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[alsa-devel] [PATCH] hda: STAC92HD71 codec mixer
by Matthew Ranostay 07 Nov '07

07 Nov '07

Added analog loopback support and missing ADC capture mixer for the STAC92HD71 codec family. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 43708ece3f6f pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Nov 06 11:56:17 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Nov 06 21:05:54 2007 -0500 @@ -428,6 +428,11 @@ static struct hda_verb stac92hd71bxx_cor { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* connect ports 0d and 0f to audio mixer */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + /* unmute dac0 input in audio mixer */ + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -512,8 +517,7 @@ static struct hda_verb stac9205_core_ini .put = stac92xx_volknob_put, \ .private_value = 127 | (knob_nid << 16), \ } - - + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -530,12 +534,19 @@ static struct snd_kcontrol_new stac92hd7 STAC_VOLKNOB(0x28), /* hardware gain controls */ - HDA_CODEC_VOLUME("Digital Mic 1 Volume", 0x18, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Digital Mic 2 Volume", 0x19, 0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Capture Volume", 0x1c, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1c, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x1a, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + + HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), { } /* end */ };

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[alsa-devel] Mr. Takashi Iwai Help-me please
by Eduardo 07 Nov '07

07 Nov '07

Mr. Takashi Escuse-me for my question, but i want a developer a driver for MCM590 i have a dump for USB and the manuals to develop audio driver for linux, but nothing for a manufacter documentation. This is a Philips Microsystem with a USB port This USB periferic need a special boot sequence, i debug this sequence in the linux kernel and the default boot for "usbaudio.c" don't solve the problem. Do you can help-me? Eduardo Please answer this mail with no or yes tks!

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[alsa-devel] [PATCH] soc/sh: help GCC improve generated code
by Manuel Lauss 06 Nov '07

06 Nov '07

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[alsa-devel] [PATCH] hda: Added new IDT codec family
by Matthew Ranostay 06 Nov '07

