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

[alsa-devel] Fix capture mixers of ALC662 models
by Herton Ronaldo Krzesinski 19 Dec '07

19 Dec '07

The below patches fixes capture mixers of ALC662 models. The first one applies to current hg tree, the second one to Linus tree (2.6.24-rcX). The commit that added support for ASUS P701 eeepc also changed the mixers of other ALC662 models, duplicating entries for the Capture items, making them to not work anymore. This fixes it by removing duplicated entries using where possible the common alc662_capture_mixer. Also alc662_capture_mixer should use alc662* functions and not alc882 (I checked /proc/asound/card0/codec* on an eepc model and it's ok). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> --- alsa-kernel/pci/hda/patch_realtek.c.orig 2007-12-18 21:57:02.000000000 -0200 +++ alsa-kernel/pci/hda/patch_realtek.c 2007-12-18 22:01:24.000000000 -0200 @@ -12364,17 +12364,6 @@ static struct snd_kcontrol_new alc662_3S HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12398,17 +12387,6 @@ static struct snd_kcontrol_new alc662_3S HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12422,17 +12400,6 @@ static struct snd_kcontrol_new alc662_le HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12463,6 +12430,25 @@ static struct snd_kcontrol_new alc662_ch { } /* end */ }; +/* capture mixer elements */ +static struct snd_kcontrol_new alc662_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc662_mux_enum_info, + .get = alc662_mux_enum_get, + .put = alc662_mux_enum_put, + }, + { } /* end */ +}; + static struct hda_verb alc662_init_verbs[] = { /* ADC: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -12585,26 +12571,6 @@ static struct hda_verb alc662_auto_init_ { } }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc662_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, - { } /* end */ -}; - static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { unsigned int present; @@ -12746,8 +12712,7 @@ static struct alc_config_preset alc662_p .input_mux = &alc662_capture_source, }, [ALC662_5ST_DIG] = { - .mixers = { alc662_base_mixer, alc662_chmode_mixer, - alc662_capture_mixer }, + .mixers = { alc662_base_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, The commit that added support for ASUS P701 eeepc also changed the mixers of other ALC662 models, duplicating entries for the Capture items, making them to not work anymore. This fixes it by removing duplicated entries using where possible the common alc662_capture_mixer. Also alc662_capture_mixer should use alc662* functions and not alc882 (I checked /proc/asound/card0/codec* on an eepc model and it's ok). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1c50278..85e54d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11728,17 +11728,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -11762,17 +11751,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -11786,17 +11764,6 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -11827,6 +11794,25 @@ static struct snd_kcontrol_new alc662_chmode_mixer[] = { { } /* end */ }; +/* capture mixer elements */ +static struct snd_kcontrol_new alc662_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc662_mux_enum_info, + .get = alc662_mux_enum_get, + .put = alc662_mux_enum_put, + }, + { } /* end */ +}; + static struct hda_verb alc662_init_verbs[] = { /* ADC: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -11949,26 +11935,6 @@ static struct hda_verb alc662_auto_init_verbs[] = { { } }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc662_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, - { } /* end */ -}; - static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { unsigned int present; @@ -12109,8 +12075,7 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_5ST_DIG] = { - .mixers = { alc662_base_mixer, alc662_chmode_mixer, - alc662_capture_mixer }, + .mixers = { alc662_base_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, -- []'s Herton

2 3

[alsa-devel] [PATCH] Dmux mixer conflict
by Matthew Ranostay 19 Dec '07

19 Dec '07

DMUX mixers need to be created before input source mixers, otherwise some STAC9228 codecs fail to record correctly or at all. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r fbe3540c8df4 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Dec 18 15:27:34 2007 -0700 @@ -834,15 +834,16 @@ static int stac92xx_build_controls(struc if (err < 0) return err; - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } if (spec->num_dmuxes > 0) { stac_dmux_mixer.count = spec->num_dmuxes; err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&stac_dmux_mixer, codec)); + if (err < 0) + return err; + } + + for (i = 0; i < spec->num_mixers; i++) { + err = snd_hda_add_new_ctls(codec, spec->mixers[i]); if (err < 0) return err; }

