}, + {"Analog L1 Playback Mixer", NULL, "DAC Left1"}, {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, @@ -1076,37 +1047,46 @@ static const struct snd_soc_dapm_route intercon[] = { /* Internal playback routings */ /* Earpiece */ - {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, - {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, - {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + {"Earpiece Mixer", "Voice", "VDL_APGA"}, + {"Earpiece Mixer", "AudioL1", "ARXL1_APGA"}, + {"Earpiece Mixer", "AudioL2", "ARXL2_APGA"}, + {"Earpiece Mixer", "AudioR1", "ARXR1_APGA"}, /* PreDrivL */ - {"PredriveL Mux", "DACL1", "ARXL1_APGA"}, - {"PredriveL Mux", "DACL2", "ARXL2_APGA"}, - {"PredriveL Mux", "DACR2", "ARXR2_APGA"}, + {"PredriveL Mixer", "Voice", "VDL_APGA"}, + {"PredriveL Mixer", "AudioL1", "ARXL1_APGA"}, + {"PredriveL Mixer", "AudioL2", "ARXL2_APGA"}, + {"PredriveL Mixer", "AudioR2", "ARXR2_APGA"}, /* PreDrivR */ - {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, - {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, - {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, + {"PredriveR Mixer", "Voice", "VDL_APGA"}, + {"PredriveR Mixer", "AudioR1", "ARXR1_APGA"}, + {"PredriveR Mixer", "AudioR2", "ARXR2_APGA"}, + {"PredriveR Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetL */ - {"HeadsetL Mux", "DACL1", "ARXL1_APGA"}, - {"HeadsetL Mux", "DACL2", "ARXL2_APGA"}, + {"HeadsetL Mixer", "Voice", "VDL_APGA"}, + {"HeadsetL Mixer", "AudioL1", "ARXL1_APGA"}, + {"HeadsetL Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetR */ - {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, - {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, + {"HeadsetR Mixer", "Voice", "VDL_APGA"}, + {"HeadsetR Mixer", "AudioR1", "ARXR1_APGA"}, + {"HeadsetR Mixer", "AudioR2", "ARXR2_APGA"}, /* CarkitL */ - {"CarkitL Mux", "DACL1", "ARXL1_APGA"}, - {"CarkitL Mux", "DACL2", "ARXL2_APGA"}, + {"CarkitL Mixer", "Voice", "VDL_APGA"}, + {"CarkitL Mixer", "AudioL1", "ARXL1_APGA"}, + {"CarkitL Mixer", "AudioL2", "ARXL2_APGA"}, /* CarkitR */ - {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, - {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, + {"CarkitR Mixer", "Voice", "VDL_APGA"}, + {"CarkitR Mixer", "AudioR1", "ARXR1_APGA"}, + {"CarkitR Mixer", "AudioR2", "ARXR2_APGA"}, /* HandsfreeL */ - {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"}, - {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"}, - {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"}, + {"HandsfreeL Mux", "Voice", "VDL_APGA"}, + {"HandsfreeL Mux", "AudioL1", "ARXL1_APGA"}, + {"HandsfreeL Mux", "AudioL2", "ARXL2_APGA"}, + {"HandsfreeL Mux", "AudioR2", "ARXR2_APGA"}, /* HandsfreeR */ - {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"}, - {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"}, - {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"}, + {"HandsfreeR Mux", "Voice", "VDL_APGA"}, + {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, + {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, + {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, -- 1.5.6.3

4 6

[alsa-devel] Acer Ferrari 5000
by Russ Dill 24 Mar '09

24 Mar '09

GIven the recent changes in gnome land (pulseaudio), I'd like to be able to get alsa support for my hardware fixed up. My biggest annoyance right now is that when I plug headphones in, it doesn't mute the speakers. Another issue I have is that I'd like to have support for audio out on HDMI (which is supported by Vista). On with the show. I'm currently running 2.6.28-9-generic (Ubuntu 9.04). All my info is below, what can I do to get everything exposed and working? 00:14.2 Audio device: ATI Technologies Inc IXP SB4x0 High Definition Audio Controller (rev 01) Subsystem: Acer Incorporated [ALI] Device 010a Flags: bus master, slow devsel, latency 64, IRQ 16 Memory at b0000000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel HDA Intel 0000:00:14.2: PCI INT A disabled HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 hda-intel: Invalid position buffer, using LPIB read method instead. hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0x101f000a hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x101f000a (methinks these timeouts are no good) devices: 2: : timer 3: [ 0- 4]: digital audio capture 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control pcm0c/info: card: 0 device: 0 subdevice: 0 stream: CAPTURE id: ALC883 Analog name: ALC883 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 pcm0p/info: card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: ALC883 Analog name: ALC883 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 pcm4c/info: card: 0 device: 4 subdevice: 0 stream: CAPTURE id: ALC883 Analog name: ALC883 Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 codec#0: Codec: Conexant ID 2bfa Address: 0 Vendor Id: 0x14f12bfa Subsystem Id: 0x1025010a Revision Id: 0x90000 Modem Function Group: 0x2 codec#1: Codec: Realtek ALC883 Address: 1 Vendor Id: 0x10ec0883 Subsystem Id: 0x10ec0883 Revision Id: 0x100002 No Modem Function Group found Default PCM: N/A Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x98 0x98] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x15 0x15] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x15 0x15] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01014030: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01011031: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01016032: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x3, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01012033: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x3, Sequence = 0x3 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19850: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19c80: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x8, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x01813050: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x02214c40: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x99331150: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Black DefAssociation = 0x5, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x00000100: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x01441170: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Black DefAssociation = 0x7, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x01c46160: [Jack] SPDIF In at Ext Rear Conn = RCA, Color = Orange DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

2 22

[alsa-devel] A small annoyance with alsa-driver
by "Rafał Mużyło" 23 Mar '09

23 Mar '09

As your bugtrack seem unresponsive, I'm asking here: could you look at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4274 and either correct the docs or make stuff work like it did before ?

1 0

[alsa-devel] trouble using dmix with tts systems (accessibility )
by Halim Sahin 23 Mar '09

23 Mar '09

hello, I have asked this question to alsa user list but got no useful answer. I am a blind linux user and need asistive Technology to use my pc. The problem I have experienced with alsa is that it is sometimes too slow to stop previous output before starting a new one. The result of this ends in several speech outputs at a time. The problem only occurs if the dmix plugin is used. Is there a way to modify the buffer to avoid these delays??? I have tried to customize my buffer_size vallue in ~/.asoundrc but I wasn't able to to create a working setup with dmix. All help appreciated ;-) Regards halim

2 2

[alsa-devel] [PATCH] alsa: Intel HDA output and align Ids.
by Pascal de Bruijn | Unilogic Networks B.V. 23 Mar '09

23 Mar '09

This patch does two things: Output Intel HDA Function Id in /proc/asound/cardX/codec#X Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros Before: Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x100100 After: Function Id: 0x1 Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x0100100 As report on the Kernel Bugzilla #12888 Signed-off-by: Pascal de Bruijn <pascal(a)unilogicnetworks.net> diff -Nurp linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.c linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.c --- linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.c 2009-03-20 16:05:15.000000000 +0100 +++ linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.c 2009-03-23 10:37:30.000000000 +0100 @@ -647,9 +647,9 @@ static void /*__devinit*/ setup_fg_nodes total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, + AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; diff -Nurp linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.h linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.h --- linux-2.6.29-rc8-orig/sound/pci/hda/hda_codec.h 2009-03-20 16:05:15.000000000 +0100 +++ linux-2.6.29-rc8-func/sound/pci/hda/hda_codec.h 2009-03-20 16:10:07.000000000 +0100 @@ -739,6 +739,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; diff -Nurp linux-2.6.29-rc8-orig/sound/pci/hda/hda_proc.c linux-2.6.29-rc8-func/sound/pci/hda/hda_proc.c --- linux-2.6.29-rc8-orig/sound/pci/hda/hda_proc.c 2009-03-20 16:05:15.000000000 +0100 +++ linux-2.6.29-rc8-func/sound/pci/hda/hda_proc.c 2009-03-20 16:10:07.000000000 +0100 @@ -467,8 +467,9 @@ static void print_codec_info(struct snd_ snd_iprintf(buffer, "Codec: %s\n", codec->name ? codec->name : "Not Set"); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); if (codec->mfg)

2 1

[alsa-devel] How to find sampling rate supported
by reetesh chandran 23 Mar '09

23 Mar '09

Hi all, I am new to the group and to Alsa. For our project, we have to find the max sampling rate and other pcm hardware parameters that the connected USB headset supports. I looked at the alsa library, but could not find an API which returned this information. I found snd_pcm_hw_params_test_* function to test a specific value is supported, but I was wondering if there was a way to query the device to get the values. Thanks in advance, Reetesh

2 1

Re: [alsa-devel] Bugs on aspire one A150
by Andreas Mohr 23 Mar '09

23 Mar '09

Hi, On Mon, Mar 16, 2009 at 05:19:38PM +0100, Takashi Iwai wrote: > At Mon, 16 Mar 2009 17:06:35 +0100, > 私 wrote: > > > > > > What are "sliders"? > > > > > > Umm, volume level controls. > > > > Yes but there are many of such :) > > > > More exactly, from the driver perspective, there are no volume > > controls but only there are control elements with integer values. > > Do you mean "Capture Volume" control or which one? Hmm, ok, this needs to be more precise: In gamix (codec "HDA Intel : Realtek ALC268"), the Capture Volume control. > > And, is the behavior consistent regardless of the value high, i.e. > > the key is only whether the values for both channels are identical? > > BTW, what if you record with the following definition? > Put the below to ~/.asoundrc > > pcm.imix { > type plug > slave.pcm "hw" > ttable.0.0 0.5 > ttable.0.1 -0.5 > } > > and record like > > % aplay -Dimix -c1 foo.wav Does NOT exhibit the "equal sliders == no sound" bug (apart from this sliders are acting normally, i.e. slider low == no sound), despite being a "plug" type definition (this is what you wanted to discern, right? ;). Thanks! Andreas

3 23

[alsa-devel] MSI GX620 support
by okias 22 Mar '09

22 Mar '09

MSI GX620X-061CZ, but probably is this audio chip with this configuration on all GX620 Connectors: Green, standart jack 3.5 headphones Blue, line-in Pink, mic Grey, side DOLBY HOME THEATER KO 127 3stack-dig 3-jack with SPDIF I/O *OK 128 6stack-dig 6-jack digital with SPDIF I/O // work SIDE+jack, but SIDE can't be configured, not deactivating integrated repro *OK 129 3stack-6ch 3-jack 6-channel // headphone power off integrated repro, 6ch working, no 8ch OK 130 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O //same 3stacj-6ch KO 131 6stack-dig-demo 6-jack digital for Intel demo board OK 132 acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)//jack+side, but SIDE can't be configured, headphone power off integr., notworking 6ch KO 133 acer-aspire Acer Aspire 9810 *OK 134 acer-aspire-4930g Acer Aspire 4930G // Front switch shutting down both repro (external in jack, integrated), 6channels + SIDE (should be 8ch), and if is plugged something into jack integrated repro still playing KO 135 medion Medion Laptops KO 136 medion-md2 Medion MD2 !! 137 targa-dig Targa/MSI // If is plugged jack, then integrated repro power down! but missing settings to keep integrated repro playing and 8ch isn't present(side not working). KO 138 targa-2ch-dig Targs/MSI with 2-channel KO 139 laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) // SIDE working, but 6ch conf not, only side + repro output KO 140 lenovo-101e Lenovo 101E // nothing new... less settings, no side... no 6ch KO 141 lenovo-nb0763 Lenovo NB0763 KO 142 lenovo-ms7195-dig Lenovo MS7195 KO 143 lenovo-sky Lenovo Sky KO 144 haier-w66 Haier W66 KO 145 3stack-hp HP machines with 3stack (Lucknow, Samba boards) // 4. channel not working, etc. KO 146 6stack-dell Dell machines with 6stack (Inspiron 530) // if is activated 8ch conf, side is activated, but 4 and 6th channel not working with this conf KO 147 mitac Mitac 8252D KO 148 clevo-m720 Clevo M720 laptop series KO 149 fujitsu-pi2515 Fujitsu AMILO Pi2515 *OK 150 fujitsu-xa3530 Fujitsu AMILO XA3530 // 6ch mode working, but SIDE working even if is set only 6ch KO 151 3stack-6ch-intel Intel DG33* boards OK 152 auto auto-config reading BIOS (default) // SIDE+JACK, BOTH MIC OK, can't be configured to 4,6 or 8ch, if is pluged into JACK repro then integrated playing... OK means, it's in usable state, KO mean something important doesn't work, or nothing new in compare to before tested configurations. *OK means very good configurations 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Micro-Star International Co., Ltd. Device 6510 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f8ef8000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Mask- 64bit+ Count=1/1 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel If you need any addional information, testing patchs (gentoo), please contact me :-) okias -- Jabber/XMPP: okias(a)isgeek.info Webové stránky: http://www.jabber.xf.cz

1 0

[alsa-devel] [PATCH] alsa: Intel HDA output and align Ids.
by Pascal de Bruijn | Unilogic Networks B.V. 20 Mar '09