06 Nov '07

Added initial support for the STAC92HD71BXX family of codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 3e1a17f8baca pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Wed Oct 31 17:36:20 2007 +0100 +++ b/pci/hda/hda_codec.h Sat Nov 03 22:55:14 2007 -0500 @@ -92,6 +92,8 @@ enum { */ #define AC_VERB_SET_STREAM_FORMAT 0x200 #define AC_VERB_SET_AMP_GAIN_MUTE 0x300 +#define AC_VERB_SET_RIGHT_AMP_IN_MUTE 0x350 +#define AC_VERB_SET_LEFT_AMP_IN_MUTE 0x360 #define AC_VERB_SET_PROC_COEF 0x400 #define AC_VERB_SET_COEF_INDEX 0x500 #define AC_VERB_SET_CONNECT_SEL 0x701 diff -r 3e1a17f8baca pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Oct 31 17:36:20 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Nov 05 09:34:05 2007 -0500 @@ -62,6 +62,11 @@ enum { }; enum { + STAC_92HD71BXX_REF, + STAC_92HD71BXX_MODELS +}; + +enum { STAC_925x_REF, STAC_M2_2, STAC_MA6, @@ -172,6 +177,23 @@ static hda_nid_t stac9200_dac_nids[1] = 0x02, }; +static hda_nid_t stac92hd71bxx_adc_nids[2] = { + 0x12, 0x13, +}; + +static hda_nid_t stac92hd71bxx_mux_nids[2] = { + 0x1a, 0x1b +}; + +static hda_nid_t stac92hd71bxx_dac_nids[2] = { + 0x10, /*0x11, */ +}; + +#define STAC92HD71BXX_NUM_DMICS 2 +static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { + 0x18, 0x19, 0 +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -236,6 +258,11 @@ static hda_nid_t stac922x_pin_nids[10] = static hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, +}; + +static hda_nid_t stac92hd71bxx_pin_nids[10] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x14, 0x18, 0x19, 0x1e, }; static hda_nid_t stac927x_pin_nids[14] = { @@ -393,6 +420,25 @@ static struct hda_verb stac9200_eapd_ini /* set dac0mux for dac converter */ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {} +}; + +static struct hda_verb stac92hd71bxx_core_init[] = { + /* set master volume and direct control */ + { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* unmute right and left channels for node 0x0a */ + { 0x0a, AC_VERB_SET_LEFT_AMP_IN_MUTE, 0x0}, + { 0x0a, AC_VERB_SET_RIGHT_AMP_IN_MUTE, 0x0}, + /* connect headphone jack to dac1 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* unmute right and left channels for node 0x0d */ + { 0x0d, AC_VERB_SET_LEFT_AMP_IN_MUTE, 0x0}, + { 0x0d, AC_VERB_SET_RIGHT_AMP_IN_MUTE, 0x0}, + /* unmute right and left channels for node 0x0f */ + { 0x0f, AC_VERB_SET_LEFT_AMP_IN_MUTE, 0x0}, + { 0x0f, AC_VERB_SET_RIGHT_AMP_IN_MUTE, 0x0}, + /* unmute mono out node */ + { 0x14, AC_VERB_SET_LEFT_AMP_IN_MUTE, 0x0}, {} }; @@ -480,6 +526,21 @@ static struct snd_kcontrol_new stac9200_ HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), + STAC_INPUT_SOURCE(2), + STAC_VOLKNOB(0x28), + + /* hardware gain controls */ + HDA_CODEC_VOLUME("Digital Mic 1 Volume", 0x18, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Digital Mic 2 Volume", 0x19, 0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Capture Volume", 0x1c, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1c, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Mux Volume", 0x1a, 0, HDA_OUTPUT), { } /* end */ }; @@ -835,6 +896,27 @@ static struct snd_pci_quirk stac925x_cfg SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_MA6), SND_PCI_QUIRK(0x107b, 0x0681, "Gateway NX860", STAC_PA6), SND_PCI_QUIRK(0x1002, 0x437b, "Gateway MX6453", STAC_M2_2), + {} /* terminator */ +}; + +static unsigned int ref92hd71bxx_pin_configs[10] = { + 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, + 0x0181302e, 0x01114010, 0x01a19020, 0x90a000f0, + 0x90a000f0, 0x01452050, +}; + +static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, +}; + +static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_REF] = "ref", +}; + +static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { + /* SigmaTel reference board */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, + "DFI LanParty", STAC_92HD71BXX_REF), {} /* terminator */ }; @@ -2614,6 +2696,77 @@ static int patch_stac925x(struct hda_cod return 0; } + +static int patch_stac92hd71bxx(struct hda_codec *codec) { + struct sigmatel_spec *spec; + int err = 0; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->pin_nids = stac92hd71bxx_pin_nids; + + spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, + stac92hd71bxx_models, + stac92hd71bxx_cfg_tbl); +again: + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC92HD71BXX, " + "using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } + else { + spec->pin_configs = stac92hd71bxx_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + stac92xx_enable_gpio_mask(codec); + + spec->init = stac92hd71bxx_core_init; + spec->mixer = stac92hd71bxx_mixer; + + spec->mux_nids = stac92hd71bxx_mux_nids; + spec->adc_nids = stac92hd71bxx_adc_nids; + spec->dmic_nids = stac92hd71bxx_dmic_nids; + spec->dmux_nid = 0x1c; + + spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); + spec->num_dmics = STAC92HD71BXX_NUM_DMICS; + + spec->multiout.num_dacs = 2; + spec->multiout.hp_nid = 0x11; + spec->multiout.dac_nids = stac92hd71bxx_dac_nids; + + err = stac92xx_parse_auto_config(codec, 0x21, 0x23); + if (!err) { + if (spec->board_config < 0) { + printk(KERN_WARNING "hda_codec: No auto-config is " + "available, default to model=ref\n"); + spec->board_config = STAC_92HD71BXX_REF; + goto again; + } + err = -EINVAL; + } + + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +}; static int patch_stac922x(struct hda_codec *codec) { @@ -3176,5 +3329,7 @@ struct hda_codec_preset snd_hda_preset_s { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d76b0, .name = "STAC92HD71BXX", .patch = patch_stac92hd71bxx }, {} /* terminator */ }; +

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[alsa-devel] Wyse Winterm CS5535 Driver
by Darryl Bond 05 Nov '07