2 2

[alsa-devel] rate plugin issue
by Takashi Iwai 19 Dec '07

19 Dec '07

Hi, I committed your patch now on HG tree, as it's likely OK from the possible breakage on other apps, too. Anyway, I swear that I'll never read any mails during my next vacation. That was so exhausting to be dragged to a lengthy discussion for band-aiding a badly designed compoment... :) Takashi

3 11

[alsa-devel] speaker-test stalls
by Dave Dillow 19 Dec '07

19 Dec '07

While working with Fernando to get the sis7019 driver working in his buildroot setup, we kept running into a lockup of the PCM stream when using speaker-test. aplay was fine, as was other I finally tracked it down to a combination of uclibc not doing block reads, which led to XRUNs, and what I thought was issue in snd_pcm_lib_write1() when trying to fill up the tail of the PCM buffer with a large write, when there was an intervening small write. As it turns out, the lockup problem may simply be in speaker-write's sw_params settings -- it calls snd_pcm_sw_params_set_avail_min() to set the minium write to the period size, and snd_pcm_sw_params_set_start_threshold() to start once the buffer has been filled. This doesn't account for odd wave file lengths and XRUNs. The problem arises when the wave data is not a multiple of the period size, and you get an XRUN when there is less than buffer_size samples remaining in the current loop. So for example, period size is 8192, buffer is 32768, and the wave file has 71042 samples (Front_Left.wav for speaker-test). speaker-test will break that into 8 writes of 8192 samples, and one of 5506. If you get a XRUN in the last 32k of the file, the PCM will be stopped, and prepared, and we'll start loading data into again. Because we have a tail of 5506 samples, we get to the point where the available space is less than our minimum, and then snd_pcm_lib_write1() will wait forever for the available space to be >= runtime->avail_min, even though the channel is stopped. [snip from a trace, patch to pcm_lib.c attached] writei size 8192 avail 32768 min 8192 xfer 1 state 2 writei size 8192 avail 24576 min 8192 xfer 1 state 2 writei size 5506 avail 16384 min 8192 xfer 1 state 2 writei size 8192 avail 10878 min 8192 xfer 1 state 2 writei size 8192 avail 2686 min 8192 xfer 1 state 2 writei loop avail 2686 min 8192 writei loop avail 2686 min 8192 writei loop avail 2686 min 8192 writei loop avail 2686 min 8192 Assuming that this is desired behavior from the kernel side, then I see three options to fix this in speaker-test: 1) snd_pcm_sw_params_set_avail_min() to 1 sample 2) snd_pcm_sw_params_set_start_threshold() to start when (nperiods - 1) are ready (or be even more aggressive) 3) call snd_pcm_drain() at the end of each loop to clear out the buffer. This won't help if the wave file has fewer samples than the buffer size. Options 1 or 2 seem to be best. Thoughts? Dave

2 5

[alsa-devel] [PATCH] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h
by Andrew Paprocki 19 Dec '07

