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Re: [alsa-devel] sound-problems with ALC262
by Takashi Iwai 15 Aug '08

15 Aug '08

At Thu, 14 Aug 2008 07:09:41 +0200, Tobias Kaefer wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > > Takashi Iwai schrieb: > | At Wed, 13 Aug 2008 08:07:37 +0200, > | Tobias Kaefer wrote: > |> ... > |> "stan" told me to use this script > |> http://www.alsa-project.org/alsa-info.sh to create more information > |> about my system. Here is the pastebin link of the output of that script: > |> http://pastebin.ca/1169153 > | > | At a quick glance, I see no problem in the configuration. > | > | Does the problem exist even if you play via PCM "hw" device, such as > | > | % aplay -Dhw foo.wav > | > | ?? > | If your WAV file isn't with the supported format/rate, use plughw: > | > | % aplay -Dplughw foo.wav > > Yes. The problem still exists when using the PCM "hw" or "plughw" device. What happens when you load the driver with bdl_pos_adj=0 option? Takashi

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Re: [alsa-devel] [PATCH] hda: dynamically create capture mux controls
by Matthew Ranostay 15 Aug '08

15 Aug '08

Dynamically create capture mux volume controls when a output amp is detected. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ff56432..eae6df1 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -794,7 +794,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { STAC_INPUT_SOURCE(1), 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 */ }; @@ -908,12 +907,9 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { 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), - /* analog pc-beep replaced with digital beep support */ /* HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), @@ -931,11 +927,9 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { 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), { } /* end */ }; @@ -943,7 +937,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), { } /* end */ }; @@ -953,12 +946,9 @@ static struct snd_kcontrol_new stac9205_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x1A, 0x0, HDA_OUTPUT), - { } /* end */ }; @@ -967,11 +957,9 @@ static struct snd_kcontrol_new stac922x_mixer[] = { STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x13, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -982,15 +970,12 @@ static struct snd_kcontrol_new stac927x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x2, 0x17, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -2351,7 +2336,8 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { }; /* add dynamic controls */ -static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char *name, unsigned long val) +static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type, + int idx, const char *name, unsigned long val) { struct snd_kcontrol_new *knew; @@ -2371,6 +2357,7 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char knew = &spec->kctl_alloc[spec->num_kctl_used]; *knew = stac92xx_control_templates[type]; + knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); if (! knew->name) return -ENOMEM; @@ -2379,6 +2366,14 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char return 0; } + +/* add dynamic controls */ +static int stac92xx_add_control(struct sigmatel_spec *spec, int type, + const char *name, unsigned long val) +{ + return stac92xx_add_control_idx(spec, type, 0, name, val); +} + /* flag inputs as additional dynamic lineouts */ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg) { @@ -2780,6 +2775,26 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, return 0; } +static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int wcaps, nid, i, err = 0; + + for (i = 0; i < spec->num_muxes; i++) { + nid = spec->mux_nids[i]; + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + err = stac92xx_add_control_idx(spec, + STAC_CTL_WIDGET_VOL, i, "Mux Capture Volume", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +}; + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -3078,6 +3093,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; + if (spec->num_muxes > 0) { + err = stac92xx_auto_create_mux_input_ctls(codec); + if (err < 0) + return err; + } spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2)

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[alsa-devel] [PATCH] hda: 92HD75xx fixes
by Matthew Ranostay 15 Aug '08