20 Mar '09

Output Intel HDA Function Id in /proc/asound/cardX/codec#X Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros As submitted to Kernel Bugzilla #12888 Signed-off-by: Pascal de Bruijn <pascal(a)unilogicnetworks.net> diff -Nurp linux-2.6.28.8-orig/sound/pci/hda/hda_codec.c linux-2.6.28.8-func/sound/pci/hda/hda_codec.c --- linux-2.6.28.8-orig/sound/pci/hda/hda_codec.c 2009-03-17 01:50:03.000000000 +0100 +++ linux-2.6.28.8-func/sound/pci/hda/hda_codec.c 2009-03-17 10:39:45.000000000 +0100 @@ -519,9 +519,8 @@ static void __devinit setup_fg_nodes(str total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; diff -Nurp linux-2.6.28.8-orig/sound/pci/hda/hda_codec.h linux-2.6.28.8-func/sound/pci/hda/hda_codec.h --- linux-2.6.28.8-orig/sound/pci/hda/hda_codec.h 2009-03-17 01:50:03.000000000 +0100 +++ linux-2.6.28.8-func/sound/pci/hda/hda_codec.h 2009-03-17 10:39:45.000000000 +0100 @@ -693,6 +693,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; diff -Nurp linux-2.6.28.8-orig/sound/pci/hda/hda_proc.c linux-2.6.28.8-func/sound/pci/hda/hda_proc.c --- linux-2.6.28.8-orig/sound/pci/hda/hda_proc.c 2009-03-17 01:50:03.000000000 +0100 +++ linux-2.6.28.8-func/sound/pci/hda/hda_proc.c 2009-03-17 10:40:08.000000000 +0100 @@ -520,9 +520,10 @@ static void print_codec_info(struct snd_ snd_hda_get_codec_name(codec, buf, sizeof(buf)); snd_iprintf(buffer, "Codec: %s\n", buf); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); - snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); + snd_iprintf(buffer, "Revision Id: 0x%08x\n", codec->revision_id); if (codec->mfg) snd_iprintf(buffer, "Modem Function Group: 0x%x\n", codec->mfg);

2 1

Re: [alsa-devel] [PATCH 6/9] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by Medisetty, Naresh 19 Mar '09

19 Mar '09

> On Mon, Mar 16, 2009 at 08:05:25AM -0400, Naresh Medisetty wrote: > > > + CLK("soc-audio.0", NULL, &mcasp0_clk), > > + CLK("soc-audio.1", NULL, &mcasp1_clk), > > Assuming this is a shorthand for defining two platform devices for two > ASoC cards I'd suggest that you instead move towards registering clocks > for the DAIs and probing them as platform devices - take a look at how > the PXA AC97 driver now does this for an example. The soc-audio device > will hopefully get removed at some point and you can't guarantee that > the user won't want to put the two DAIs into a single sound card. Yes, This is a shorthand for defining two platform devices for two ASoC cards. I will re-work on this and submit a patch

4 4

[alsa-devel] [PATCH 1/9] ARM: DaVinci: ASoC: Adds ASoC driver support for TI DM646X EVM platform
by Naresh Medisetty 19 Mar '09

19 Mar '09

Adds ASoC driver support for TI DM646X EVM platform Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=commi…. sound/soc/davinci/davinci-evm.c | 131 ++++++++++++++++++++++++++++++++++++--- 1 files changed, 121 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 95c44bf..9e832c9 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -25,10 +25,15 @@ #include <mach/asp.h> #include <mach/edma.h> #include <mach/mux.h> +#include <mach/cpu.h> + #include "../codecs/tlv320aic3x.h" +#include "../codecs/codec_stubs.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-i2s-mcasp.h" + #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -43,14 +48,14 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (cpu_is_davinci_dm355() || cpu_is_davinci_dm646x()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by * board-dm644x-evm.c using GPIOs from U18. There are six * options; here we "know" we use a 48 KHz sample rate. */ - else if (machine_is_davinci_evm()) + else if (cpu_is_davinci_dm644x()) sysclk = 12288000; else @@ -144,6 +149,25 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = davinci_iis_mcasp_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "MCASP SPDIF", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[1], + .codec_dai = dit_stub_dai, + .ops = &evm_ops, + }, +}; + + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -152,12 +176,27 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = 2, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -165,6 +204,14 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, +}; + + /* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ static struct resource evm_snd_resources[] = { { @@ -174,11 +221,64 @@ static struct resource evm_snd_resources[] = { }, }; +static struct resource dm6467_evm_snd_resources[] = { + { + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, +}; + + +static u8 dm6467_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm6467_dit_serializer_direction[] = { + TX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + + static struct evm_snd_platform_data evm_snd_data = { .tx_dma_ch = DAVINCI_DMA_ASP0_TX, .rx_dma_ch = DAVINCI_DMA_ASP0_RX, }; +static struct evm_snd_platform_data dm6467_evm_snd_data[] = { + { + .clk_name = "McASPCLK0", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .rx_dma_ch = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = 4, + .tdm_slots = 2, + .serial_dir = dm6467_iis_serializer_direction, + .eventq_no = EVENTQ_0, + .codec_fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + { + .clk_name = "McASPCLK1", + .tx_dma_ch = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .rx_dma_ch = -1, + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = 4, + .tdm_slots = 32, + .serial_dir = dm6467_dit_serializer_direction, + .eventq_no = EVENTQ_0, + .codec_fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, +}; + + /* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ static struct resource dm335evm_snd_resources[] = { { @@ -199,23 +299,33 @@ static int __init evm_init(void) { struct resource *resources; struct evm_snd_platform_data *data; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; + int res_size; - if (machine_is_davinci_evm()) { + if (cpu_is_davinci_dm644x()) { davinci_cfg_reg(DM644X_MCBSP); - + evm_snd_dev_data = &evm_snd_devdata; resources = evm_snd_resources; data = &evm_snd_data; + res_size = ARRAY_SIZE(evm_snd_resources); index = 0; - } else if (machine_is_davinci_dm355_evm()) { + } else if (cpu_is_davinci_dm355()) { /* we don't use ASP1 IRQs, or we'd need to mux them ... */ davinci_cfg_reg(DM355_EVT8_ASP1_TX); davinci_cfg_reg(DM355_EVT9_ASP1_RX); - + evm_snd_dev_data = &evm_snd_devdata; resources = dm335evm_snd_resources; + res_size = ARRAY_SIZE(dm335evm_snd_resources); data = &dm335evm_snd_data; index = 1; + } else if (cpu_is_davinci_dm646x()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + resources = dm6467_evm_snd_resources; + res_size = ARRAY_SIZE(dm6467_evm_snd_resources); + data = dm6467_evm_snd_data; + index = 0; } else return -EINVAL; @@ -223,11 +333,12 @@ static int __init evm_init(void) if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, data, sizeof(data)); + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = data; - ret = platform_device_add_resources(evm_snd_device, resources, 1); + ret = platform_device_add_resources(evm_snd_device, + resources, res_size); if (ret) { platform_device_put(evm_snd_device); return ret; -- 1.5.6

5 5

[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 19 Mar '09

19 Mar '09

The following changes since commit da88b48b84e1a504b6a19aff9d5b8236a59e228a: Mark Brown (1): Merge branch 'pxa-ssp' into for-2.6.30 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Jarkko Nikula (2): ASoC: OMAP: N810: Mark not connected input pins ASoC: OMAP: N810: Add more jack functions Lopez Cruz, Misael (1): ASoC: Declare Headset as Mic and Headphone widgets for SDP3430 Mark Brown (3): ASoC: Further optimise WM8400 bias configuration sequence ASoC: Add separate AVDD for WM8400 ASoC: Add FLL support for WM8400 sound/soc/codecs/wm8400.c | 159 ++++++++++++++++++++++++++++++++++++++------- sound/soc/omap/n810.c | 33 +++++++++- sound/soc/omap/sdp3430.c | 24 +++++-- 3 files changed, 181 insertions(+), 35 deletions(-)

2 1

[alsa-devel] [PATCH 3/3] ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR
by Philipp Zabel 19 Mar '09

19 Mar '09

Those macros are just screwed as soon as CONFIG_PXA25x is enabled. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 30 ++++++++++++++++++++++++++---- 1 files changed, 26 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c7c1996..cc8dbd4 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -283,9 +283,32 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) static void ssp_set_scr(struct ssp_dev *dev, u32 div) { struct ssp_device *ssp = dev->ssp; - u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + ssp_write_reg(ssp, SSCR0, sscr0); +} - ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); +/** + * ssp_get_clkdiv - get SSP clock divider + */ +static u32 ssp_get_scr(struct ssp_dev *dev) +{ + struct ssp_device *ssp = dev->ssp; + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 div; + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; } /* @@ -674,8 +697,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_I2S: sspsp = ssp_read_reg(ssp, SSPSP); - if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && - (width == 16)) { + if ((ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. -- 1.6.2

2 3

[alsa-devel] [PATCH] ASoC: Declare Headset as Mic and Headphone widgets for SDP3430
by Lopez Cruz, Misael 19 Mar '09

19 Mar '09

Headset was declared previously as a Headphone widget connecting HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver. The capture path becomes invalid as the Headphone widget is not a valid input endpoint. Instead of that, the Headset is declared as separate Microphone and Headphone widgets. Current patch modifies audio map: - Headset Mic: HSMIC with bias - Headset Stereophone: HSOL, HSOR Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/sdp3430.c | 24 ++++++++++++++++-------- 1 files changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 0a41de6..10f1c86 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -90,8 +90,12 @@ static struct snd_soc_jack hs_jack; /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin hs_jack_pins[] = { { - .pin = "Headset Jack", - .mask = SND_JACK_HEADSET, + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, }, }; @@ -109,7 +113,8 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIC("Ext Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_HP("Headset Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -123,11 +128,13 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HFL"}, {"Ext Spk", NULL, "HFR"}, - /* Headset: HSMIC (with bias), HSOL, HSOR */ - {"Headset Jack", NULL, "HSOL"}, - {"Headset Jack", NULL, "HSOR"}, + /* Headset Mic: HSMIC with bias */ {"HSMIC", NULL, "Headset Mic Bias"}, - {"Headset Mic Bias", NULL, "Headset Jack"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Headset Stereophone (Headphone): HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, }; static int sdp3430_twl4030_init(struct snd_soc_codec *codec) @@ -146,7 +153,8 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) /* SDP3430 connected pins */ snd_soc_dapm_enable_pin(codec, "Ext Mic"); snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); /* TWL4030 not connected pins */ snd_soc_dapm_nc_pin(codec, "AUXL"); -- 1.5.4.3

2 1

[alsa-devel] [PATCH 2/3] ASoC: Add Magician machine support
by Philipp Zabel 19 Mar '09