19 Dec '07

Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet rev 1.3. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r fbe3540c8df4 -r 54a57ee44eb6 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/hda_codec.c Tue Dec 18 21:48:58 2007 -0500 @@ -1435,7 +1435,8 @@ int snd_hda_create_spdif_out_ctls(struct return err; } codec->spdif_ctls = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } @@ -1482,7 +1483,7 @@ static int snd_hda_spdif_in_status_get(s unsigned short val; unsigned int sbits; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); sbits = convert_to_spdif_status(val); ucontrol->value.iec958.status[0] = sbits; ucontrol->value.iec958.status[1] = sbits >> 8; @@ -1533,7 +1534,8 @@ int snd_hda_create_spdif_in_ctls(struct return err; } codec->spdif_in_enable = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0) & AC_DIG1_ENABLE; return 0; } diff -r fbe3540c8df4 -r 54a57ee44eb6 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/hda_codec.h Tue Dec 18 21:48:58 2007 -0500 @@ -77,12 +77,14 @@ enum { #define AC_VERB_GET_PIN_SENSE 0x0f09 #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -110,6 +112,7 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -190,6 +193,10 @@ enum { #define AC_PINCAP_OUT (1<<4) /* output capable */ #define AC_PINCAP_IN (1<<5) /* input capable */ #define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification, + * but is marked reserved in the Intel HDA specification. + */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ diff -r fbe3540c8df4 -r 54a57ee44eb6 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/patch_realtek.c Tue Dec 18 21:48:58 2007 -0500 @@ -572,7 +572,7 @@ static int alc_spdif_ctrl_get(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, 0x00); + AC_VERB_GET_DIGI_CONVERT_1, 0x00); *valp = (val & mask) != 0; return 0; @@ -586,7 +586,7 @@ static int alc_spdif_ctrl_put(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long val = *ucontrol->value.integer.value; unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, + AC_VERB_GET_DIGI_CONVERT_1, 0x00); /* Set/unset the masked control bit(s) as needed */