15 Aug '08

Fixed several noise issues with DACs and ADCs on some 92HD75xxx based codecs with certain revision id's. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ff56432..9ec0610 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -145,6 +145,9 @@ struct sigmatel_spec { unsigned int gpio_data; unsigned int gpio_mute; + /* stream */ + unsigned int stream_delay; + /* analog loopback */ unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -190,6 +193,7 @@ struct sigmatel_spec { unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; + unsigned int powerdown_adcs; /* i/o switches */ unsigned int io_switch[2]; @@ -2010,6 +2014,8 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + if (spec->stream_delay) + msleep(spec->stream_delay); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } @@ -2073,9 +2079,14 @@ static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->adc_nids[substream->number]; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); + if (spec->powerdown_adcs) { + msleep(40); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); return 0; } @@ -2084,8 +2095,12 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->adc_nids[substream->number]; - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); + snd_hda_codec_cleanup_stream(codec, nid); + if (spec->powerdown_adcs) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); return 0; } @@ -3275,6 +3290,12 @@ static int stac92xx_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->init); + /* power down adcs initially */ + if (spec->powerdown_adcs) + for (i = 0; i < spec->num_adcs; i++) + snd_hda_codec_write_cache(codec, + spec->adc_nids[i], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ @@ -3909,6 +3930,47 @@ again: return 0; } +#ifdef SND_HDA_NEEDS_RESUME +static void stac92hd71xx_set_power_state(struct hda_codec *codec, int pwr) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + snd_hda_codec_write_cache(codec, codec->afg, 0, + AC_VERB_SET_POWER_STATE, pwr); + + msleep(1); + for (i = 0; i < spec->num_adcs; i++) { + snd_hda_codec_write_cache(codec, + spec->adc_nids[i], 0, + AC_VERB_SET_POWER_STATE, pwr); + } +}; + +static int stac92hd71xx_resume(struct hda_codec *codec) +{ + stac92hd71xx_set_power_state(codec, AC_PWRST_D0); + return stac92xx_resume(codec); +} + +static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state) +{ + stac92hd71xx_set_power_state(codec, AC_PWRST_D3); + return 0; +}; + +#endif + +static struct hda_codec_ops stac92hd71bxx_patch_ops = { + .build_controls = stac92xx_build_controls, + .build_pcms = stac92xx_build_pcms, + .init = stac92xx_init, + .free = stac92xx_free, + .unsol_event = stac92xx_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .resume = stac92hd71xx_resume, + .suspend = stac92hd71xx_suspend, +#endif +}; static int patch_stac92hd71bxx(struct hda_codec *codec) { @@ -3920,6 +3982,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->patch_ops = stac92xx_patch_ops; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; @@ -3951,6 +4014,14 @@ again: spec->init = stac92hd71bxx_core_init; break; case 0x111d7608: /* 5 Port with Analog Mixer */ + if ((codec->revision_id & 0xf) == 0 || + (codec->revision_id & 0xf) == 1) { +#ifdef SND_HDA_NEEDS_RESUME + codec->patch_ops = stac92hd71bxx_patch_ops; +#endif + spec->stream_delay = 40; /* 40 milliseconds */ + } + /* no output amps */ spec->num_pwrs = 0; spec->mixer = stac92hd71bxx_analog_mixer; @@ -3960,6 +4031,13 @@ again: stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ + if ((codec->revision_id & 0xf) == 1) { +#ifdef SND_HDA_NEEDS_RESUME + codec->patch_ops = stac92hd71bxx_patch_ops; +#endif + spec->stream_delay = 40; /* 40 milliseconds */ + } + /* no output amps */ spec->num_pwrs = 0; /* fallthru */ @@ -3976,6 +4054,7 @@ again: spec->gpio_dir = 0x01; spec->gpio_data = 0x01; + spec->powerdown_adcs = 1; spec->digbeep_nid = 0x26; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -4008,8 +4087,6 @@ again: return err; } - codec->patch_ops = stac92xx_patch_ops; - return 0; };

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[alsa-devel] is anybody currently building alsa git on gentoo?
by John L. Utz III 15 Aug '08

15 Aug '08

aclocal revision management is not working, the automake-wrapper thingy is installed, but i get warnings about the lack of an aclocal-1.6 and then patching eventually fails. tnx for any help you can provide. johnu

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[alsa-devel] [PATCH] hda: dynamically create capture mux contols
by Matthew Ranostay 14 Aug '08