19 Mar '09

HTC Magician has a Philips UDA1380 codec connected via SSP1 (playback) and I2S (capture). There is a flip-flop between the SSP frame clock output and the codec's word select input pin. To make the codec see proper I2S input, the SSP has to send two frames per sample. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/Kconfig | 10 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 560 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 572 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/magician.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 5998ab3..ad8a10f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -116,6 +116,16 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_PXA2XX_SOC_MAGICIAN + tristate "SoC Audio support for HTC Magician" + depends on SND_PXA2XX_SOC && MACH_MAGICIAN + select SND_PXA2XX_SOC_I2S + select SND_PXA_SOC_SSP + select SND_SOC_UDA1380 + help + Say Y if you want to add support for SoC audio on the + HTC Magician. + config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 8ed881c..4b90c3c 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -20,6 +20,7 @@ snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o +snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o @@ -31,5 +32,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c new file mode 100644 index 0000000..f7c4544 --- /dev/null +++ b/sound/soc/pxa/magician.c @@ -0,0 +1,560 @@ +/* + * SoC audio for HTC Magician + * + * Copyright (c) 2006 Philipp Zabel <philipp.zabel(a)gmail.com> + * + * based on spitz.c, + * Authors: Liam Girdwood <lrg(a)slimlogic.co.uk> + * Richard Purdie <richard(a)openedhand.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/magician.h> +#include <asm/mach-types.h> +#include "../codecs/uda1380.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-i2s.h" +#include "pxa-ssp.h" + +#define MAGICIAN_MIC 0 +#define MAGICIAN_MIC_EXT 1 + +static int magician_hp_switch; +static int magician_spk_switch = 1; +static int magician_in_sel = MAGICIAN_MIC; + +static void magician_ext_control(struct snd_soc_codec *codec) +{ + if (magician_spk_switch) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + if (magician_hp_switch) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + switch (magician_in_sel) { + case MAGICIAN_MIC: + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case MAGICIAN_MIC_EXT: + snd_soc_dapm_disable_pin(codec, "Call Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + break; + } + + snd_soc_dapm_sync(codec); +} + +static int magician_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->card->codec; + + /* check the jack status at stream startup */ + magician_ext_control(codec); + + return 0; +} + +/* + * Magician uses SSP port for playback. + */ +static int magician_playback_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 acps, acds, width, rate; + unsigned int div4 = PXA_SSP_CLK_SCDB_4; + int ret = 0; + + rate = params_rate(params); + width = snd_pcm_format_physical_width(params_format(params)); + + /* + * rate = SSPSCLK / (2 * width(16 or 32)) + * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) + */ + switch (params_rate(params)) { + case 8000: + /* off by a factor of 2: bug in the PXA27x audio clock? */ + acps = 32842000; + switch (width) { + case 16: + /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_16; + break; + case 32: + /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_8; + } + break; + case 11025: + acps = 5622000; + switch (width) { + case 16: + /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_4; + break; + case 32: + /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + } + break; + case 22050: + acps = 5622000; + switch (width) { + case 16: + /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 44100: + acps = 5622000; + switch (width) { + case 16: + /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 48000: + acps = 12235000; + switch (width) { + case 16: + /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 96000: + acps = 12235000; + switch (width) { + case 16: + /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + break; + case 32: + /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + div4 = PXA_SSP_CLK_SCDB_1; + break; + } + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + if (ret < 0) + return ret; + + /* set audio clock as clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set the SSP audio system clock ACDS divider */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + PXA_SSP_AUDIO_DIV_ACDS, acds); + if (ret < 0) + return ret; + + /* set the SSP audio system clock SCDB divider4 */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + PXA_SSP_AUDIO_DIV_SCDB, div4); + if (ret < 0) + return ret; + + /* set SSP audio pll clock */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps); + if (ret < 0) + return ret; + + return 0; +} + +/* + * Magician uses I2S for capture. + */ +static int magician_capture_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 ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the I2S system clock as output */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops magician_capture_ops = { + .startup = magician_startup, + .hw_params = magician_capture_hw_params, +}; + +static struct snd_soc_ops magician_playback_ops = { + .startup = magician_startup, + .hw_params = magician_playback_hw_params, +}; + +static int magician_get_hp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_hp_switch; + return 0; +} + +static int magician_set_hp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (magician_hp_switch == ucontrol->value.integer.value[0]) + return 0; + + magician_hp_switch = ucontrol->value.integer.value[0]; + magician_ext_control(codec); + return 1; +} + +static int magician_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_spk_switch; + return 0; +} + +static int magician_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (magician_spk_switch == ucontrol->value.integer.value[0]) + return 0; + + magician_spk_switch = ucontrol->value.integer.value[0]; + magician_ext_control(codec); + return 1; +} + +static int magician_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_in_sel; + return 0; +} + +static int magician_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (magician_in_sel == ucontrol->value.integer.value[0]) + return 0; + + magician_in_sel = ucontrol->value.integer.value[0]; + + switch (magician_in_sel) { + case MAGICIAN_MIC: + gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1); + break; + case MAGICIAN_MIC_EXT: + gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0); + } + + return 1; +} + +static int magician_spk_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int magician_hp_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int magician_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +/* magician machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power), + SND_SOC_DAPM_SPK("Speaker", magician_spk_power), + SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias), + SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias), +}; + +/* magician machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Headphone connected to VOUTL, VOUTR */ + {"Headphone Jack", NULL, "VOUTL"}, + {"Headphone Jack", NULL, "VOUTR"}, + + /* Speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* Mics are connected to VINM */ + {"VINM", NULL, "Headset Mic"}, + {"VINM", NULL, "Call Mic"}, +}; + +static const char *input_select[] = {"Call Mic", "Headset Mic"}; +static const struct soc_enum magician_in_sel_enum = + SOC_ENUM_SINGLE_EXT(2, input_select); + +static const struct snd_kcontrol_new uda1380_magician_controls[] = { + SOC_SINGLE_BOOL_EXT("Headphone Switch", + (unsigned long)&magician_hp_switch, + magician_get_hp, magician_set_hp), + SOC_SINGLE_BOOL_EXT("Speaker Switch", + (unsigned long)&magician_spk_switch, + magician_get_spk, magician_set_spk), + SOC_ENUM_EXT("Input Select", magician_in_sel_enum, + magician_get_input, magician_set_input), +}; + +/* + * Logic for a uda1380 as connected on a HTC Magician + */ +static int magician_uda1380_init(struct snd_soc_codec *codec) +{ + int err; + + /* NC codec pins */ + snd_soc_dapm_nc_pin(codec, "VOUTLHP"); + snd_soc_dapm_nc_pin(codec, "VOUTRHP"); + + /* FIXME: is anything connected here? */ + snd_soc_dapm_nc_pin(codec, "VINL"); + snd_soc_dapm_nc_pin(codec, "VINR"); + + /* Add magician specific controls */ + err = snd_soc_add_controls(codec, uda1380_magician_controls, + ARRAY_SIZE(uda1380_magician_controls)); + if (err < 0) + return err; + + /* Add magician specific widgets */ + snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + + /* Set up magician specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + return 0; +} + +/* magician digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link magician_dai[] = { +{ + .name = "uda1380", + .stream_name = "UDA1380 Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK], + .init = magician_uda1380_init, + .ops = &magician_playback_ops, +}, +{ + .name = "uda1380", + .stream_name = "UDA1380 Capture", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE], + .ops = &magician_capture_ops, +} +}; + +/* magician audio machine driver */ +static struct snd_soc_card snd_soc_card_magician = { + .name = "Magician", + .dai_link = magician_dai, + .num_links = ARRAY_SIZE(magician_dai), + .platform = &pxa2xx_soc_platform, +}; + +/* magician audio private data */ +static struct uda1380_setup_data magician_uda1380_setup = { + .i2c_address = 0x18, + .dac_clk = UDA1380_DAC_CLK_WSPLL, +}; + +/* magician audio subsystem */ +static struct snd_soc_device magician_snd_devdata = { + .card = &snd_soc_card_magician, + .codec_dev = &soc_codec_dev_uda1380, + .codec_data = &magician_uda1380_setup, +}; + +static struct platform_device *magician_snd_device; + +static int __init magician_init(void) +{ + int ret; + + if (!machine_is_magician()) + return -ENODEV; + + ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER"); + if (ret) + goto err_request_power; + ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET"); + if (ret) + goto err_request_reset; + ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); + if (ret) + goto err_request_spk; + ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER"); + if (ret) + goto err_request_ep; + ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER"); + if (ret) + goto err_request_mic; + ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0"); + if (ret) + goto err_request_in_sel0; + ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1"); + if (ret) + goto err_request_in_sel1; + + gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1); + gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0); + + /* we may need to have the clock running here - pH5 */ + gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1); + udelay(5); + gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0); + + magician_snd_device = platform_device_alloc("soc-audio", -1); + if (!magician_snd_device) { + ret = -ENOMEM; + goto err_pdev; + } + + platform_set_drvdata(magician_snd_device, &magician_snd_devdata); + magician_snd_devdata.dev = &magician_snd_device->dev; + ret = platform_device_add(magician_snd_device); + if (ret) { + platform_device_put(magician_snd_device); + goto err_pdev; + } + + return 0; + +err_pdev: + gpio_free(EGPIO_MAGICIAN_IN_SEL1); +err_request_in_sel1: + gpio_free(EGPIO_MAGICIAN_IN_SEL0); +err_request_in_sel0: + gpio_free(EGPIO_MAGICIAN_MIC_POWER); +err_request_mic: + gpio_free(EGPIO_MAGICIAN_EP_POWER); +err_request_ep: + gpio_free(EGPIO_MAGICIAN_SPK_POWER); +err_request_spk: + gpio_free(EGPIO_MAGICIAN_CODEC_RESET); +err_request_reset: + gpio_free(EGPIO_MAGICIAN_CODEC_POWER); +err_request_power: + return ret; +} + +static void __exit magician_exit(void) +{ + platform_device_unregister(magician_snd_device); + + gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0); + + gpio_free(EGPIO_MAGICIAN_IN_SEL1); + gpio_free(EGPIO_MAGICIAN_IN_SEL0); + gpio_free(EGPIO_MAGICIAN_MIC_POWER); + gpio_free(EGPIO_MAGICIAN_EP_POWER); + gpio_free(EGPIO_MAGICIAN_SPK_POWER); + gpio_free(EGPIO_MAGICIAN_CODEC_RESET); + gpio_free(EGPIO_MAGICIAN_CODEC_POWER); +} + +module_init(magician_init); +module_exit(magician_exit); + +MODULE_AUTHOR("Philipp Zabel"); +MODULE_DESCRIPTION("ALSA SoC Magician"); +MODULE_LICENSE("GPL"); -- 1.6.2

1 0

[alsa-devel] [PATCH 1/3] ASoC: pxa-ssp: Use 16-bit DMA for magician stereo
by Philipp Zabel 19 Mar '09

19 Mar '09

Now magician and similar boards can use network mode with only one active slot to explicitly set 16 bit frame width, even for S16_LE stereo sound. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 12 +++++++++--- 1 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b0bf409..c7c1996 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -627,12 +627,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); + int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; /* select correct DMA params */ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) dma = 1; /* capture DMA offset is 1,3 */ - if (chn == 2) - dma += 2; /* stereo DMA offset is 2, mono is 0 */ + /* Network mode with one active slot (ttsa == 1) can be used + * to force 16-bit frame width on the wire (for S16_LE), even + * with two channels. Use 16-bit DMA transfers for this case. + */ + if (((chn == 2) && (ttsa != 1)) || (width == 32)) + dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */ + cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); @@ -712,7 +718,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !ttsa) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } -- 1.6.2

1 0

[alsa-devel] soc underrun problem
by dfoley 19 Mar '09

19 Mar '09

I'm currently having a problem with playing wave files root@tsi-tpcxx:~# aplay /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono mapped channel 10 to 2 underrun!!! (at least 130.662 ms long) underrun!!! (at least 140.809 ms long) I've narrowed it down to the commit ed3da3d9a0ef13c6fe1414ec73c9c1be12747b62 in linux-next/sound-2.6-tiwai.git. ALSA: Rewrite hw_ptr updaters The audio works fine, when I reverse this commit. My hardware is: CPU s3c2410 @ 202.8Mhz codecs (ak4550 or tlv320) Anyone else seeing this ?

2 4

[alsa-devel] [PATCH 1/1] echomixer: little bug fix
by Giuliano Pochini 19 Mar '09

19 Mar '09

This patch fixes a stupid bug that was introduced in the last change. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-tools-1.0.19/echomixer/echomixer.c__orig 2009-03-18 23:08:45.000000000 +0100 +++ alsa-tools-1.0.19/echomixer/echomixer.c 2009-03-18 23:54:49.000000000 +0100 @@ -1247,7 +1247,7 @@ void Vmixer_volume_changed(GtkWidget *wi int o, v; channel=(int)(long)ch; - val=INVERT((int)GTK_ADJUSTMENT(widget)->value); + val=rval=INVERT((int)GTK_ADJUSTMENT(widget)->value); #ifdef REVERSE v=channel; -- Giuliano.

1 0

[alsa-devel] [PATCH 3/3] Echoaudio: add support for Indigo express cards
by Giuliano Pochini 19 Mar '09

19 Mar '09

This patch adds to alsa-firmware the firmware needed by the IndigoIOx and IndigoDJx cards. It's too long (134K) to post here inline. Please download the patch here: Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> http://www.webalice.it/g_pochini/alsa-firmware_indigox.patch or http://www.webalice.it/g_pochini/alsa-firmware_indigox.patch.bz2 -- Giuliano.

1 0

[alsa-devel] [PATCH 2/3] Echoaudio: add support for Indigo express cards
by Giuliano Pochini 19 Mar '09

19 Mar '09

This patch adds support for IndigoIOx and IndigoDJx. (alsa-driver part) Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-driver-1.0.19__orig/pci/echoaudio/Makefile 2009-01-19 12:17:17.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/Makefile 2009-03-15 17:14:39.000000000 +0100 @@ -11,7 +11,8 @@ include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf clean-files := echoaudio.c darla20.c darla24.c echo3g.c gina20.c gina24.c \ - indigo.c indigodj.c indigoio.c layla20.c layla24.c mia.c mona.c + indigo.c indigodj.c indigoio.c indigodjx.c indigoiox.c \ + layla20.c layla24.c mia.c mona.c include $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/Makefile @@ -26,6 +27,8 @@ gina24.c: gina24.patch $(SND_TOPDIR)/als indigo.c: indigo.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigo.c echoaudio.c indigodj.c: indigodj.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigodj.c echoaudio.c indigoio.c: indigoio.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigoio.c echoaudio.c +indigodjx.c: indigodjx.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigodjx.c echoaudio.c +indigoiox.c: indigoiox.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigoiox.c echoaudio.c layla20.c: layla20.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/layla20.c echoaudio.c layla24.c: layla24.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/layla24.c echoaudio.c mia.c: mia.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/mia.c echoaudio.c --- alsa-driver-1.0.19__orig/pci/echoaudio/indigoiox.patch 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigoiox.patch 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,18 @@ +--- ../../alsa-kernel/pci/echoaudio/indigoiox.c 2009-03-01 15:26:34.000000000 +0100 ++++ indigoiox.c 2009-03-01 15:32:56.000000000 +0100 +@@ -1,3 +1,9 @@ ++#include "adriver.h" ++ ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0) ++#define pci_device(chip) pci_name(chip->pci) ++#endif ++ + /* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> +@@ -107,3 +113,5 @@ static struct snd_pcm_hardware pcm_hardw + #include "echoaudio_dsp.c" + #include "echoaudio.c" + ++ ++EXPORT_NO_SYMBOLS; --- alsa-driver-1.0.19__orig/pci/echoaudio/indigodjx.patch 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigodjx.patch 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,18 @@ +--- ../../alsa-kernel/pci/echoaudio/indigodjx.c 2009-03-01 15:26:27.000000000 +0100 ++++ indigodjx.c 2009-03-01 19:02:55.000000000 +0100 +@@ -1,3 +1,9 @@ ++#include "adriver.h" ++ ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0) ++#define pci_device(chip) pci_name(chip->pci) ++#endif ++ + /* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> +@@ -105,3 +111,5 @@ static struct snd_pcm_hardware pcm_hardw + #include "indigo_express_dsp.c" + #include "echoaudio_dsp.c" + #include "echoaudio.c" ++ ++EXPORT_NO_SYMBOLS; --- alsa-driver-1.0.19__orig/pci/echoaudio/indigo_express_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigo_express_dsp.c 2009-03-15 17:21:56.000000000 +0100 @@ -0,0 +1 @@ +#include "../../alsa-kernel/pci/echoaudio/indigo_express_dsp.c" --- alsa-driver-1.0.19__orig/pci/echoaudio/indigoiox_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigoiox_dsp.c 2009-03-15 17:19:36.000000000 +0100 @@ -0,0 +1 @@ +#include "../../alsa-kernel/pci/echoaudio/indigoiox_dsp.c" --- alsa-driver-1.0.19__orig/pci/echoaudio/indigodjx_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigodjx_dsp.c 2009-03-15 17:19:18.000000000 +0100 @@ -0,0 +1 @@ +#include "../../alsa-kernel/pci/echoaudio/indigodjx_dsp.c" -- Giuliano.