2 1

[alsa-devel] Firmware problem
by Adrian McMenamin 19 Dec '07

19 Dec '07

My user - yes there is one - of the aica sound driver on the Dreamcast is having a problem with formware loading. This is what he has sent me - anyone know what is wrong? When loading snd-aica it gives: [ 55.119676] ALSA sound/sh/aica.c:641: ALSA Driver for Yamaha AICA Super Inter then after like 15 or 20 seconds it will give: FATAL: Error inserting snd_aica (/lib/modules/2.6.24-rc4/kernel/sound/sh/snd-aica.ko): Unknown symbol in module, or unknown parameter (see dmesg) and dmesg gives this: [ 54.632531] fill_kobj_path: path = '/module/snd_page_alloc' [ 54.643434] kobject notes: registering. parent: snd_page_alloc, set: <NULL> [ 54.645602] kobject_uevent_env [ 54.646181] kobject filter function caused the event to drop! [ 54.669703] kobject soundcore: registering. parent: <NULL>, set: module [ 54.671737] kobject holders: registering. parent: soundcore, set: <NULL> [ 54.673616] kobject_uevent_env [ 54.675375] kobject filter function caused the event to drop! [ 54.677096] kobject_uevent_env [ 54.678780] fill_kobj_path: path = '/module/soundcore' [ 54.690116] kobject notes: registering. parent: soundcore, set: <NULL> [ 54.690904] kobject_uevent_env [ 54.693891] kobject filter function caused the event to drop! [ 54.695845] device class 'sound': registering [ 54.697748] kobject sound: registering. parent: <NULL>, set: class [ 54.699657] kobject_uevent_env [ 54.701350] fill_kobj_path: path = '/class/sound' [ 54.787597] kobject snd: registering. parent: <NULL>, set: module [ 54.789664] kobject holders: registering. parent: snd, set: <NULL> [ 54.791557] kobject_uevent_env [ 54.793340] kobject filter function caused the event to drop! [ 54.795228] kobject_uevent_env [ 54.797010] fill_kobj_path: path = '/module/snd' [ 54.808092] kobject notes: registering. parent: snd, set: <NULL> [ 54.808886] kobject_uevent_env [ 54.811852] kobject filter function caused the event to drop! [ 54.849584] kobject snd_timer: registering. parent: <NULL>, set: module [ 54.850378] kobject holders: registering. parent: snd_timer, set: <NULL> [ 54.853372] kobject_uevent_env [ 54.855118] kobject filter function caused the event to drop! [ 54.856902] kobject_uevent_env [ 54.858680] fill_kobj_path: path = '/module/snd_timer' [ 54.869736] kobject notes: registering. parent: snd_timer, set: <NULL> [ 54.871802] kobject_uevent_env [ 54.872452] kobject filter function caused the event to drop! [ 54.875569] DEV: registering device: ID = 'timer' [ 54.877400] kobject sound: registering. parent: virtual, set: <NULL> [ 54.881274] kobject_uevent_env [ 54.883351] unset subsystem caused the event to drop! [ 54.886599] kobject timer: registering. parent: sound, set: devices [ 54.888593] kobject_uevent_env [ 54.890420] fill_kobj_path: path = '/devices/virtual/sound/timer' [ 55.000017] kobject snd_pcm: registering. parent: <NULL>, set: module [ 55.002208] kobject holders: registering. parent: snd_pcm, set: <NULL> [ 55.004074] kobject_uevent_env [ 55.004636] kobject filter function caused the event to drop! [ 55.007502] kobject_uevent_env [ 55.009274] fill_kobj_path: path = '/module/snd_pcm' [ 55.020295] kobject notes: registering. parent: snd_pcm, set: <NULL> [ 55.021108] kobject_uevent_env [ 55.024045] kobject filter function caused the event