14 Aug '08

Dynamically create capture mux volume controls when a output amp is detected. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ff56432..621cc48 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -794,7 +794,9 @@ static struct snd_kcontrol_new stac9200_mixer[] = { STAC_INPUT_SOURCE(1), 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 */ }; @@ -908,14 +910,15 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { 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), - /* analog pc-beep replaced with digital beep support */ /* + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), */ @@ -931,11 +934,15 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { 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), + */ { } /* end */ }; @@ -943,7 +950,9 @@ static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), + */ { } /* end */ }; @@ -957,8 +966,9 @@ static struct snd_kcontrol_new stac9205_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x1A, 0x0, HDA_OUTPUT), - + */ { } /* end */ }; @@ -971,7 +981,9 @@ static struct snd_kcontrol_new stac922x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT), + /* HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x13, 0x0, HDA_OUTPUT), + */ { } /* end */ }; @@ -990,7 +1002,9 @@ static struct snd_kcontrol_new stac927x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x2, 0x17, 0x0, HDA_OUTPUT), + */ { } /* end */ }; @@ -2780,6 +2794,35 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, return 0; } +/* labels for mux inputs */ +static const char *stac92xx_mux_labels[4] = { + "Capture Mux 1", "Capture Mux 2", "Capture Mux 3", + "Capture Mux 4" +}; + +static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int wcaps, nid, i, err = 0; + char name[32]; + + for (i = 0; i < spec->num_muxes; i++) { + nid = spec->mux_nids[i]; + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + sprintf(name, "%s Capture Volume", + stac92xx_mux_labels[i]); + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +}; + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -3078,6 +3121,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; + if (spec->num_muxes > 0) { + err = stac92xx_auto_create_mux_input_ctls(codec); + if (err < 0) + return err; + } spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2)

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Re: [alsa-devel] Swissonic TS22PCI / Teratec Phase 22 / driver problems with ice1724
by Takashi Iwai 14 Aug '08

14 Aug '08

At Thu, 14 Aug 2008 15:59:06 +0200, Manuel Schneider wrote: > > Takashi Iwai schrieb, Am 14.08.2008 15:47: > > > Pass model=phase22 module option to snd-ice1724 driver. Then it > > handles the card as if it's phase22. > > Thanks for this hin. Unfortunately it changes nothing - still kernel > error message, still no usable sound device. Are you sure? With the proper model name, at least you won't get EEPROM version error any more. > > I also tried the following kernel patch, which gives me an accessible > sound device and usable mixer (and no kernel error), but recording does > not work (just noise). Likely an incompatible hardware design for the recording... Takashi > The patch is quite simple - it just adds the subvendor ID of the > Swissonic card and adds all phase22 functionality to it. > > Manuel > > --- > -:[/usr/src/linux/sound/pci/ice1712]#> diff -uN phase.h.bak phase.h > --- phase.h.bak 2008-08-12 10:13:49.000000000 +0200 > +++ phase.h 2008-08-12 09:49:23.000000000 +0200 > @@ -25,9 +25,11 @@ > */ > > #define PHASE_DEVICE_DESC "{Terratec,Phase 22},"\ > + "{Swissonic,TS22PCI},"\ > "{Terratec,Phase 28}," > > #define VT1724_SUBDEVICE_PHASE22 0x3b155011 > +#define VT1724_SUBDEVICE_TS22PCI 0x3b157b11 > #define VT1724_SUBDEVICE_PHASE28 0x3b154911 > > /* entry point */ > --- > > --- > -:[/usr/src/linux/sound/pci/ice1712]#> diff -uN phase.c.bak phase.c > --- phase.c.bak 2008-08-12 09:51:28.000000000 +0200 > +++ phase.c 2008-08-12 09:51:28.000000000 +0200 > @@ -124,6 +124,11 @@ > ice->num_total_adcs = 2; > ice->vt1720 = 1; // Envy24HT-S have 16 bit wide GPIO > break; > + case VT1724_SUBDEVICE_TS22PCI: > + ice->num_total_dacs = 2; > + ice->num_total_adcs = 2; > + ice->vt1720 = 1; // Envy24HT-S have 16 bit wide GPIO > + break; > default: > snd_BUG(); > return -EINVAL; > @@ -139,6 +144,10 @@ > if ((err = snd_ice1712_akm4xxx_init(ak, &akm_phase22, > &akm_phase22_priv, ice)) < 0) > return err; > break; > + case VT1724_SUBDEVICE_TS22PCI: > + if ((err = snd_ice1712_akm4xxx_init(ak, &akm_phase22, > &akm_phase22_priv, ice)) < 0) > + return err; > + break; > } > > return 0; > @@ -153,6 +162,10 @@ > err = snd_ice1712_akm4xxx_build_controls(ice); > if (err < 0) > return err; > + case VT1724_SUBDEVICE_TS22PCI: > + err = snd_ice1712_akm4xxx_build_controls(ice); > + if (err < 0) > + return err; > } > return 0; > } > @@ -896,6 +909,15 @@ > .eeprom_data = phase22_eeprom, > }, > { > + .subvendor = VT1724_SUBDEVICE_TS22PCI, > + .name = "Swissonic TS22PCI", > + .model = "phase22", > + .chip_init = phase22_init, > + .build_controls = phase22_add_controls, > + .eeprom_size = sizeof(phase22_eeprom), > + .eeprom_data = phase22_eeprom, > + }, > + { > .subvendor = VT1724_SUBDEVICE_PHASE28, > .name = "Terratec PHASE 28", > .model = "phase28", > --- > > > > -- > Kanal Ratte 104,5 MHz > ...das Freie Radio im Wiesental > > Hauptstrasse 82 > 79650 Schopfheim > > Tel: +49 7622 6692-53 > Fax: +49 7622 6692-54 >