1 0

[alsa-devel] [PATCH 1/3] Echoaudio: add support for Indigo express cards
by Giuliano Pochini 19 Mar '09

19 Mar '09

This patch adds support for IndigoIOx and IndigoDJx. (alsa-kernel part) Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.h alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio_dsp.h --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.h 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio_dsp.h 2009-03-15 17:38:47.000000000 +0100 @@ -576,8 +576,13 @@ SET_LAYLA24_FREQUENCY_REG command. #define E3G_ASIC_NOT_LOADED 0xffff #define E3G_BOX_TYPE_MASK 0xf0 -#define EXT_3GBOX_NC 0x01 -#define EXT_3GBOX_NOT_SET 0x02 +/* Indigo express control register values */ +#define INDIGO_EXPRESS_32000 0x02 +#define INDIGO_EXPRESS_44100 0x01 +#define INDIGO_EXPRESS_48000 0x00 +#define INDIGO_EXPRESS_DOUBLE_SPEED 0x10 +#define INDIGO_EXPRESS_QUAD_SPEED 0x04 +#define INDIGO_EXPRESS_CLOCK_MASK 0x17 /* diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.h 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.h 2009-03-15 17:14:39.000000000 +0100 @@ -189,6 +189,9 @@ #define INDIGO 0x0090 #define INDIGO_IO 0x00a0 #define INDIGO_DJ 0x00b0 +#define DC8 0x00c0 +#define INDIGO_IOX 0x00d0 +#define INDIGO_DJX 0x00e0 #define ECHO3G 0x0100 diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx.c alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx.c 2009-03-18 22:22:49.000000000 +0100 @@ -0,0 +1,107 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_DJX +#define ECHOCARD_NAME "Indigo DJx" +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 0 */ +#define PX_DIGITAL_IN 8 /* 0 */ +#define PX_NUM 8 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 4 */ +#define BX_DIGITAL_OUT 4 /* 0 */ +#define BX_ANALOG_IN 4 /* 0 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_djx_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_DJX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_djx_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 4, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigodjx_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx_dsp.c 2009-03-18 22:14:24.000000000 +0100 @@ -0,0 +1,68 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini(a)shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo DJx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) + return -ENODEV; + + err = init_dsp_comm_page(chip); + if (err < 0) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + err = load_firmware(chip); + if (err < 0) + return err; + chip->bad_board = FALSE; + + err = init_line_levels(chip); + if (err < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_express_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigo_express_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_express_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigo_express_dsp.c 2009-03-18 22:15:38.000000000 +0100 @@ -0,0 +1,119 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini(a)shiny.it> + +*************************************************************************/ + +static int set_sample_rate(struct echoaudio *chip, u32 rate) +{ + u32 clock, control_reg, old_control_reg; + + if (wait_handshake(chip)) + return -EIO; + + old_control_reg = le32_to_cpu(chip->comm_page->control_register); + control_reg = old_control_reg & ~INDIGO_EXPRESS_CLOCK_MASK; + + switch (rate) { + case 32000: + clock = INDIGO_EXPRESS_32000; + break; + case 44100: + clock = INDIGO_EXPRESS_44100; + break; + case 48000: + clock = INDIGO_EXPRESS_48000; + break; + case 64000: + clock = INDIGO_EXPRESS_32000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 88200: + clock = INDIGO_EXPRESS_44100|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 96000: + clock = INDIGO_EXPRESS_48000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + default: + return -EINVAL; + } + + control_reg |= clock; + if (control_reg != old_control_reg) { + chip->comm_page->control_register = cpu_to_le32(control_reg); + chip->sample_rate = rate; + clear_handshake(chip); + return send_vector(chip, DSP_VC_UPDATE_CLOCKS); + } + return 0; +} + + + +/* This function routes the sound from a virtual channel to a real output */ +static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, + int gain) +{ + int index; + + if (snd_BUG_ON(pipe >= num_pipes_out(chip) || + output >= num_busses_out(chip))) + return -EINVAL; + + if (wait_handshake(chip)) + return -EIO; + + chip->vmixer_gain[output][pipe] = gain; + index = output * num_pipes_out(chip) + pipe; + chip->comm_page->vmixer[index] = gain; + + DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + return 0; +} + + + +/* Tell the DSP to read and update virtual mixer levels in comm page. */ +static int update_vmixer_level(struct echoaudio *chip) +{ + if (wait_handshake(chip)) + return -EIO; + clear_handshake(chip); + return send_vector(chip, DSP_VC_SET_VMIXER_GAIN); +} + + + +static u32 detect_input_clocks(const struct echoaudio *chip) +{ + return ECHO_CLOCK_BIT_INTERNAL; +} + + + +/* The IndigoIO has no ASIC. Just do nothing */ +static int load_asic(struct echoaudio *chip) +{ + return 0; +} diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox.c alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox.c 2009-03-18 22:22:35.000000000 +0100 @@ -0,0 +1,109 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_IOX +#define ECHOCARD_NAME "Indigo IOx" +#define ECHOCARD_HAS_MONITOR +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 2 */ +#define PX_DIGITAL_IN 10 /* 0 */ +#define PX_NUM 10 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 2 */ +#define BX_DIGITAL_OUT 2 /* 0 */ +#define BX_ANALOG_IN 2 /* 2 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_iox_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_IOX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_iox_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 8, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigoiox_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" + diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox_dsp.c 2009-03-18 22:14:44.000000000 +0100 @@ -0,0 +1,68 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini(a)shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo IOx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) + return -ENODEV; + + err = init_dsp_comm_page(chip); + if (err < 0) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + err = load_firmware(chip); + if (err < 0) + return err; + chip->bad_board = FALSE; + + err = init_line_levels(chip); + if (err < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/Makefile alsa-driver-1.0.19/sound/pci/echoaudio/Makefile --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/Makefile 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/Makefile 2009-03-15 17:14:39.000000000 +0100 @@ -15,6 +15,8 @@ snd-echo3g-objs := echo3g.o snd-indigo-objs := indigo.o snd-indigoio-objs := indigoio.o snd-indigodj-objs := indigodj.o +snd-indigoiox-objs := indigoiox.o +snd-indigodjx-objs := indigodjx.o obj-$(CONFIG_SND_DARLA20) += snd-darla20.o obj-$(CONFIG_SND_GINA20) += snd-gina20.o @@ -28,3 +30,5 @@ obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o obj-$(CONFIG_SND_INDIGO) += snd-indigo.o obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o +obj-$(CONFIG_SND_INDIGOIOX) += snd-indigoiox.o +obj-$(CONFIG_SND_INDIGODJX) += snd-indigodjx.o --- alsa-driver-1.0.19__orig/alsa-kernel/pci/Kconfig 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/Kconfig 2009-03-15 17:14:39.000000000 +0100 @@ -400,6 +400,26 @@ config SND_INDIGODJ To compile this driver as a module, choose M here: the module will be called snd-indigodj +config SND_INDIGOIOX + tristate "(Echoaudio) Indigo IOx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo IOx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigoiox + +config SND_INDIGODJX + tristate "(Echoaudio) Indigo DJx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo DJx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigodjx + config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" select FW_LOADER -- Giuliano.

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[alsa-devel] [2.6.30][topic/asoc][PATCH 1/2] ASoC: OMAP: N810: Mark not connected input pins
by Jarkko Nikula 18 Mar '09

18 Mar '09

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/n810.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 86471fd..f203221 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -254,6 +254,11 @@ static int n810_aic33_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); snd_soc_dapm_nc_pin(codec, "HPLCOM"); snd_soc_dapm_nc_pin(codec, "HPRCOM"); + snd_soc_dapm_nc_pin(codec, "MIC3L"); + snd_soc_dapm_nc_pin(codec, "MIC3R"); + snd_soc_dapm_nc_pin(codec, "LINE1R"); + snd_soc_dapm_nc_pin(codec, "LINE2L"); + snd_soc_dapm_nc_pin(codec, "LINE2R"); /* Add N810 specific controls */ err = snd_soc_add_controls(codec, aic33_n810_controls, -- 1.6.2

2 2

Re: [alsa-devel] DELL SP2309W Monitor with Integrated microphone
by Takashi Iwai 18 Mar '09

18 Mar '09

At Wed, 18 Mar 2009 08:24:09 -0700, Jason Holmes wrote: > > No the device doesn't work either when using plughw:Webcam. What error did you get? Takashi

2 5

[alsa-devel] [PATCH 0/9] ARM: DaVinci: ASoC: Adds Audio support for TI DM646X
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds Audio support for TI DM646X EVM platform Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch set adds audio support for DM646x This series is dependant on the following 2 patches submitted to the DaVinci git: [PATCH 1/2] ARM: DaVinci: EDMA: Move queue to tc mapping to dm<soc>.c [PATCH 2/2] ARM: DaVinci: EDMA: Enable EDMA support for dm646x Patch 1: Adds ASoC driver support for TI DM646X EVM platform Applies to the DaVinci git. Patch 2: Adds ALSA SoC McASP Audio Layer for TI DM646X processor Applies to the ASoC tree. Patch 3: Adds header file for McASP Audio Layer for TI DM646X processor Applies to the ASoC tree. Patch 4: Adds ALSA PCM interface for the TI DM646X processor Applies to the ASoC tree. Patch 5: Adds Audio Serial Port support for TI DM646X processor Applies to the DaVinci git. Patch 6: Adds McASP clock support for TI DM646X processor Applies to the DaVinci git. Patch 7: Adds ALSA SoC DIT driver support for DM646X Applies to the ASoC tree. Patch 8: ASoC: Adds header file for DM646X DIT driver Applies to the ASoC tree. Patch 9: Adds build support for DM646X audio driver Applies to the ASoC tree. These patches are based on the work done by Steve Chen <schen(a)mvista.com> on Monstavista 2.6.18 kernel. Even without these patches the ASoC tree (http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…) build for DaVinci is broken.

3 2

[alsa-devel] [PATCH 9/9] ARM: DaVinci: ASoC: Adds build support for DM646X audio driver
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds build support for DM646X audio driver Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/codecs/Kconfig | 3 +++ sound/soc/codecs/Makefile | 2 ++ sound/soc/davinci/Kconfig | 14 ++++++++++++++ sound/soc/davinci/Makefile | 3 +++ 4 files changed, 22 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f33c07..d34b38b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,9 @@ config SND_SOC_TLV320AIC23 tristate depends on I2C +config SND_SOC_CODEC_STUBS + tristate + config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9c61037..3d1d270 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -12,6 +12,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-codec_stubs-objs := codec_stubs.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o @@ -54,6 +55,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_CODEC_STUBS) += snd-soc-codec_stubs.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index b502741..ec554c9 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_I2S_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM @@ -18,6 +21,17 @@ config SND_DAVINCI_SOC_EVM Say Y if you want to add support for SoC audio on TI DaVinci EVM platform. +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM646X_EVM + select SND_DAVINCI_SOC_I2S_MCASP + select SND_SOC_TLV320AIC3X + select SND_SOC_CODEC_STUBS + help + Say Y if you want to add support for SoC audio on TI + DaVinci DM646x EVM platform. + + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1..152d59c 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,16 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-i2s-mcasp-objs:= davinci-i2s-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S_MCASP) += snd-soc-davinci-i2s-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o -- 1.5.6

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[alsa-devel] [PATCH 7/9] ARM: DaVinci: ASoC: Adds ALSA SoC DIT/DIR driver support for DM646X
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds ALSA SoC DIT/DIR driver support for DM646X Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/codecs/codec_stubs.c | 63 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 63 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/codec_stubs.c diff --git a/sound/soc/codecs/codec_stubs.c b/sound/soc/codecs/codec_stubs.c new file mode 100644 index 0000000..60d3446 --- /dev/null +++ b/sound/soc/codecs/codec_stubs.c @@ -0,0 +1,63 @@ +/* + * ALSA SoC DaVinci DIT/DIR driver + * + * TI DaVinci audio controller can operate in DIT/DIR (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> +#include <sound/pcm.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai dit_stub_dai[] = { + { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + } +}; + +struct snd_soc_dai dir_stub_dai[] = { + { + .name = "DIR", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + } +}; + +static int __init dit_modinit(void) +{ + return snd_soc_register_dai(dit_stub_dai); +} +module_init(dit_modinit); + +static void __exit dit_exit(void) +{ + snd_soc_unregister_dai(dit_stub_dai); +} +module_exit(dit_exit); + -- 1.5.6

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Re: [alsa-devel] [PATCH 1/2] Echoaudio: Add support for Indigo express cards
by Takashi Iwai 18 Mar '09

18 Mar '09

Hi Giuliano, At Mon, 16 Mar 2009 09:14:20 +0100 (CET), Giuliano Pochini wrote: > > On Mon, 16 Mar 2009, Takashi Iwai wrote: > > >> This patch adds support for IndigoIOx and IndigoDJx. > >> > >> Signed-off-by Giuliano Pochini <pochini(a)shiny.it> > > > > The code looks almost good, but could you do the following? > > > > - split to two patches, one for kernel tree and one for alsa-driver > > tree > > There is a little problem: since about a month ago the git server is > SLOWWW. It takes ages to download the alsa-kernel repository at 5KB/s :( > I'll try from some server at work which stay on 24/7. > > > > - run checkpatch.pl against the patch and fix issues reported there > > Ok ! Any progress on this? I'm asking it because 2.6.29 kernel will likely go out soon, and all patches should be merged before the merge-window is opened. If you have no time, either we skip it or I'll do a quick fix. thanks, Takashi

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[alsa-devel] [PATCH 3/9] ARM: DaVinci: ASoC: Adds header file for McASP Audio Layer for TI DM646X processor
by Naresh Medisetty 18 Mar '09