to drop! [ 55.053022] kobject snd_aica: registering. parent: <NULL>, set: module [ 55.055131] kobject holders: registering. parent: snd_aica, set: <NULL> [ 55.057074] kobject_uevent_env [ 55.057643] kobject filter function caused the event to drop! [ 55.060587] kobject_uevent_env [ 55.061260] fill_kobj_path: path = '/module/snd_aica' [ 55.073491] kobject notes: registering. parent: snd_aica, set: <NULL> [ 55.074288] kobject_uevent_env [ 55.077295] kobject filter function caused the event to drop! [ 55.079239] bus platform: add driver AICA [ 55.079929] kobject AICA: registering. parent: <NULL>, set: drivers [ 55.082948] kobject_uevent_env [ 55.084712] fill_kobj_path: path = '/bus/platform/drivers/AICA' [ 55.089946] Registering platform device 'AICA'. Parent at platform [ 55.091828] DEV: registering device: ID = 'AICA' [ 55.093558] kobject AICA: registering. parent: platform, set: devices [ 55.093744] bus platform: add device AICA [ 55.093849] kobject_uevent_env [ 55.093918] fill_kobj_path: path = '/devices/platform/AICA' [ 55.094045] platform: Matched Device AICA with Driver AICA [ 55.094096] platform: Probing driver AICA with device AICA [ 55.094492] DEV: registering device: ID = 'card0' [ 55.094564] kobject sound: registering. parent: AICA, set: <NULL> [ 55.094646] kobject_uevent_env [ 55.094682] unset subsystem caused the event to drop! [ 55.094739] kobject card0: registering. parent: sound, set: devices [ 55.094888] kobject_uevent_env [ 55.094959] fill_kobj_path: path = '/devices/platform/AICA/sound/card0' [ 55.095133] DEV: registering device: ID = 'pcmC0D0p' [ 55.095195] kobject pcmC0D0p: registering. parent: card0, set: devices [ 55.095454] kobject_uevent_env [ 55.095528] fill_kobj_path: path = '/devices/platform/AICA/sound/card0/pcmC0D0p' [ 55.095745] DEV: registering device: ID = 'controlC0' [ 55.095811] kobject controlC0: registering. parent: card0, set: devices [ 55.095980] kobject_uevent_env [ 55.096051] fill_kobj_path: path = '/devices/platform/AICA/sound/card0/controlC0' [ 55.119676] ALSA sound/sh/aica.c:641: ALSA Driver for Yamaha AICA Super Intelligent Sound Processor [ 55.128339] bound device 'AICA' to driver 'AICA' [ 55.128394] platform: Bound Device AICA to Driver AICA [ 57.018452] DEV: registering device: ID = 'firmware-AICA' [ 57.019121] kobject firmware: registering. parent: AICA, set: <NULL> [ 57.022447] kobject_uevent_env [ 57.023025] unset subsystem caused the event to drop! [ 57.023099] kobject firmware-AICA: registering. parent: firmware, set: devices [ 57.027620] kobject_uevent_env [ 57.029196] kobject filter function caused the event to drop! [ 57.029842] kobject_uevent_env [ 57.032535] fill_kobj_path: path = '/devices/platform/AICA/firmware/firmware-AICA' [ 115.955604] DEV: Unregistering device. ID = 'firmware-AICA' [ 115.955788] kobject_uevent_env [ 115.955898] fill_kobj_path: path = '/devices/platform/AICA/firmware/firmware-AICA' [ 115.956090] kobject firmware-AICA: cleaning up [ 115.956144] kobject firmware: cleaning up [ 115.960813] kobject holders: unregistering [ 115.960863] kobject_uevent_env [ 115.960900] kobject filter function caused the event to drop! [ 115.960972] kobject holders: cleaning up [ 115.961018] kobject snd_aica: unregistering [ 115.961059] kobject_uevent_env [ 115.961128] fill_kobj_path: path = '/module/snd_aica'

