November 2007 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] snd_pcm_hw_params_get_xxx question
by Ben C 07 Nov '07

07 Nov '07

Takashi, You were right, it was a linking issue. I'm doing cross-compiling for a target of different processor type. I just had to make sure I'm pointing to the right location for the includes and libs. Thanks, Ben ----- Original Message ---- From: Takashi Iwai <tiwai(a)suse.de> To: Ben C <bchang_2(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Sent: Wednesday, November 7, 2007 4:24:04 AM Subject: Re: [alsa-devel] snd_pcm_hw_params_get_xxx question At Tue, 6 Nov 2007 18:21:01 -0800 (PST), Ben C wrote: > > Hello, > > I'm trying to develop an external PCM plugin. I'm calling snd_pcm_hw_params_get_xxx functions to retrieve some values. I got everything to work as expected on alsa-lib 1.0.15. However the client's setup is 1.0.13, and I don't think they want to upgrade. > > For 1.0.13, everything compiled and linked fine after some minor changes. But when I called say snd_pcm_hw_params_get_rate(hw, &val, &dir), I was expecting the return value to be a status, and val to contain "44100" which was the case for 1.0.15. However for 1.0.13, val is 0, but returned status = 44100? > > I looked in the header files, and 1.0.13 seems to be the same as 1.0.15 in regards to this function. Am I accidentally using some sort of old API that was intended for backward compatibility? If so, what should I do to use the "new" API's? Yes, it sounds like you're accessing the old API. Do you use libasound from a module? If it's used from a module, the module itself should be built once with link to libasound. For example, $(CC) -shared -o MODULE.so -lasound In such a way, the properly versioned symbol is chosen. The weak link results in the old API. This is irrelveant whether it's ALSA 1.0.13 or 1.0.15. Takashi __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] [HDA Intel/STAC92xx] Alsa 1.0.14
by Greg Jizmagian 07 Nov '07

07 Nov '07

I am trying to fix the sound on the m1330 in OS X. It currently only works out of one of the headphone jacks. I stumbled across the patch you wrote for ubuntu and hoped that I could adapt it for OS X. Or is there a way to see a post patch linux dump that would fix the problem? Any suggestions/help would be greatly appreciated. Thanks. Greg

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[alsa-devel] snd_pcm_hw_params_get_xxx question
by Ben C 07 Nov '07

07 Nov '07

Hello, I'm trying to develop an external PCM plugin. I'm calling snd_pcm_hw_params_get_xxx functions to retrieve some values. I got everything to work as expected on alsa-lib 1.0.15. However the client's setup is 1.0.13, and I don't think they want to upgrade. For 1.0.13, everything compiled and linked fine after some minor changes. But when I called say snd_pcm_hw_params_get_rate(hw, &val, &dir), I was expecting the return value to be a status, and val to contain "44100" which was the case for 1.0.15. However for 1.0.13, val is 0, but returned status = 44100? I looked in the header files, and 1.0.13 seems to be the same as 1.0.15 in regards to this function. Am I accidentally using some sort of old API that was intended for backward compatibility? If so, what should I do to use the "new" API's? Thanks, Ben __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] [PATCH] hda: STAC9228 Subsystem update
by Matthew Ranostay 07 Nov '07

07 Nov '07

Added more laptops subsystem id's that have STAC9228 DMIC support. --- 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 14:01:53 2007 -0500 @@ -2921,8 +2921,10 @@ static int patch_stac927x(struct hda_cod } switch (codec->subsystem_id) { - case 0x1028020A: /* STAC 9228 */ - case 0x10280209: /* STAC 9228 */ + case 0x10280242: /* STAC 9228 */ + case 0x102801f3: + case 0x1028020A: + case 0x10280209: spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; spec->dmux_nid = 0x1b;

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[alsa-devel] [PATCH] cmipci - utilize ADC48K44K bit
by Timofei V. Bondarenko 07 Nov '07

07 Nov '07

Hi. Setting the ADC48K44K greatly improves capture quality at 48k sampling rate. With this bit clear ADC does ZOH interpolation of every 22th sample at 48k. At frequencies higher than 48k there ADC performs a little better with ADC48K44K bit set. At 44.1k ADC performs a little better with this bit clear. At frequencies below 44.1k there is no difference. Regards, Tim.

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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 06 Nov '07

06 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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