18 Mar '09

Adds header file for McASP Audio Layer for TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/davinci/davinci-i2s-mcasp.h | 312 +++++++++++++++++++++++++++++++++ 1 files changed, 312 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.h diff --git a/sound/soc/davinci/davinci-i2s-mcasp.h b/sound/soc/davinci/davinci-i2s-mcasp.h new file mode 100644 index 0000000..7d2f7bb --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.h @@ -0,0 +1,312 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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 DAVINCI_I2S_MCASP_H +#define DAVINCI_I2S_MCASP_H + +#include <linux/io.h> +#include "davinci-pcm.h" + + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left (even TDM Slot) Channel Status Register File*/ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right (odd TDM slot) Channel Status RegisterFile*/ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left (even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right (odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define FREE BIT(0) +#define SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 +#define DISMOD(val) (val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ + /* and Transmit Bad Clock Detect /32 Counter + Reset */ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + + +extern struct snd_soc_dai davinci_iis_mcasp_dai[]; +extern struct snd_soc_dai davinci_dit_mcasp_dai; + + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + + + +#endif /* DAVINCI_I2S_MCASP_H */ -- 1.5.6

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[alsa-devel] Correct macro for Headset widgets
by Lopez Cruz, Misael 18 Mar '09

18 Mar '09

What is the correct way to declare a Headset widget in a machine driver? I originally thought that by using SND_SOC_DAPM_HP macro was enough, but it's not since the headset is both headphones and microphone. In my machine driver I have the following widget and map for headset: SND_SOC_DAPM_HP("Headset Jack", NULL) ... /* Headset: HSMIC (with bias), HSOL, HSOR */ {"Headset Jack", NULL, "HSOL"}, {"Headset Jack", NULL, "HSOR"}, {"HSMIC", NULL, "Headset Mic Bias"}, {"Headset Mic Bias", NULL, "Headset Jack"}, but the "Headset Mic Bias" widget never gets powered when the "Headset Jack" pin is enabled. When the audio map is scanned, only valid routes are powered: DAC-Output, Input-DAC, Input-Output, DAC-ADC. And "Headset Jack" (declared as SND_SOC_DAPM_HP) is not a valid input endpoint, then the capture route becomes invalid. Can we have a macro specifically for Headset widgets? Thanks, -Misa

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[alsa-devel] DELL SP2309W Monitor with Integrated microphone
by Jason Holmes 18 Mar '09

18 Mar '09

Hi all, I have a Dell SP2309W monitor which contains an integrated webcam and microphones. I have been unable to get the integrated microphones to record any sound. I ran utils_alsa-info.sh and the link with the results in posted below: http://www.alsa-project.org/db/?f=590e122a466ce09cd1f911ba7146a87df2b5cbeb The integrated microphone seems to be made by OmniVision Technologies. The results of running arecord -l are: **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Webcam [Monitor Webcam], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: default [USB Sound Device ], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 and the results of arecord --device=default:CARD=Webcam test are: ALSA lib pcm_direct.c:866:(snd1_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dsnoop.c:604:(snd_pcm_dsnoop_open) unable to initialize slave arecord: main:590: audio open error: Invalid argument Thanks for your help. -- Jason Holmes

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[alsa-devel] Cleanup patches for snd-usb-caiaq
by Daniel Mack 18 Mar '09

18 Mar '09

Following up to this mail are 3 patches to clean up the snd-usb-caiaq driver. [PATCH 1/3] snd-usb-caiaq: only warn once on streaming errors [PATCH 2/3] snd-usb-caiaq: drop bogus iso packets [PATCH 3/3] snd-usb-caiaq: bump version number

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Re: [alsa-devel] Bad page state in process 'rmmod' cs4281
by "Timo T.Hyrsylä" 18 Mar '09

18 Mar '09

On 13.3.2009 13:04, Takashi Iwai wrote: > At Mon, 09 Mar 2009 12:24:51 +0100, > wrote: >> At Mon, 09 Mar 2009 10:32:02 +0200, >> Timo T.Hyrsylä wrote: >>> Trying to make Alsa work on our embedded arm system with ixdp425 and CS4281. >>> >>> kernel is 2.6.20.3 and the Alsas (driver+lib+utils) latest 1.0.19 >>> >>> Other than the two below errors, the system seems to work ok. >>> >>> When removing the snd_cs4281 module, dmesg reports multiple errors about >>> bad page >>> >>> <0>Bad page state in process 'rmmod' >>> <0>page:c4007300 flags:0x00004000 mapping:00000000 mapcount:0 count:0 >>> <0>Trying to fix it up, but a reboot is needed >>> <0>Backtrace: >>> <0>Bad page state in process 'rmmod' >>> <0>page:c4007320 flags:0x00004000 mapping:00000000 mapcount:0 count:0 >>> <0>Trying to fix it up, but a reboot is needed >>> <0>Backtrace: >>> >>> I've managed to track down the error upto dma_free_coherent. >>> >>> snd_pcm_free > snd_pcm_lib_preallocate_free_for_all > >>> snd_pcm_lib_preallocate_free > snd_pcm_lib_preallocate_dma_free > >>> snd_dma_free_pages > snd_free_dev_pages > dma_free_coherent >>> >>> >>> Another issue, possibly related, is that when running aplay with --mmap >>> it will crash with: >>> >>> Internal error: Oops: f3 [#1] >>> PC is at snd_pcm_mmap_data_nopage+0x10c/0x148 [snd_pcm] >>> LR is at __handle_mm_fault+0x16c/0x8ac >> This is likely the problem with dma_alloc_coherent() and mmap on >> a non-coherent architecture. I have patches for these, but only for >> recent kernels. The patches are found in test/dma-fix branch of sound >> git tree: >> git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git test/dma-fix > > Alternatively, an easier "fix" would be simply disable mmap support. > Remove SNDRV_PCM_INFO_MMAP* from snd_pcm_hardware definitions. > > This may disable the conversion plugin usage in the earlier alsa-lib > versions, though. > > > Takashi Tried to apply the patch changes into my version, but it did not affect the unloading 'bad page' problem at all. Instead it did make all play attempts (even non MMAP) to crash. (no crash details available since I desided to forget about it for now, working on unrelated issues for awhile) I did disable the MMAP so that player wont crash when requesting MMAP. System still dumps the bad pages at unload, play and capture seem to work OK, not tested extensively thou. -- Timo

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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 17 Mar '09

17 Mar '09

The following changes since commit 852fd9e50f62b4ea7afe26eee0710464de4801b8: Mark Brown (1): ASoC: Each PXA AC97 DAI needs a separate ops are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Atsushi Nemoto (1): ASoC: Only deregister AC97 dev if it's name was not "AC97" Daniel Mack (1): ASoC: switch PXA SSP driver from network mode to PSP Mark Brown (3): ASoC: Fix non-networked I2S mode for PXA SSP ASoC: Fix Zylonite for non-networked SSP mode Merge branch 'pxa-ssp' into for-2.6.30 sound/soc/pxa/pxa-ssp.c | 55 ++++++++++++++++++++++++++++++++++++---------- sound/soc/pxa/zylonite.c | 55 +++++++++++++++++++++------------------------- sound/soc/soc-core.c | 3 +- 3 files changed, 70 insertions(+), 43 deletions(-)

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[alsa-devel] Tenx USB audio adapter
by PETER EASTHOPE 17 Mar '09

17 Mar '09

Folk, I was advised to ask about the Tenx USB audio adapter here. lshal claims it is a standard HID device and a standard USB device. Ref. http://carnot.yi.org/lshal1.out Nevertheless, sound in and sound out both fail in my system. Some information is at the bottom of this page. "http://carnot.yi.org/DebianPage.html#USBAudio" Any suggestions? Thanks, ... Peter E. -- http://members.shaw.ca/peasthope/ http://carnot.yi.org/ = http://carnot.pathology.ubc.ca/

1 0

[alsa-devel] Fedora 10 x86_64 alsa problem: Realtek ALC889A Audio Codec(ATI azalia) not detected on alsa
by Emmanuel Chanel 17 Mar '09

17 Mar '09

I wrote this on https://bugzilla.redhat.com/show_bug.cgi?id=489828 I've used the driver of kernel package. And Mr. Jaroslav Kysela replied me that I could try snapshot source. (Thx, Mr. Jaroslav Kysela. And on #alsa on Freenode, Thingol and gnubien advised me much. Thank you!) But I couldn't get sound back none of both: stable(alsa-driver-1.0.19) and snapshot (alsa-driver-1.0.19.34.g70f78.363.g0ec3b). My alsa info is on http://www.alsa-project.org/db/?f=af6a6e787d4bd5212bf766bf5d09fda1ab5e2300 I use desktop. Its mother board is GA-MA69H-S3H . The soundcard is onboard. Realtek ALC889A Audio Codec. # lspci |grep -i "multi\|audio\|040[0-3]" # Audio device: ATI Technologies Inc SB600 Azalia (Intel HDA) When I installed stable release of alsa-driver --with-debug=verbose , dmesg says HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:2218: chipset global capabilities = 0x4401 ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:786: codec_mask = 0x8 ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:1214: hda_intel: codec #3 probed OK ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/../../alsa-kernel/pci/hda/hda_codec.c:788: hda_codec: no AFG or MFG node found ALSA /home/emmanuel/Softwares/alsa-driver-1.0.19/pci/hda/hda_intel.c:1298: hda-intel: no codecs initialized HDA Intel 0000:00:14.2: PCI INT A disabled -- Emmanuel Chanel <emmanuelchanel(a)gmail.com>

3 27

[alsa-devel] How to tell user level hardware not support a certain format in a certain sample rate
by Richard Zhao 17 Mar '09

17 Mar '09

Hi, runtime->hw.rates and runtime->hw.formats indicate the rate and formats hardware can support, so the user level plugins can auto-convert the audio data. But hardware may not support a certain combin of sample rate and format, and will cause hw_parm fail. In such case, plugin auto-convert will not work. How can I let userlevel know the failure conbin, so let auto-convert work? Thanks Richard

3 6

[alsa-devel] Need help on getting raw audio stream from a dummy sound card
by Santo Chow 17 Mar '09

17 Mar '09

Hi everyone :) I'm currently assigned to a task that requires me to capture raw audio stream from a dummy sound card. Is this even possible? Previously, I have been able to capture raw audio stream from my default sound card, by using the following example I got from http://www.linuxjournal.com/article/6735 /* This example reads from the default PCM device and writes to standard output for 5 seconds of data. */ /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API #include <alsa/asoundlib.h> int main() { long loops; int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer; /* Open PCM device for recording (capture). */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0); if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params); /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2); /* 44100 bits/second sampling rate (CD quality) */ val = 44100; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); /* Set period size to 32 frames. */ frames = 32; snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, &dir); size = frames * 4; /* 2 bytes/sample, 2 channels */ buffer = (char *) malloc(size); /* We want to loop for 5 seconds */ snd_pcm_hw_params_get_period_time(params, &val, &dir); loops = 5000000 / val; while (loops > 0) { loops--; rc = snd_pcm_readi(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means overrun */ fprintf(stderr, "overrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from read: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short read, read %d frames\n", rc); } rc = write(1, buffer, size); if (rc != size) fprintf(stderr, "short write: wrote %d bytes\n", rc); } snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); return 0; } The above example allows me to capture 5 seconds of raw audio. I can use aplay to play the recorded sound and it plays nicely. But right now, I'm working on a project that involves the usage of a small development board. Basically, this board does not have a sound card or an actual speaker. So I thought, I can use the dummy soundcard provided by the linux kernel by calling out modprobe snd-dummy. Right now, I'm still testing it in my PC so to simulate the board's environment. I have configured the .asoundrc file and create a new dummy pcm by putting the lines below onto my .asoundrc file. pcm.dummy{ type hw card Dummy } So, I change the above ecample code, so it listens to this new dummy pcm rather than the 'default'. Like this : rc = snd_pcm_open(&handle, "dummy", SND_PCM_STREAM_CAPTURE, 0); Supposedly, the above modification would allow me to record from this dummy pcm, no? I do this by playing a song, using aplay -f cd -D dummy song.wav and at the same time, execute the above example. As expected, no sound was coming out from my speaker. When the program finished recording for 5 seconds, I play back the result, but all i heard was just noise. Of curiousity, I try using file plugins. I modify my .asoundrc file like this: pcm.dummy{ type plug slave{ pcm file format S16_LE channels 2 rate 44100 } } pcm.file{ type file slave{ pcm d } file /home/mydir/out.raw } pcm.d{ type hw card Dummy } Then i called aplay -f cd -D dummy song.wav again, well.. still no sound coming out from the speaker, but it does output the raw audio file into my /home/mydir/out.raw file. I play the out.raw using aplay -f cd /home/mydir/out.raw it's flawless. But I can't use this for my implementation. What I need is actually a way to read raw audio data (or stream) from the dummy sound card, to a buffer inside my program. I need this because I'm going to stream the buffer to my server, so I can listen the sound from my server. I can't afford to use the file plugin approach, basically because later on, in my development board, i won't have that much space. So, the question is: capturing raw audio data from a dummy soundcard, is this possible? I'm pretty sure that if the file plugin works, means that the raw audio data is there. It's just that i'm probably doing a wrong approach to read it. Hence, needs explanation and help.. Please help me.. :( Thank you so much for reading my long request.