05 Nov '07

I have a Wyse S50 terminal that I am booting diskless. I have been trying to get sound working. The box has a cs5535 chipset. After much fiddling in the kernel source (2.6.22.5 <http://2.6.22.5> ) I have been able to make it work. I'm not sure if it is a cs5535 problem as the code that fixes it is in ac97_pcm.c. It would appear that the card is getting passed an unexpected rate. The function sound/pci/ac97/ac97_pcm.c snd_ac97_set_rate() has a section: case AC97_PCM_FRONT_DAC_RATE: case AC97_PCM_LR_ADC_RATE: if ((ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_VRA) == 0) /* VRA */ if (rate != 48000 && rate != 96000) return -EINVAL; break; If I use 'aplay wavfile' the rate passed to this function is 44100. I I changethe code to: if (rate != 48000 && rate != 96000 && rate != 44100) it plays but at a slightly incorrect rate. If I use the Citrix ICA client (which is what I really want to work) the rate passed in is 8000. If I add 8000 to the test, it works, but is far too fast. Any hints as to what I should try? Regards Darryl Bond __________________________________________________________________________________ DISCLAIMER The contents of this electronic message and any attachments are intended only for the addressee and may contain legally privileged, personal, sensitive or confidential information. If you are not the intended addressee, and have received this email, any transmission, distribution, downloading, printing or photocopying of the contents of this message or attachments is strictly prohibited. Any legal privilege or confidentiality attached to this message and attachments is not waived, lost or destroyed by reason of delivery to any person other than intended addressee. If you have received this message and are not the intended addressee you should notify the sender by return email and destroy all copies of the message and any attachments. Unless expressly attributed, the views expressed in this email do not necessarily represent the views of the company.

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[PATCH] ASoC: sh: improve generated code for HAC module (AC97)
by Manuel Lauss 05 Nov '07

05 Nov '07

Change loops in ac97_read/write functions to count down to zero rather than up. Gcc will then use the 'dt' (decrement-and-test) op instead of an increment/compare op-pair. Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- sound/soc/sh/hac.c | 13 +++++-------- 1 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 8e3f039..34b77b9 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -105,7 +105,7 @@ static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, unsigned int to1, to2, i; unsigned short adr; - for (i = 0; i < AC97_READ_RETRY; ++i) { + for (i = AC97_READ_RETRY; i; i--) { *v = 0; /* wait for HAC to receive something from the codec */ for (to1 = TMO_E4; @@ -132,7 +132,7 @@ static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, udelay(21); } HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); - return (i < AC97_READ_RETRY); + return i; } static unsigned short hac_read_codec_aux(struct hac_priv *hac, @@ -141,7 +141,7 @@ static unsigned short hac_read_codec_aux(struct hac_priv *hac, unsigned short val; unsigned int i, to; - for (i = 0; i < AC97_READ_RETRY; i++) { + for (i = AC97_READ_RETRY; i; i--) { /* send_read_request */ local_irq_disable(); HACREG(HACTSR) &= ~(TSR_CMDAMT); @@ -159,10 +159,7 @@ static unsigned short hac_read_codec_aux(struct hac_priv *hac, break; } - if (i == AC97_READ_RETRY) - return ~0; - - return val; + return i ? val : ~0; } static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, @@ -172,7 +169,7 @@ static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct hac_priv *hac = &hac_cpu_data[unit_id]; unsigned int i, to; /* write_codec_aux */ - for (i = 0; i < AC97_WRITE_RETRY; i++) { + for (i = AC97_WRITE_RETRY; i; i--) { /* send_write_request */ local_irq_disable(); HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); -- 1.5.3.4

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[alsa-devel] [PATCH] [ML403-AC97CR] Fix capture/periodic overrun bug.
by Joachim Foerster 05 Nov '07