2 1

[alsa-devel] How do I use SNDRV_PCM_INFO_JOINT_DUPLEX?
by Timur Tabi 19 Dec '07

19 Dec '07

I'm working on an ASoC driver. I have a limitation in my hardware that if I have an active substream, I cannot re-program the hardware registers that control the sample size. This means, for example, if I'm playing 16-bit audio, and I want to start capture, I cannot program the sample size into the hardware. My only choice is to have already programmed the capture sample size *before* I started playback. The easiest way to implement this is to make sure that the second stream has the same sample size as the first. That way, I won't need to reprogram the registers, because they'll already be programmed correctly. I discovered the SNDRV_PCM_INFO_JOINT_DUPLEX option, but I can't figure out how to use it. I know it needs to be set in my snd_pcm_hardware.info, but then what? How do I tell ALSA that the "joint" part is just that the sample sizes must be the same? -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] uniwill (0x1558, 0x9075) dup
by Andy Shevchenko 18 Dec '07

18 Dec '07

Hi! I investigated the duplicate problem and try to compare two cases. I don't deep to search of the specifications, but the current variant (with asus-laptop) should be more correct. In last HG snapshot second case already comented. I just noticed on my reseaches. -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

1 0

[alsa-devel] [PATCH] hda: STAC927x DMIC Cleanup
by Matthew Ranostay 18 Dec '07

18 Dec '07

Cleaned up STAC927x and added several subsystem id's for more laptops. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 4a204306b295 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Dec 14 14:42:41 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Dec 14 12:37:16 2007 -0700 @@ -107,6 +107,7 @@ enum { STAC_D965_3ST, STAC_D965_5ST, STAC_DELL_3ST, + STAC_DELL_BIOS, STAC_927X_MODELS }; @@ -1408,22 +1409,24 @@ static unsigned int dell_3st_pin_configs static unsigned int dell_3st_pin_configs[14] = { 0x02211230, 0x02a11220, 0x01a19040, 0x01114210, 0x01111212, 0x01116211, 0x01813050, 0x01112214, - 0x403003fa, 0x40000100, 0x40000100, 0x404003fb, + 0x403003fa, 0x90a60040, 0x90a60040, 0x404003fb, 0x40c003fc, 0x40000100 }; static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { - [STAC_D965_REF] = ref927x_pin_configs, - [STAC_D965_3ST] = d965_3st_pin_configs, - [STAC_D965_5ST] = d965_5st_pin_configs, - [STAC_DELL_3ST] = dell_3st_pin_configs, + [STAC_D965_REF] = ref927x_pin_configs, + [STAC_D965_3ST] = d965_3st_pin_configs, + [STAC_D965_5ST] = d965_5st_pin_configs, + [STAC_DELL_3ST] = dell_3st_pin_configs, + [STAC_DELL_BIOS] = NULL, }; static const char *stac927x_models[STAC_927X_MODELS] = { - [STAC_D965_REF] = "ref", - [STAC_D965_3ST] = "3stack", - [STAC_D965_5ST] = "5stack", - [STAC_DELL_3ST] = "dell-3stack", + [STAC_D965_REF] = "ref", + [STAC_D965_3ST] = "3stack", + [STAC_D965_5ST] = "5stack", + [STAC_DELL_3ST] = "dell-3stack", + [STAC_DELL_BIOS] = "dell-bios", }; static struct snd_pci_quirk stac927x_cfg_tbl[] = { @@ -1450,13 +1453,21 @@ static struct snd_pci_quirk stac927x_cfg SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_D965_3ST), /* Dell 3 stack systems */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_3ST), + /* Dell 3 stack systems with verb table in BIOS */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0243, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_DELL_BIOS), /* 965 based 5 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), @@ -1992,6 +2003,7 @@ static int stac92xx_add_dyn_out_pins(str for (i = 0; i < codec->num_nodes; i++) { wcaps = codec->wcaps[i]; wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wtype == AC_WID_AUD_OUT && !(wcaps & AC_WCAP_DIGITAL)) num_dacs++; } @@ -2079,7 +2091,6 @@ static int stac92xx_auto_fill_dac_nids(s wcaps = snd_hda_param_read(codec, conn[j], AC_PAR_AUDIO_WIDGET_CAP); wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) continue; @@ -3293,8 +3304,11 @@ static int patch_stac927x(struct hda_cod stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0) { - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC927x, using BIOS defaults\n"); + if (spec->board_config < 0 || + stac927x_brd_tbl[spec->board_config] != NULL) { + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + "STAC927x, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); if (err < 0) { stac92xx_free(codec); @@ -3306,59 +3320,46 @@ static int patch_stac927x(struct hda_cod stac92xx_set_config_regs(codec); } + spec->adc_nids = stac927x_adc_nids; + spec->mux_nids = stac927x_mux_nids; + spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + spec->multiout.dac_nids = spec->dac_nids; + switch (spec->board_config) { case STAC_D965_3ST: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + case STAC_D965_5ST: + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->num_dmics = 0; + spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; - case STAC_D965_5ST: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + case STAC_DELL_BIOS: + case STAC_DELL_3ST: + /* GPIO2 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000004; + spec->dmic_nids = stac927x_dmic_nids; + spec->num_dmics = STAC927X_NUM_DMICS; + spec->init = d965_core_init; spec->mixer = stac927x_mixer; + spec->dmux_nids = stac927x_dmux_nids; break; default: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->num_dmics = 0; + spec->init = stac927x_core_init; spec->mixer = stac927x_mixer; } - switch (codec->subsystem_id) { - case 0x10280242: /* STAC 9228 */ - case 0x102801f3: - case 0x1028020A: - case 0x10280209: - spec->dmic_nids = stac927x_dmic_nids; - spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nids = stac927x_dmux_nids; - - /* 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; - - /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; - } - - spec->multiout.dac_nids = spec->dac_nids; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; + stac92xx_enable_gpio_mask(codec); - err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) {

3 2

[alsa-devel] snd-hda-intel: HG regression - NULL pointer dereference
by Ivan N. Zlatev 18 Dec '07