2 3

Re: [alsa-devel] Bugs on aspire one A150
by Maxim Levitsky 16 Mar '09

16 Mar '09

Alan Jenkins wrote: > Maxim Levitsky wrote: >> I have just bought an Aspire one A150, XP version, >> as it was the only available here, and installed ubuntu on it. >> >> Bugs I discovered so far: >> >> >> ** 1 - embedded controler works in polling mode, due to this: >> >> [ 0.708571] ACPI: EC: non-query interrupt received, switching to interrupt mode >> [ 1.224043] ACPI: EC: missing confirmations, switch off interrupt mode. >> >> >> Maybe this is the reason for the fact that gnome power manager freezes when I unplug >> the AC, and freezes often when I try to see battery status. >> >> >> (Note: same is seen on my acer aspire 5720) > > That sounds like a known issue. It has been resolved by "ACPI: EC: revert msleep patch". Happily Len submitted it for mainline this week. You will also find it if you try the acpi-test git tree. We're all hoping 2.6.28 will be much improved in terms of reliable operation of different ECs :). > Yes, this almost fixes all issues with that. Almost because, it looks like the EC changes screen brightness on his own when battery is plugged/unplugged, but so does the gnome-power-manager, and thus it still hangs as before on battery removal (but doesn't hang otherwise) I disabled that behaver in gnome-power-manager and now no more hangs. I see that this fix fixes the polled mode. Any chance to make irq mode work? >> ** 2 - wireless: not to mention the fact that ath5k wasn't installed by default in ubuntu... >> wireless more or less works, but kernel log is full of backtraces. > > Well, that doesn't tell us much. Did they still happen after upgrading to 2.6.28-rc3? Can we see them? Fixed completely, looking through the git log I have seen a commit that fixes exactly same backtrace I had. No more errors in kernel log. > >> Was able to connect to my WPA2 access point. >> Sometimes wireless fails completely, especially after suspend to ram. >> Advanced features like monitor/injection work, but when I changed the card's >> mac address it stopped working. >> I also noticed that if I then start airodump, then wireless works with new mac. This still not fixed, I ask about this again in separate mail. >> >> >> >> ** 3 - internal mic doesn't work. >> tried model=acer, model=auto. >> Overall it seems that alsa misprograms O/B realteck >> codec. >> I talk about this later. Surprisingly, this is almost fixed. I can record from both internal mic and external mic (Quality is not so good, but is the same as in windows) Almost, is that to increase capture level, I have to set "one" channel of it. Eg: if I set 100% left and 100% right then capture level is very low. But when I set 100% left and 0% right or vice versa it works fine. Also the "mic boost" doesn't change capture volume for internal mic. also the "mic boost" if set to high value works like analog loopback (I hear what I say in mic in headphones, same happens on my main notebook acer 5720) >> Have same issue on my acer 5720 >> >> >> >> ** 4 - wireless led doesn't work. >> ath5k devs, can you fix this? >> >> >> ** 5 - coretemp doesn't show cpu temperature, >> I have seen somewhere that atom support same thermal diode as core2 >> and only patch to detect it is needed. >> Please include such path in 2.6.28 if exists. >> >> >> ** 6 - both card readers are missing from lspci, is this normal? > > A similar bug has been reported as a regression: > > <http://bugzilla.kernel.org/show_bug.cgi?id=11828> > > so one assumes that it worked on the machines with linux pre-installed. Hopefully without requiring any hacks. > > It seems that for now a workaround may be to pass the option debug_quirks=1 to the sdhci module... > > <http://marc.info/?l=linux-kernel&m=122509648027303&w=2> > > ...or that it may help if you insert an SD card before booting. > > Apparently the reporter also investigated pcie hotplug. Probably the BIOS doesn't provide the normal support. You can try "modprobe pciehp pciehp_force=1", maybe it helps the kernel discover the devices. It worked for something else on my EeePC. But then it will reportedly disappear the ethernet controller. > > However, at the moment pciehp can cause delays of 10s of seconds during resume. I am aware of this, but don't yet have a SD card to test. > >> ** 7 - opengl is broken, I tried running neverball in fullscreen, but it shows wrong/corrupted >> textures (compiz is running, and working well thought) Without compiz neveball works fine (slow, but this is very slow hardware) >> >> Now I installed 2.6.28-rc3 (actually latest -git) >> >> >> Sadly nothing improved. >> I only found few regressions: >> >> >> >> * Wireless now drops connection from my WPA2 access point after few minutes. >> I changed temporarily its wireless protection to WEP and open mode, and both work >> fine, maybe this is related to periodic key exchange WPA does? Now fixed, big thanks. >> >> * System doesn't reboot/shutdown, just hangs. > > Could be this: > > <http://bugzilla.kernel.org/show_bug.cgi?id=11942> > > As a fix, 8fd145917fb62368a9b80db59562c20576238f5a was reverted from the acpi tree. This has also been submitted to mainline. Also fixed big thanks. Also noticed another bug: If I suspend/resume with compiz running, on resume I see wallpaper and mouse cursor, everything hangs for minute or two, and sometimes forever. 2.6.27 worked fine. Also hibernate doesn't work on my main notebook, but it is probably fixed, I update kernel to really latest -git and tell you. Big thanks for bugfixes, you saved me a lot of work and bisecting. Biggest solvable problem now is sound support now I think, datasheets are there, and I take a look at them. Best regards, Maxim Levitsky

3 25

[alsa-devel] [PATCH 6/9] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds McASP clock support for TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=commi…. arch/arm/mach-davinci/dm646x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index af040cf..0aee774 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -185,6 +185,18 @@ static struct clk gpio_clk = { .lpsc = DM646X_LPSC_GPIO, }; +static struct clk mcasp0_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP0, +}; + +static struct clk mcasp1_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP1, +}; + static struct clk aemif_clk = { .name = "aemif", .parent = &pll1_sysclk3, @@ -253,6 +265,8 @@ struct davinci_clk dm646x_clks[] = { CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK(NULL, "gpio", &gpio_clk), + CLK("soc-audio.0", NULL, &mcasp0_clk), + CLK("soc-audio.1", NULL, &mcasp1_clk), CLK(NULL, "aemif", &aemif_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK(NULL, "pwm0", &pwm0_clk), -- 1.5.6

2 1

[alsa-devel] [PATCH 2/9] ARM: DaVinci: ASoC: Adds ALSA SoC McASP Audio Layer for TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds ALSA SoC McASP Audio Layer for TI DM646X processor. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/davinci/davinci-i2s-mcasp.c | 641 +++++++++++++++++++++++++++++++++ 1 files changed, 641 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.c diff --git a/sound/soc/davinci/davinci-i2s-mcasp.c b/sound/soc/davinci/davinci-i2s-mcasp.c new file mode 100644 index 0000000..eb632cd --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.c @@ -0,0 +1,641 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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. + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s-mcasp.h" + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + mcasp_set_bits(regs, val); + while ((mcasp_get_reg(regs) & val) != val) + ; +} +int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + + return 0; +} +EXPORT_SYMBOL(davinci_i2s_startup); + +int davinci_i2s_mcasp_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int tmp, link_cnt; + int count = 0; + int backup_count = 0; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai; + struct davinci_audio_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *parray = pdev->dev.platform_data; + struct evm_snd_platform_data *pdata; + struct davinci_pcm_dma_params *dma_data; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev) * card->num_links, + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * + (card->num_links << 1), GFP_KERNEL); + if (!dma_data) + goto err_release_dev; + + for (link_cnt = 0; link_cnt < card->num_links; link_cnt++) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, link_cnt); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + backup_count = 0; + goto err_release_data; + } + + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + backup_count = 1; + goto err_release_data; + } + + + cpu_dai = card->dai_link[link_cnt].cpu_dai; + cpu_dai->private_data = &dev[link_cnt]; + pdata = &parray[link_cnt]; + dev[link_cnt].clk = clk_get(&pdev->dev, NULL); + + if (IS_ERR(dev[link_cnt].clk)) { + ret = -ENODEV; + backup_count = 2; + goto err_release_data; + } + clk_enable(dev[link_cnt].clk); + + dev[link_cnt].base = (void __iomem *)IO_ADDRESS(mem->start); + dev[link_cnt].op_mode = pdata->op_mode; + dev[link_cnt].tdm_slots = pdata->tdm_slots; + dev[link_cnt].num_serializer = pdata->num_serializer; + dev[link_cnt].serial_dir = pdata->serial_dir; + dev[link_cnt].codec_fmt = pdata->codec_fmt; + + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].channel = pdata->tx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_PLAYBACK] = + &dma_data[count]; + + + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].channel = pdata->rx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_CAPTURE] = + &dma_data[count]; + count++; + + } + return 0; + +err_release_data: + for (tmp = link_cnt; tmp >= 0; tmp--) { + if (backup_count > 2) + clk_disable(dev[tmp].clk); + + if (backup_count > 1) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, tmp); + release_mem_region(mem->start, + (mem->end - mem->start) + 1); + } + backup_count = 3; + } + kfree(dma_data); + +err_release_dev: + kfree(dev); + + return ret; +} +EXPORT_SYMBOL(davinci_i2s_mcasp_probe); + +void davinci_i2s_mcasp_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int i; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct davinci_audio_dev *dev_list = cpu_dai->private_data; + struct davinci_audio_dev *dev; + struct resource *mem; + struct davinci_pcm_dma_params *ptr; + + for (i = 0; i < card->num_links; i++) { + dev = &dev_list[i]; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + } + ptr = dev_list->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(ptr); + kfree(dev_list); +} +EXPORT_SYMBOL(davinci_i2s_mcasp_remove); + + +void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + /* wait for TX ready */ + while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(0)) + & TXSTATE)) + ; + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_i2s_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct snd_pcm_substream *substream, + int channel_size) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03 << 4; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05 << 4; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07 << 4; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09 << 4; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B << 4; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D << 4; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F << 4; + break; + + default: + return -1; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, fmt, (0x0F << 4)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, fmt, (0x0F << 4)); + return 0; +} + +static void davinci_hw_common_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} +static void davinci_hw_iis_param(struct snd_pcm_substream *substream) +{ + int i, active_slots; + u32 mask = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + + } else { + /* bit stream is MSB first with no delay */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + FSRDLY(1) | RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_i2s_mcasp_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(substream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(substream); + else + davinci_hw_iis_param(substream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(substream, word_length); + + return 0; +} + +static int davinci_i2s_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +struct snd_soc_dai davinci_iis_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .probe = davinci_i2s_mcasp_probe, + .remove = davinci_i2s_mcasp_remove, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + }, + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + }, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_iis_mcasp_dai); + +static int __init davinci_i2s_mcasp_init(void) +{ + return snd_soc_register_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + +} +module_init(davinci_i2s_mcasp_init); + +static void __exit davinci_i2s_mcasp_exit(void) +{ + return snd_soc_unregister_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + +} +module_exit(davinci_i2s_mcasp_exit); + + + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McASP) SoC Interface"); +MODULE_LICENSE("GPL"); + -- 1.5.6

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Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.29-rc8-519-g236e303
by Takashi Iwai 16 Mar '09

16 Mar '09

At Mon, 16 Mar 2009 15:01:57 +0100 (CET), noreply-git(a)alsa-project.org wrote: > > Hello, > > This is an automated email from the git hooks/update script, it was > generated because a ref change was pushed to the repository. > > Updating branch, master, > via 236e3036199f7ef6817967cb4aee5e6686ef2d52 (commit) > from 1cebbf3bef5ee355894375b92946911068130e22 (commit) > > - Log ----------------------------------------------------------------- > commit 236e3036199f7ef6817967cb4aee5e6686ef2d52 > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Mon Mar 16 14:56:58 2009 +0100 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Mon Mar 16 15:00:48 2009 +0100 > > ALSA: snd-hda-intel: Fix ALC662/ALC663 Beep Amplifier Index > > ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID. > Confirmed by testing on real hardware. ALC662 has really the index 0x05 although ALC663 and ALC272 have 0x04. Fixed on sound git tree now... Takashi > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> > > ----------------------------------------------------------------------- > > Diffstat: > sound/pci/hda/patch_realtek.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > > Patch -------------- (max 1500 lines) > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index b794cba..672103d 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -16951,7 +16951,7 @@ static int patch_alc662(struct hda_codec *codec) > > if (!spec->cap_mixer) > set_capture_mixer(spec); > - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); > + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); > > spec->vmaster_nid = 0x02; > > ------- End of patch > > hooks/update > --- > Git Source Code Management System > hooks/update refs/heads/master \ > 1cebbf3bef5ee355894375b92946911068130e22 \ > 236e3036199f7ef6817967cb4aee5e6686ef2d52 > > ------------------------------------------------------------------------------ > Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are > powering Web 2.0 with engaging, cross-platform capabilities. Quickly and > easily build your RIAs with Flex Builder, the Eclipse(TM)based development > software that enables intelligent coding and step-through debugging. > Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog >

1 0

[alsa-devel] [PATCH 1/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ce4b43b..5ed5f1b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- 1.5.6.3 -- - Joonyoung Shim

4 5

[alsa-devel] [PATCH RESEND 1/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ce4b43b..5ed5f1b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- 1.5.6.3

2 1

[alsa-devel] [PATCH RESEND 1/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ce4b43b..5ed5f1b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- 1.5.6.3

1 0

[alsa-devel] [PATCH 8/9] ARM: DaVinci: ASoC: Adds header file for DM646X DIT/DIR driver
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds header file for DM646X DIT/DIR driver Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/codecs/codec_stubs.h | 23 +++++++++++++++++++++++ 1 files changed, 23 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/codec_stubs.h diff --git a/sound/soc/codecs/codec_stubs.h b/sound/soc/codecs/codec_stubs.h new file mode 100644 index 0000000..693797b --- /dev/null +++ b/sound/soc/codecs/codec_stubs.h @@ -0,0 +1,23 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.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 CODEC_STUBS_H +#define CODEC_STUBS_H + +#ifdef CONFIG_SND_SOC_CODEC_STUBS +extern struct snd_soc_dai dit_stub_dai[]; +extern struct snd_soc_dai dir_stub_dai[]; +#else +#define dit_stub_dai NULL +#define dir_stub_dai NULL +#endif + +#endif /* CODEC_STUBS_H */ -- 1.5.6