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[alsa-devel] Swissonic TS22PCI / Teratec Phase 22 / driver problems with ice1724
by Manuel Schneider 14 Aug '08

14 Aug '08

Hi, we had a Teratec Phase 22 on our transmitter line which works great under Linux, using the ice1724 driver from alsa. As the original sound card broke due to electrical surge on the transmitter line, we had to replace it. The Phase 22 is not available anymore, so we bought a Swissonic TS22PCI which is based on the Phase 22: http://www.thomann.de/gb/swissonic_ts22pci.htm The PCB looks exactly the same as the Phase 22, but the driver does not initialize the card correctly: --- dmesg: ice1724: No matching model found for ID 0x3b157b11 ice1724: Invalid EEPROM version 0 AC'97 0 does not respond - RESET AC'97 0 access is not valid [0xffffffff], removing mixer. ice1712: cannot initialize pro ac97, skipped --- I already updated to Linux 2.6.25 and tried the current alsa-drivers package (1.0.17), but I get always the same errors in the kernel log. I also deleted /etc/asound.* and /var/lib/alsa/asound.state, ran alsa-conf - no luck. Do you have any ideas how I can get this card to work? Thanks a lot for your suggestions, more information see attached. Manuel --- -:[~]#> lspci -vv 00:0a.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) Subsystem: TERRATEC Electronic GmbH Unknown device 117b Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 32 Interrupt: pin A routed to IRQ 18 Region 0: I/O ports at c000 [size=32] Region 1: I/O ports at bc00 [size=128] Capabilities: [80] Power Management version 1 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- --- -:[~]#> cat /etc/modules.d/alsa alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # --- BEGIN: Generated by ALSACONF, do not edit. --- # --- ALSACONF version 1.0.16 --- alias snd-card-0 snd-ice1724 alias sound-slot-0 snd-ice1724 # --- END: Generated by ALSACONF, do not edit. --- --- -:[~]#> lsmod Module Size Used by snd_pcm_oss 24224 0 snd_mixer_oss 14720 1 snd_pcm_oss snd_ice1724 92328 0 snd_ice17xx_ak4xxx 3648 1 snd_ice1724 snd_ac97_codec 105704 1 snd_ice1724 snd_ak4114 8448 1 snd_ice1724 snd_pcm 59660 4 snd_pcm_oss,snd_ice1724,snd_ac97_codec,snd_ak4114 snd_timer 19272 1 snd_pcm snd_page_alloc 8272 1 snd_pcm snd_pt2258 4032 1 snd_ice1724 snd_i2c 5312 2 snd_ice1724,snd_pt2258 snd_ak4xxx_adda 8512 2 snd_ice1724,snd_ice17xx_ak4xxx snd_mpu401_uart 6912 1 snd_ice1724 snd_rawmidi 19808 1 snd_mpu401_uart snd 49288 12 snd_pcm_oss,snd_mixer_oss,snd_ice1724,snd_ac97_codec,snd_ak4114,snd_pcm,snd_timer,snd_pt2258,snd_i2c,snd_ak4xxx_adda,snd_mpu401_uart,snd_rawmidi capi 14272 0 capifs 5072 2 capi kernelcapi 31936 1 capi nfs 118264 3 lockd 62064 2 nfs sunrpc 167432 17 nfs,lockd rtc 11384 0 via_rhine 19976 0 mii 4928 1 via_rhine ac97_bus 1984 1 snd_ac97_codec soundcore 6624 1 snd --- -:[~]#> modinfo snd_ice1724 filename: /lib/modules/2.6.25-gentoo-r7/kernel/sound/pci/ice1712/snd-ice1724.ko license: GPL description: VIA ICEnsemble ICE1724/1720 (Envy24HT/PT) author: Jaroslav Kysela <perex(a)perex.cz> alias: pci:v00001412d00001724sv*sd*bc*sc*i* depends: snd,snd-ac97-codec,snd-pcm,snd-i2c,snd-mpu401-uart,snd-ice17xx-ak4xxx,snd-pt2258,snd-ak4xxx-adda,snd-ak4114 vermagic: 2.6.25-gentoo-r7 mod_unload parm: index:Index value for ICE1724 soundcard. (array of int) parm: id:ID string for ICE1724 soundcard. (array of charp) parm: enable:Enable ICE1724 soundcard. (array of bool) parm: model:Use the given board model. (array of charp) --- -:[~]#> cat /proc/asound/cards 0 [ICE1724 ]: ICE1724 - ICEnsemble ICE1724 ICEnsemble ICE1724 at 0xc000, irq 18 --- -:[~]#> cat /proc/asound/ICE1724/ice1724 ICEnsemble ICE1724 at 0xc000, irq 18 EEPROM: Subvendor : 0x3b157b11 Size : 32 bytes Version : 0 System Config : 0x0 ACLink : 0x0 I2S : 0x0 S/PDIF : 0x0 GPIO direction : 0xffff00 GPIO mask : 0xffffff GPIO state : 0xffffff Extra #18 : 0xff Extra #19 : 0xff Extra #20 : 0xff Extra #21 : 0xff Extra #22 : 0xff Extra #23 : 0xff Extra #24 : 0xff Extra #25 : 0xff Extra #26 : 0x0 Extra #27 : 0x0 Extra #28 : 0x0 Extra #29 : 0x0 Extra #30 : 0x0 Extra #31 : 0x0 ... --- -- Kanal Ratte 104,5 MHz ...das Freie Radio im Wiesental Hauptstrasse 82 79650 Schopfheim Tel: +49 7622 6692-53 Fax: +49 7622 6692-54