05 Nov '07

Hi all, this is a bug fix for my ALSA driver I posted in August. Since then I hadn't the time to look at it, but some days ago I took some time and found the reason why there are so many overruns while capturing. Furthermore, this patch removes the one "heavy" printk for capture debugging. From: Joachim Foerster <JOFT(a)gmx.de> We have to do fairly accurate counting of the minimal periods, instead of being lazy and just setting the number to zero as soon as one period elapses. Signed-off-by: Joachim Foerster <JOFT(a)gmx.de> --- sound/drivers/ml403-ac97cr.c | 6 ++---- sound/drivers/pcm-indirect2.c | 20 ++------------------ 2 files changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 2636249..1b16a21 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -28,11 +28,9 @@ * accesses to a minimum, because after a variable amount of accesses, the AC97 * controller doesn't raise the register access finished bit anymore ... * - * - Capture support works - basically, but after ~30s (with rates > ~20kHz) - * ALSA stops reading captured samples from the intermediate buffer and - * therefore a overrun happens - ATM I don't know what's wrong. - * * - Playback support seems to be pretty stable - no issues here. + * - Capture support "works" now, too. Overruns don't happen any longer so often. + * But there might still be some ... */ #include <sound/driver.h> diff --git a/sound/drivers/pcm-indirect2.c b/sound/drivers/pcm-indirect2.c index a36215a..ef74a38 100644 --- a/sound/drivers/pcm-indirect2.c +++ b/sound/drivers/pcm-indirect2.c @@ -403,7 +403,7 @@ snd_pcm_indirect2_playback_interrupt(struct snd_pcm_substream *substream, rec->min_multiple); rec->mul_elapsed++; #endif - rec->min_periods = 0; + rec->min_periods = (rec->min_periods % rec->min_multiple); snd_pcm_period_elapsed(substream); } } @@ -568,24 +568,8 @@ snd_pcm_indirect2_capture_interrupt(struct snd_pcm_substream *substream, rec->mul_elapsed_real += (rec->min_periods / rec->min_multiple); rec->mul_elapsed++; - - if (!(rec->mul_elapsed % 4)) { - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int appl_ptr = - frames_to_bytes(runtime, - (unsigned int)runtime->control-> - appl_ptr) % rec->sw_buffer_size; - int diff = rec->sw_data - appl_ptr; - if (diff < 0) - diff += rec->sw_buffer_size; - snd_printk(KERN_DEBUG - "STAT: mul_elapsed: %d, sw_data: %u, " - "appl_ptr (bytes): %u, diff: %d\n", - rec->mul_elapsed, rec->sw_data, appl_ptr, - diff); - } #endif - rec->min_periods = 0; + rec->min_periods = (rec->min_periods % rec->min_multiple); snd_pcm_period_elapsed(substream); } } -- 1.5.2.5

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[alsa-devel] [PATCH] hda: STAC9228 DMIC
by Matthew Ranostay 05 Nov '07

05 Nov '07

Added support for the dmics and enabled EAPD for several laptops with STAC9228 cards. --- Signed-off-by: Matthew Ranostay <mranostay(a)gmail.com> diff -r 808153682813 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Oct 26 15:10:15 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Thu Nov 01 15:46:17 2007 -0500 @@ -2778,14 +2778,21 @@ static int patch_stac927x(struct hda_cod spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; spec->dmux_nid = 0x1b; + + /* Enable DMIC0 */ + stac92xx_set_config_reg(codec, 0x13, 0x90a60040); + + /* GPIO2 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000004; break; default: - spec->num_dmics = 0; + spec->num_dmics = 0; + + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; } spec->multiout.dac_nids = spec->dac_nids; - /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; stac92xx_enable_gpio_mask(codec); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);

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[alsa-devel] HDA: Add Asus VX1 support.
by Tobin Davis 05 Nov '07

05 Nov '07

Simple patch to add the Asus VX1 laptop to the Analog Devices pci quirk list. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Upgrading alsa drivers. current and hg from suse site
by Nigel Henry 02 Nov '07

02 Nov '07

I try to help folks now and then with sound problems. I don't have an hda intel card, but try to point folks in the right direction to get help. Question: I've sorted out the upgrading of the alsa driver on Fedora, Archlinux, and Debian, even though I don't need too, as my audigy2 soundblaster, and on the other machine, an ensoniq (ens1371) are fine. this was from the latest 1.0.15 release at http://alsa-project.org. My question is to do with the nightly hg releases obtainable from. ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver/ Will these install using the same ./configure, make, (su to root), then make install, as with upgrading to the current 1.0.15 version from. http://alsa-project.org. Sorry if this is a trivial question, but I don't want to give bad information to folks looking for help. I've only recently subscribed to the alsa-devel list, and hadn't realised how much effort was being put in to resolve soundcard problems, particularly hda intel ones. Great work folks. I'm not clued up enough to help out, but do appreciate your hard work. Nigel.

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