1 0

[alsa-devel] [PATCH 5/9] ARM: DaVinci: ASoC: Adds Audio Serial Port support for TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds Audio Serial Port support for TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=commi…. arch/arm/mach-davinci/include/mach/asp.h | 14 ++++++++++++-- 1 files changed, 12 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..670afe9 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -6,16 +6,26 @@ #include <mach/irqs.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ + +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 + +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT -- 1.5.6

1 0

[alsa-devel] [PATCH 4/9] ARM: DaVinci: ASoC: Adds ALSA PCM interface for the TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds ALSA PCM interface for the TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/davinci/davinci-pcm.h | 22 ++++++++++++++++++---- 1 files changed, 18 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..c7d8711 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,16 +12,30 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ + enum dma_event_q eventq_no; /* event queue number */ }; struct evm_snd_platform_data { + char *clk_name; int tx_dma_ch; int rx_dma_ch; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; }; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

1 0

[alsa-devel] Using snd-aloop as default soundcard (need urgent help T.T)
by Santo Chow 16 Mar '09

16 Mar '09

Hi every one.. I'm currently on a project that requires me to capture a raw audio data from a dummy soundcard. I've so far manage to do that, by using snd-aloop. Now the problem i have is.. when i'm running several audio program at the same time, i can only hear the sound from the first program that i ran. i've followed the example os dmix, but this doesn't help either. So, right now, i have a Loopback soundcard as my default. My .asoundrc looks like this: pcm.!default{ type hw card 1 } Card 0 is my actual soundcard (that can actually produce sound). Card 1, when i checked on the /proc/asound/cards actually pointing to Loopback card. So right now, the way i test the card 1, is to call arecord from card 1, and play it to card 0. Right now, i'm still developing this in my ubuntu PC. so i ran this arecord -f cd -D hw:1,1 | aplay -f cd -D hw:0 in one terminal, and in the other terminal i can play mplayer, watching youtube, etc.. and the sound would come out nicely. (Thanks to all of your efforts, nevertheless) But I notice that, when i play mplayer, then i open a web-browser, and surf youtube, the sound from youtube is gone. But if i close the browser, close the mplayer, then reopen the browser, go to youtube and play a video, i can hear the sound nicely. but then, if during the time i start mplayer, it will complain about 'no sound'. Turns out, I have to use dmix. So, i modify my .asoundrc like this: pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1025 slave { pcm "hw:0,0" rate 44100 } bindings { 0 0 1 1 } } i put 'pcm "hw:0,0" ' over there, because i want to make sure that this setting works with my actual sound card first. If it works with my card 0, it should work with my card 1. with the above .asoundrc file, i can play multiple sound from different application at the same time. I even tried to remove the "bindings" tag, and it still able to play multiple sound very well. so i guess, this should work on my card 1. then i change the .asoundrc become like this again: pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1025 slave { pcm "hw:1,1" period_time 0 period_size 881 buffer_size 3524 rate 44100 } }the value for perio_time, period_size, and buffer_size have been adjusted by me to suits my hardware requirement. This is the setting that works on my machine. It's just that, in the end, I'm still ended up with one sound at a time. I can't run mplayer + youtube at the same time. I can only run one of them. And furthermore, if i just paused the mplayer, the audio device is still locks. It will be good enough if someone can hint me a way to unlock the audio device. So, what i want to ask is.. can this Loopback card handle more than 1 sound at a time? Have anyone tried this before? I have tried changing the value of .asoundrc, switching the slave pcm to pcm "hw:1,0" (not working) pcm "hw:1" (works, but only 1 sound too). Any idea guys? -santo-

1 0

[alsa-devel] Problem with Pulse Audio plugin?
by Tom Watson 16 Mar '09

16 Mar '09

I'm doing some experimenting and found that a simple sequential play of a small sound file caused an assertion in the pulse audio plugin. The base system is Fedora 8, and it uses alsa 1.0.16 versions. I looked at the 1.0.19 versions and they still have assertions, so the problem may still be there. It is pretty easy to trip up: $ aplay blah.wav blah.wav Playing WAVE 'blah.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Playing WAVE 'blah.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_pulse.c:524: pulse_hw_params: Assertion `!pcm->stream' failed. Aborted by signal Aborted... (where "blah.wav" is a simple short wav file) The first time through, it plays find, but the second time it doesn't. If this has been solved in 1.0.19, (I haven't installed it), please let me know and I'll go away. If not, it may be a problem in the pulse audio thing. Thanks. -- Tom Watson tsw(a)johana.com

2 2

Re: [alsa-devel] [PATCH 1/2] Echoaudio: Add support for Indigo express cards
by Takashi Iwai 16 Mar '09

16 Mar '09

At Mon, 16 Mar 2009 09:14:20 +0100 (CET), Giuliano Pochini wrote: > > > > On Mon, 16 Mar 2009, Takashi Iwai wrote: > > >> This patch adds support for IndigoIOx and IndigoDJx. > >> > >> Signed-off-by Giuliano Pochini <pochini(a)shiny.it> > > > > The code looks almost good, but could you do the following? > > > > - split to two patches, one for kernel tree and one for alsa-driver > > tree > > There is a little problem: since about a month ago the git server is > SLOWWW. It takes ages to download the alsa-kernel repository at 5KB/s :( > I'll try from some server at work which stay on 24/7. Better to try my sound git tree on kernel.org. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git The bare alsa-driver tree is: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-driver-build.git The corresponding alsa-driver snapshot is: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… Takashi

1 0

[alsa-devel] [PATCH 1/1] echomixer: Fix echomixer to work with the new drivers
by Giuliano Pochini 16 Mar '09

16 Mar '09

There is a long standing bug in the drivers for cards with a vmixer because I overlooked a detail in the c++ generic driver by echoaudio. Those cards do not have a line-out volume control. It is a virtual control provided by the generic driver. The bug is harmless because the DSP just ignores the command to change the volume. Since that control has been removed, echomixer must be updated. With this patch it uses the vmixer to fake the line-out volume. This patch makes echomixer work with the new drivers. Signed-off-by Giuliano Pochini <pochini(a)shiny.it> --- alsa-tools-1.0.19/echomixer/echomixer.c__orig2 2009-03-15 19:03:24.000000000 +0100 +++ alsa-tools-1.0.19/echomixer/echomixer.c 2009-03-15 19:36:23.000000000 +0100 @@ -1178,6 +1178,29 @@ void UpdateMixerVolume(int outchannel) { +// Changes the vmixer volume according to the current Line-out volume for vmixer cards. +void UpdateVMixerVolume(int outchannel) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_value_t *control; + int err, val, ch; + + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_value_alloca(&control); + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); + + for (ch=0; ch<nPOut; ch++) { + val=Add_dB(vmixerControl.mixer[outchannel][ch].Gain, lineoutControl.Gain[outchannel]); + ClampOutputVolume(&val); + snd_ctl_elem_id_set_numid(id, vmixerControl.mixer[outchannel][ch].id); + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_value_set_integer(control, 0, val); + if ((err = snd_ctl_elem_write(ctlhandle, control)) < 0) + printf("Control %s element write error: %s\n", card, snd_strerror(err)); + } +} + + + void LineOut_volume_changed(GtkWidget *widget, gpointer ch) { char str[16]; int err, channel, val; @@ -1204,6 +1227,9 @@ void LineOut_volume_changed(GtkWidget *w snd_ctl_elem_value_set_integer(control, channel, val); if ((err=snd_ctl_elem_write(ctlhandle, control))<0) printf("Control %s element write error: %s\n", card, snd_strerror(err)); + } else if (vmixerId) { + UpdateVMixerVolume(channel); + UpdateMixerVolume(channel); } else { // Otherwise we have to emulate it. UpdatePCMVolume(channel); UpdateMixerVolume(channel); @@ -1217,10 +1243,11 @@ void LineOut_volume_changed(GtkWidget *w void Vmixer_volume_changed(GtkWidget *widget, gpointer ch) { char str[16]; - int val, channel; + int val, rval, channel; int o, v; channel=(int)(long)ch; + val=INVERT((int)GTK_ADJUSTMENT(widget)->value); #ifdef REVERSE v=channel; @@ -1230,13 +1257,17 @@ void Vmixer_volume_changed(GtkWidget *wi o=channel; #endif - val=INVERT((int)GTK_ADJUSTMENT(widget)->value); + // Emulate the line-out volume if this card can't do it in hw. + if (!lineoutId) { + rval=Add_dB(val, lineoutControl.Gain[o]); + ClampOutputVolume(&rval); + } - SetMixerGain(&vmixerControl.mixer[o][v], val); + SetMixerGain(&vmixerControl.mixer[o][v], rval); vmixerControl.mixer[o][v].Gain=val; if (Gang) { - SetMixerGain(&vmixerControl.mixer[o^1][v^1], val); + SetMixerGain(&vmixerControl.mixer[o^1][v^1], rval); vmixerControl.mixer[o^1][v^1].Gain=val; } -- Giuliano.

2 1

[alsa-devel] [PATCH 1/2] Echoaudio: remove line-out volume from vmixer cards
by Giuliano Pochini 16 Mar '09

16 Mar '09

There is a long standing bug in the drivers for cards with a vmixer because I overlooked a detail in the c++ generic driver by echoaudio. Those cards do not have a line-out volume control. It is a virtual control provided by the generic driver. The bug is harmless because the DSP just ignores the command to change the volume. *NB:* It breaks alsa-tools/echomixer. A patch for it will follow. This patch removes the line-out volume control from vmixer-equipped cards. Signed-off-by Giuliano Pochini <pochini(a)shiny.it> diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.c 2009-03-15 16:51:17.000000000 +0100 @@ -950,6 +950,8 @@ static int __devinit snd_echo_new_pcm(st Control interface ******************************************************************************/ +#ifndef ECHOCARD_HAS_VMIXER + /******************* PCM output volume *******************/ static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1001,18 +1003,6 @@ static int snd_echo_output_gain_put(stru return changed; } -#ifdef ECHOCARD_HAS_VMIXER -/* On Vmixer cards this one controls the line-out volume */ -static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = { - .name = "Line Playback Volume", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = snd_echo_output_gain_info, - .get = snd_echo_output_gain_get, - .put = snd_echo_output_gain_put, - .tlv = {.p = db_scale_output_gain}, -}; -#else static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1022,6 +1012,7 @@ static struct snd_kcontrol_new snd_echo_ .put = snd_echo_output_gain_put, .tlv = {.p = db_scale_output_gain}, }; + #endif @@ -2037,8 +2028,6 @@ static int __devinit snd_echo_probe(stru #ifdef ECHOCARD_HAS_VMIXER snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip); - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0) - goto ctl_error; if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0) goto ctl_error; #else -- Giuliano.

2 1

[alsa-devel] [PATCH 1/2] Echoaudio: Add support for Indigo express cards
by Giuliano Pochini 16 Mar '09