18 Dec '07

When loading snd-hda-intel from Alsa HG I get a NULL pointer dereference and the trace below. The card is with the following codec: Codec: SigmaTel STAC9228 Address: 0 Vendor Id: 0x83847616 Subsystem Id: 0x10280227 Revision Id: 0x100201 The trace: ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2792: autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2796: speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2800: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2808: inputs: mic=0x13, fmic=0x0, line=0xe, fline=0x0, cd=0x0, aux=0x0 ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2088: dac_nids=2 (0x5/0x4/0x0/0x0/0x0) BUG: unable to handle kernel NULL pointer dereference at virtual address 00000000 printing eip: f91cfcb0 *pde = 00000000 Oops: 0000 [#1] SMP last sysfs file: /devices/pci0000:00/0000:00:1c.1/0000:0c:00.0/cmd Modules linked in: snd_hda_intel snd_pcm snd_timer snd_page_alloc snd_hwdep snd soundcore nls_iso8859_1 nls_cp437 vfat fat usb_storage ide_core rndis_host cdc_ether usbnet mii aes_i586 ieee80211_crypt_ccmp af_packet xt_tcpudp xt_pkttype ipt_LOG xt_limit vboxdrv ipt_REJECT xt_state iptable_mangle iptable_nat nf_nat iptable_filter nf_conntrack_ipv4 nf_conntrack nfnetlink ip_tables ip6_tables x_tables cpufreq_conservative cpufreq_userspace cpufreq_powersave acpi_cpufreq speedstep_lib microcode apparmor loop dm_mod rfcomm l2cap nvidia(P) ipw3945 sdhci ohci1394 ieee1394 mmc_core ieee80211 hci_usb ac ieee80211_crypt battery i2c_i801 button tg3 rtc_cmos bluetooth rtc_core sr_mod rtc_lib cdrom intel_agp firmware_class i2c_core agpgart sg usbhid hid ff_memless sd_mod ehci_hcd uhci_hcd usbcore edd ext3 mbcache jbd fan ata_piix ahci libata scsi_mod thermal processor CPU: 1 EIP: 0060:[<f91cfcb0>] Tainted: P N VLI EFLAGS: 00010246 (2.6.22.13-0.3-default #1) EIP is at stac92xx_dmux_enum_put+0x28/0x3c [snd_hda_intel] eax: 00000000 ebx: 00000000 ecx: f54b7014 edx: d7e241ec esi: d7e24014 edi: f71e6014 ebp: 00000000 esp: d9ea1ea0 ds: 007b es: 007b fs: 00d8 gs: 0033 ss: 0068 Process alsactl (pid: 25097, ti=d9ea0000 task=c7457030 task.ti=d9ea0000) Stack: 00000000 d7e240a4 f54b7014 cbe43694 fffffff3 f911c2b3 cbe43694 c2175154 df8a4c14 df8a4d60 df8a4c14 00000000 f54b7014 df8a4dc4 f911cd85 cb940b7c c03c5aec f52693c0 f52693c0 bf8ef020 c2175154 c0172331 c016e87a dfbef2c0 Call Trace: [<f911c2b3>] snd_ctl_elem_write+0xa6/0xe6 [snd] [<f911cd85>] snd_ctl_ioctl+0x494/0x7af [snd] [<c0172331>] chrdev_open+0x0/0x133 [<c016e87a>] __dentry_open+0xe4/0x178 [<c016e988>] nameidata_to_filp+0x24/0x33 [<c016e9ce>] do_filp_open+0x37/0x3e [<f911c8f1>] snd_ctl_ioctl+0x0/0x7af [snd] [<c017a13d>] do_ioctl+0x21/0xa0 [<c017a3f3>] vfs_ioctl+0x237/0x249 [<c017a451>] sys_ioctl+0x4c/0x67 [<c0104e22>] sysenter_past_esp+0x6b/0xa9 [<c02c0000>] unix_bind+0x5e/0x28e ======================= Code: 31 c0 c3 57 89 d1 56 53 83 ec 08 8b 78 60 8b 5a 3c 2b 58 44 8b 77 4c 8d 84 9e 90 00 00 00 8b 96 8c 00 00 00 89 44 24 04 8b 46 6c <0f> b7 04 58 89 04 24 89 f8 e8 99 24 ff ff 5f 5a 5b 5e 5f c3 53 EIP: [<f91cfcb0>] stac92xx_dmux_enum_put+0x28/0x3c [snd_hda_intel] SS:ESP 0068:d9ea1ea0 -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

3 8

[alsa-devel] Status of Dolby Digital on-the-fly encoding? (via a52 plugin)
by Dr. Alexander K. Seewald 18 Dec '07