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[alsa-devel] Alsa-driver compile fail on Ubuntu powerpc
by nguyen tritai 14 Aug '08

14 Aug '08

Hi all, I'm trying to bring up audio on powerpc AMCC 440EPX board with the soundcard Sound Blaster Audigy CA0106. The board is runing Ubuntu 7.04. I configured: ./configure --with-cards=ca0106 --with-sequencer=yes --with-kernel=/usr/src/linux Then I make and get issue: make[2]: Leaving directory `/root/alsa-driver-1.0.17/usb' make[2]: Entering directory `/root/alsa-driver-1.0.17/misc' make[2]: Leaving directory `/root/alsa-driver-1.0.17/misc' make[1]: Leaving directory `/root/alsa-driver-1.0.17' make -C /usr/src/linux SUBDIRS=/root/alsa-driver-1.0.17 CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/linux' CC [M] /root/alsa-driver-1.0.17/acore/memory_wrapper.o make[1]: Leaving directory `/usr/src/linux' root@sequoia:~/alsa-driver-1.0.17# vim log In file included from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: /root/alsa-driver-1.0.17/include/adriver.h:278:1: warning: "GFP_DMA32" redefined In file included from include/linux/slab.h:14, from include/linux/percpu.h:5, from include/linux/rcupdate.h:41, from include/linux/pid.h:4, from include/linux/sched.h:76, from /root/alsa-driver-1.0.17/include/asm/elf.h:5, from include/linux/elf.h:7, from include/linux/module.h:14, from /root/alsa-driver-1.0.17/include/adriver.h:26, from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: include/linux/gfp.h:105:1: warning: this is the location of the previous definition In file included from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: /root/alsa-driver-1.0.17/include/adriver.h:828: error: static declaration of âjiffies_to_msecsâ follows non-static declaration include/linux/jiffies.h:270: error: previous declaration of âjiffies_to_msecsâ was here /root/alsa-driver-1.0.17/include/adriver.h:847: error: static declaration of âmsecs_to_jiffiesâ follows non-static declaration include/linux/jiffies.h:272: error: previous declaration of âmsecs_to_jiffiesâ was here In file included from /root/alsa-driver-1.0.17/include/adriver.h:944, from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: include/linux/pci.h:594: error: expected identifier or â(â before numeric constant In file included from /root/alsa-driver-1.0.17/include/asm/pci.h:8, from include/linux/pci.h:945, from /root/alsa-driver-1.0.17/include/adriver.h:944, from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: include/linux/mm.h:235: error: conflicting types for âsnd_compat_vmalloc_to_pageâ /root/alsa-driver-1.0.17/include/adriver.h:748: error: previous declaration of âsnd_compat_vmalloc_to_pageâ was here In file included from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: /root/alsa-driver-1.0.17/include/adriver.h: In function âsnd_pci_orig_save_stateâ: /root/alsa-driver-1.0.17/include/adriver.h:1186: error: too many arguments to function âpci_save_stateâ /root/alsa-driver-1.0.17/include/adriver.h: In function âsnd_pci_orig_restore_stateâ: /root/alsa-driver-1.0.17/include/adriver.h:1190: error: too many arguments to function âpci_restore_stateâ In file included from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: /root/alsa-driver-1.0.17/include/adriver.h:1552:1: warning: "page_to_pfn" redefined In file included from /root/alsa-driver-1.0.17/include/asm/page.h:150, from /root/alsa-driver-1.0.17/include/asm/thread_info.h:30, from include/linux/thread_info.h:35, from include/linux/preempt.h:9, from include/linux/spinlock.h:49, from include/linux/seqlock.h:29, from include/linux/time.h:8, from include/linux/stat.h:60, from include/linux/module.h:10, from /root/alsa-driver-1.0.17/include/adriver.h:26, from /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: include/asm-generic/memory_model.h:79:1: warning: this is the location of the previous definition /root/alsa-driver-1.0.17/acore/memory_wrapper.c: In function âsnd_compat_vmalloc_to_pageâ: /root/alsa-driver-1.0.17/acore/memory_wrapper.c:44: error: implicit declaration of function âVMALLOC_VMADDRâ /root/alsa-driver-1.0.17/acore/memory_wrapper.c:50: error: implicit declaration of function âpte_offsetâ /root/alsa-driver-1.0.17/acore/memory_wrapper.c:50: warning: assignment makes pointer from integer without a cast make[3]: *** [/root/alsa-driver-1.0.17/acore/memory_wrapper.o] Error 1 make[2]: *** [/root/alsa-driver-1.0.17/acore] Error 2 make[1]: *** [_module_/root/alsa-driver-1.0.17] Error 2 The issue looks like an error in alsa-driver source code. I searched the error information and It seems there're lots of people having the same issue but no helpful answers. Do anyone knows how to pass the issue? Thanks, TaiNguyen

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[alsa-devel] ASoC machine driver: use more than one at91 ssc
by Simon Tremblay 14 Aug '08

14 Aug '08

Hi, I want to use more than one SSC (2 to be more accurate) to be able to do what is indicated at Figure 4 of page 10/13 of this document http://www.atmel.com/dyn/resources/prod_documents/doc6020.pdf (how to setup SSC with a I2S codec). I saw in sound/soc/at91/eti_b1_wm8731.c use of 1 SSC. Does I need more than one snd_soc_dai_link in my snd_soc_machine? But for what I understand for now, I need one SSC for sound output and 2 SSC for sound input. I don't know if there is some example about how to do that. Simon

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[alsa-devel] BUG_ON warning
by Matthew Ranostay 14 Aug '08

14 Aug '08

Hello Takashi, I'm getting this warning with the latest snapshot from the git repo. Any ideas on what is causing this? -Matt Ranostay WARNING: at /home/mranostay/git/alsa-driver/acore/timer.c:311 snd_timer_close() BUG? (!timer) Pid: 2461, comm: aplay Not tainted 2.6.25.9 #0 [<e08f5ddf>] snd_timer_close+0x30/0x37 [snd_timer] [<e08f5e12>] snd_timer_user_release+0x2c/0x4a [snd_timer] [<c047e0d2>] __fput+0xad/0x15b [<c047e359>] fput+0x17/0x19 [<c047ba44>] filp_close+0x50/0x5a [<c047cb86>] sys_close+0x69/0x9f [<c0405a76>] syscall_call+0x7/0xb [<c0620000>] ? schedule+0x61b/0x668 =======================

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