16 Mar '09

This patch adds support for IndigoIOx and IndigoDJx. I tested IndigoIOx myself, and Carmen Rocco tested the IndigoDJx. It wasn't tested extensively, but I think it stable enough. Maybe you want to mark it as experimental anyway. The soundcard matrix should be updated as well, but the page is locked. These two cards have the same features of their respective CARDBUS versions. The links to the manufacturer's site are: http://www.echoaudio.com/Products/ExpressCard/IndigoDJx/ and http://www.echoaudio.com/Products/ExpressCard/IndigoIOx/ . I want to thank Echoaudio again for their support. This patch adds support for IndigoIOx and IndigoDJx. Signed-off-by Giuliano Pochini <pochini(a)shiny.it> diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.h alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio_dsp.h --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.h 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio_dsp.h 2009-03-15 17:38:47.000000000 +0100 @@ -576,8 +576,13 @@ SET_LAYLA24_FREQUENCY_REG command. #define E3G_ASIC_NOT_LOADED 0xffff #define E3G_BOX_TYPE_MASK 0xf0 -#define EXT_3GBOX_NC 0x01 -#define EXT_3GBOX_NOT_SET 0x02 +/* Indigo express control register values */ +#define INDIGO_EXPRESS_32000 0x02 +#define INDIGO_EXPRESS_44100 0x01 +#define INDIGO_EXPRESS_48000 0x00 +#define INDIGO_EXPRESS_DOUBLE_SPEED 0x10 +#define INDIGO_EXPRESS_QUAD_SPEED 0x04 +#define INDIGO_EXPRESS_CLOCK_MASK 0x17 /* diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/echoaudio.h 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/echoaudio.h 2009-03-15 17:14:39.000000000 +0100 @@ -189,6 +189,9 @@ #define INDIGO 0x0090 #define INDIGO_IO 0x00a0 #define INDIGO_DJ 0x00b0 +#define DC8 0x00c0 +#define INDIGO_IOX 0x00d0 +#define INDIGO_DJX 0x00e0 #define ECHO3G 0x0100 diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx.c alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx.c 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,107 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_DJX +#define ECHOCARD_NAME "Indigo DJx" +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 0 */ +#define PX_DIGITAL_IN 8 /* 0 */ +#define PX_NUM 8 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 4 */ +#define BX_DIGITAL_OUT 4 /* 0 */ +#define BX_ANALOG_IN 4 /* 0 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/io.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_djx_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_DJX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_djx_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 4, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigodjx_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigodjx_dsp.c 2009-03-15 17:17:50.000000000 +0100 @@ -0,0 +1,65 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini(a)shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo DJx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) + return -ENODEV; + + if ((err = init_dsp_comm_page(chip))) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + if ((err = load_firmware(chip)) < 0) + return err; + chip->bad_board = FALSE; + + if ((err = init_line_levels(chip)) < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_express_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigo_express_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_express_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigo_express_dsp.c 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,119 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini(a)shiny.it> + +*************************************************************************/ + +static int set_sample_rate(struct echoaudio *chip, u32 rate) +{ + u32 clock, control_reg, old_control_reg; + + if (wait_handshake(chip)) + return -EIO; + + old_control_reg = le32_to_cpu(chip->comm_page->control_register); + control_reg = old_control_reg & ~INDIGO_EXPRESS_CLOCK_MASK; + + switch (rate) { + case 32000: + clock = INDIGO_EXPRESS_32000; + break; + case 44100: + clock = INDIGO_EXPRESS_44100; + break; + case 48000: + clock = INDIGO_EXPRESS_48000; + break; + case 64000: + clock = INDIGO_EXPRESS_32000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 88200: + clock = INDIGO_EXPRESS_44100|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 96000: + clock = INDIGO_EXPRESS_48000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + default: + return -EINVAL; + } + + control_reg |= clock; + if (control_reg != old_control_reg) { + chip->comm_page->control_register = cpu_to_le32(control_reg); + chip->sample_rate = rate; + clear_handshake(chip); + return send_vector(chip, DSP_VC_UPDATE_CLOCKS); + } + return 0; +} + + + +/* This function routes the sound from a virtual channel to a real output */ +static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, + int gain) +{ + int index; + + if (snd_BUG_ON(pipe >= num_pipes_out(chip) || + output >= num_busses_out(chip))) + return -EINVAL; + + if (wait_handshake(chip)) + return -EIO; + + chip->vmixer_gain[output][pipe] = gain; + index = output * num_pipes_out(chip) + pipe; + chip->comm_page->vmixer[index] = gain; + + DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + return 0; +} + + + +/* Tell the DSP to read and update virtual mixer levels in comm page. */ +static int update_vmixer_level(struct echoaudio *chip) +{ + if (wait_handshake(chip)) + return -EIO; + clear_handshake(chip); + return send_vector(chip, DSP_VC_SET_VMIXER_GAIN); +} + + + +static u32 detect_input_clocks(const struct echoaudio *chip) +{ + return ECHO_CLOCK_BIT_INTERNAL; +} + + + +/* The IndigoIO has no ASIC. Just do nothing */ +static int load_asic(struct echoaudio *chip) +{ + return 0; +} diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox.c alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox.c 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,109 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_IOX +#define ECHOCARD_NAME "Indigo IOx" +#define ECHOCARD_HAS_MONITOR +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 2 */ +#define PX_DIGITAL_IN 10 /* 0 */ +#define PX_NUM 10 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 2 */ +#define BX_DIGITAL_OUT 2 /* 0 */ +#define BX_ANALOG_IN 2 /* 2 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/io.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_iox_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_IOX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_iox_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 8, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigoiox_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" + diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigoiox_dsp.c 2009-03-15 17:17:38.000000000 +0100 @@ -0,0 +1,65 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini(a)shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo IOx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) + return -ENODEV; + + if ((err = init_dsp_comm_page(chip))) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + if ((err = load_firmware(chip)) < 0) + return err; + chip->bad_board = FALSE; + + if ((err = init_line_levels(chip)) < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/Makefile alsa-driver-1.0.19/sound/pci/echoaudio/Makefile --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/Makefile 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/Makefile 2009-03-15 17:14:39.000000000 +0100 @@ -15,6 +15,8 @@ snd-echo3g-objs := echo3g.o snd-indigo-objs := indigo.o snd-indigoio-objs := indigoio.o snd-indigodj-objs := indigodj.o +snd-indigoiox-objs := indigoiox.o +snd-indigodjx-objs := indigodjx.o obj-$(CONFIG_SND_DARLA20) += snd-darla20.o obj-$(CONFIG_SND_GINA20) += snd-gina20.o @@ -28,3 +30,5 @@ obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o obj-$(CONFIG_SND_INDIGO) += snd-indigo.o obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o +obj-$(CONFIG_SND_INDIGOIOX) += snd-indigoiox.o +obj-$(CONFIG_SND_INDIGODJX) += snd-indigodjx.o --- alsa-driver-1.0.19__orig/alsa-kernel/pci/Kconfig 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/Kconfig 2009-03-15 17:14:39.000000000 +0100 @@ -400,6 +400,26 @@ config SND_INDIGODJ To compile this driver as a module, choose M here: the module will be called snd-indigodj +config SND_INDIGOIOX + tristate "(Echoaudio) Indigo IOx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo IOx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigoiox + +config SND_INDIGODJX + tristate "(Echoaudio) Indigo DJx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo DJx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigodjx + config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" select FW_LOADER --- alsa-driver-1.0.19__orig/pci/echoaudio/Makefile 2009-01-19 12:17:17.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/Makefile 2009-03-15 17:14:39.000000000 +0100 @@ -11,7 +11,8 @@ include $(SND_TOPDIR)/toplevel.config include $(SND_TOPDIR)/Makefile.conf clean-files := echoaudio.c darla20.c darla24.c echo3g.c gina20.c gina24.c \ - indigo.c indigodj.c indigoio.c layla20.c layla24.c mia.c mona.c + indigo.c indigodj.c indigoio.c indigodjx.c indigoiox.c \ + layla20.c layla24.c mia.c mona.c include $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/Makefile @@ -26,6 +27,8 @@ gina24.c: gina24.patch $(SND_TOPDIR)/als indigo.c: indigo.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigo.c echoaudio.c indigodj.c: indigodj.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigodj.c echoaudio.c indigoio.c: indigoio.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigoio.c echoaudio.c +indigodjx.c: indigodjx.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigodjx.c echoaudio.c +indigoiox.c: indigoiox.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/indigoiox.c echoaudio.c layla20.c: layla20.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/layla20.c echoaudio.c layla24.c: layla24.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/layla24.c echoaudio.c mia.c: mia.patch $(SND_TOPDIR)/alsa-kernel/pci/echoaudio/mia.c echoaudio.c --- alsa-driver-1.0.19__orig/pci/echoaudio/indigoiox.patch 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigoiox.patch 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,18 @@ +--- ../../alsa-kernel/pci/echoaudio/indigoiox.c 2009-03-01 15:26:34.000000000 +0100 ++++ indigoiox.c 2009-03-01 15:32:56.000000000 +0100 +@@ -1,3 +1,9 @@ ++#include "adriver.h" ++ ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0) ++#define pci_device(chip) pci_name(chip->pci) ++#endif ++ + /* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> +@@ -107,3 +113,5 @@ static struct snd_pcm_hardware pcm_hardw + #include "echoaudio_dsp.c" + #include "echoaudio.c" + ++ ++EXPORT_NO_SYMBOLS; --- alsa-driver-1.0.19__orig/pci/echoaudio/indigodjx.patch 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigodjx.patch 2009-03-15 17:14:39.000000000 +0100 @@ -0,0 +1,18 @@ +--- ../../alsa-kernel/pci/echoaudio/indigodjx.c 2009-03-01 15:26:27.000000000 +0100 ++++ indigodjx.c 2009-03-01 19:02:55.000000000 +0100 +@@ -1,3 +1,9 @@ ++#include "adriver.h" ++ ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0) ++#define pci_device(chip) pci_name(chip->pci) ++#endif ++ + /* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini(a)shiny.it> +@@ -105,3 +111,5 @@ static struct snd_pcm_hardware pcm_hardw + #include "indigo_express_dsp.c" + #include "echoaudio_dsp.c" + #include "echoaudio.c" ++ ++EXPORT_NO_SYMBOLS; --- alsa-driver-1.0.19__orig/pci/echoaudio/indigo_express_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigo_express_dsp.c 2009-03-15 17:21:56.000000000 +0100 @@ -0,0 +1 @@ +#include "../../alsa-kernel/pci/echoaudio/indigo_express_dsp.c" --- alsa-driver-1.0.19__orig/pci/echoaudio/indigoiox_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigoiox_dsp.c 2009-03-15 17:19:36.000000000 +0100 @@ -0,0 +1 @@ +#include "../../alsa-kernel/pci/echoaudio/indigoiox_dsp.c" --- alsa-driver-1.0.19__orig/pci/echoaudio/indigodjx_dsp.c 1970-01-01 01:00:00.000000000 +0100 +++ alsa-driver-1.0.19/pci/echoaudio/indigodjx_dsp.c 2009-03-15 17:19:18.000000000 +0100 @@ -0,0 +1 @@ +#include "../../alsa-kernel/pci/echoaudio/indigodjx_dsp.c" -- Giuliano.

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[alsa-devel] When I got sound back by "kernel: hda-intel: Invalid position buffer, using LPIB read method instead."(random fallover)
by Emmanuel Chanel 16 Mar '09

16 Mar '09

When I got sound back by "kernel: hda-intel: Invalid position buffer, using LPIB read method instead."(random fallover) It doesn't mean succeed. Sometimes sound get back and sometimes not, at random.

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[alsa-devel] no sound from headphones with STAC9205
by Thomas Folz-Donahue 16 Mar '09

16 Mar '09

I'm using a Gateway T-1616, which is an RS690 laptop from last year. It uses a STAC9205 codec. Somewhere between 1.0.16 and i suppose as early as 1.0.17, there was a regression wherein when I plug in my headphones, they are correctly detected and activated and the speakers are deactivated but then the headphones turn off. None of the existing model flags work. alsa-info.sh reports this: http://www.alsa-project.org/db/?f=d22d464730ca3ebd485ac0911eebb9eb60986233 and also, I ran alsa-info.sh with everything working: http://www.alsa-project.org/db/?f=b9c86006d49c0ff9ce9df02154db032ae2217786 ~thomas -- cartwheel ^(^_^)> v(.-.)v <(^_^)^

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[alsa-devel] [PATCH 0/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

Hi, I found twl4030 build error. It appears because of changes of following commit: ASoC: make ops a pointer in 'struct snd_soc_dai' --- Joonyoung Shim (1): ASoC: twl4030 - Fix build error sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) -- - Joonyoung Shim

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[alsa-devel] [PATCH 0/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

Hi -- - Joonyoung Shim

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[alsa-devel] broonie/sound-2.6.git rebase?
by Daniel Mack 15 Mar '09

15 Mar '09

Hi Mark, Could you rebase your broonie/sound-2.6.git to Linus' master? When trying to merge them, I currently get: _From git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 * branch pxa-ssp -> FETCH_HEAD Auto-merging arch/arm/mach-shark/include/mach/io.h CONFLICT (content): Merge conflict in arch/arm/mach-shark/include/mach/io.h Auto-merging sound/soc/codecs/cs4270.c Auto-merging sound/soc/codecs/tlv320aic26.c Auto-merging sound/soc/codecs/wm8400.c CONFLICT (add/add): Merge conflict in sound/soc/codecs/wm8400.c Auto-merging sound/soc/codecs/wm8580.c CONFLICT (content): Merge conflict in sound/soc/codecs/wm8580.c Auto-merging sound/soc/pxa/pxa-ssp.c Auto-merging sound/soc/pxa/pxa2xx-ac97.c Auto-merging sound/soc/s3c24xx/s3c24xx-i2s.c Automatic merge failed; fix conflicts and then commit the result. Or is there any mystery about git I still haven't fully understood? Thanks, Daniel

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[alsa-devel] [PATCH 2/2] Echoaudio: Add support for Indigo express cards
by Giuliano Pochini 15 Mar '09

15 Mar '09

This patch adds to alsa-firmware the firmware needed by the IndigoIOx and IndigoDJx cards. It's too long (134K) to post here inline. Please download the patch here: Signed-off-by Giuliano Pochini <pochini(a)shiny.it> http://www.webalice.it/g_pochini/alsa-firmware_indigox.patch or http://www.webalice.it/g_pochini/alsa-firmware_indigox.patch.bz2 -- Giuliano.

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[alsa-devel] [PATCH 2/2] Echoaudio: remove line-out volume from vmixer cards
by Giuliano Pochini 15 Mar '09

15 Mar '09

With this patch the drivers do not set the vmixer volume anymore at startup because it is actually the output volume of the voices and ALSA mandates that the volume must be 0 by default. Signed-off-by Giuliano Pochini <pochini(a)shiny.it> diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigodj_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigodj_dsp.c 2009-03-15 17:24:35.000000000 +0100 @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: vchannels 0-3 and - vchannels 4-7 are routed to real channels 0-4 */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 2, 2, 0); - set_vmixer_gain(chip, 3, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigo_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigo_dsp.c 2009-03-15 17:24:14.000000000 +0100 @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/indigoio_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/indigoio_dsp.c 2009-03-15 17:24:00.000000000 +0100 @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff -dupN alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/mia_dsp.c alsa-driver-1.0.19/sound/pci/echoaudio/mia_dsp.c --- alsa-driver-1.0.19__orig/alsa-kernel/pci/echoaudio/mia_dsp.c 2009-01-19 12:08:58.000000000 +0100 +++ alsa-driver-1.0.19/sound/pci/echoaudio/mia_dsp.c 2009-03-15 17:23:37.000000000 +0100 @@ -69,18 +69,6 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip))) return err; - /* Default routing of the virtual channels: vchannels 0-3 go to analog - outputs and vchannels 4-7 go to S/PDIF outputs */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 2, 4, 0); - set_vmixer_gain(chip, 3, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } -- Giuliano.

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[alsa-devel] [RFC/PATCH] pxa2xx_ac97 and gpio reset line
by Robert Jarzmik 15 Mar '09

15 Mar '09

Hi Mark, I can't find out if somebody had submitted that kind of patch before. This is a starting point of discussion. I'm not fully convinced by my own code, and so I post it here for advice. What I'd really like is to be able to choose the PXA AC97 gpio reset (GPIO95 or GPIO113) through platform_data, but I couldn't find a way of doing it. I hope you'll have some kind of idea. Cheers. -- Robert

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