18 Dec '07

I am trying to output Dolby Digital data via coaxial cable to the Logitech Z-5450 5.1 speaker set. Presumably, a52 is the right plugin for this, taking six channels and combining them into the right format. However, I cannot seem to make the Z-5450 understand the format... it always displays "No digital data" and does not output anything. Now, the documentation is very sparse on this, but states that Dolby Digital is supported, as well as DTS and DTS with 96kHz sampling rate and 24bit samples. The Z-5450 can play digital PCM stereo data, but this only activates left and right speaker (it is possible to set an effect to distribute sound to the other speakers, but this is insufficient for my purpose). I can play digital DVD 5.1 sound by attaching the coaxial cable to a DVD player. As of now, speakertest -c 6 tries to play sound on all speakers, but only left and right respond. For all other speakers, "No digital data" is displayed on the Z-5450 console. This is consistent with the Z-5450 not able to play six-channel PCM data. However, using -Dplug:a52 and .asoundrc = pcm.a52 { type a52 } (with various settings) always shows "No digital data" and no sound output. I had a sample DTS 96/24 file a while ago which was able to activate all speakers, so the data connection is ok. But sadly I have misplaced it; in any case DTS 96/24 support is not what I'd expect in the near future. Has anyone successfully output Dolby Digital (A52, AC3, ..) data via coax to the Z-5450 or a similar model? I'm using the newest alsa drivers and plugins on an 2.6.11, and the stereo-PCM output has been working for years... but this one keeps bugging me. Best, Alex -- Dr. Alexander K. Seewald +43(664)1106886 Seewald Solutions ------------------------------------------------ Information wants to be free; Information also wants to be expensive (S.Brant) --------------- alex.seewald.at ----------------

2 7

[alsa-devel] [PATCH] cs4270: wrong sample rate when CONFIG_SND_SOC_CS4270_VD33_ERRATA is set
by Timur Tabi 18 Dec '07

18 Dec '07

When CONFIG_SND_SOC_CS4270_VD33_ERRATA is set, there was a mismatch between the mclk_ratios[] and cs4270_mode_ratios[] arrays. The two arrays have been merged and code has been shuffled. One side effect is that the cs4270_set_dai_sysclk() and cs4270_set_dai_fmt() functions are available only if I2C has been enabled. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 261 ++++++++++++++++++++++----------------------- 1 files changed, 128 insertions(+), 133 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index abac628..bebb4ba 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -48,12 +48,130 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ }; -/* The number of MCLK/LRCK ratios supported by the CS4270 */ -#define NUM_MCLK_RATIOS 9 +/* + * The codec isn't really big-endian or little-endian, since the I2S + * interface requires data to be sent serially with the MSbit first. + * However, to support BE and LE I2S devices, we specify both here. That + * way, ALSA will always match the bit patterns. + */ +#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#ifdef USE_I2C + +/* CS4270 registers addresses */ +#define CS4270_CHIPID 0x01 /* Chip ID */ +#define CS4270_PWRCTL 0x02 /* Power Control */ +#define CS4270_MODE 0x03 /* Mode Control */ +#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */ +#define CS4270_TRANS 0x05 /* Transition Control */ +#define CS4270_MUTE 0x06 /* Mute Control */ +#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */ +#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */ + +#define CS4270_FIRSTREG 0x01 +#define CS4270_LASTREG 0x08 +#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) -/* The actual MCLK/LRCK ratios, in increasing numerical order */ -static unsigned int mclk_ratios[NUM_MCLK_RATIOS] = - {64, 96, 128, 192, 256, 384, 512, 768, 1024}; +/* Bit masks for the CS4270 registers */ +#define CS4270_CHIPID_ID 0xF0 +#define CS4270_CHIPID_REV 0x0F +#define CS4270_PWRCTL_FREEZE 0x80 +#define CS4270_PWRCTL_PDN_ADC 0x20 +#define CS4270_PWRCTL_PDN_DAC 0x02 +#define CS4270_PWRCTL_PDN 0x01 +#define CS4270_MODE_SPEED_MASK 0x30 +#define CS4270_MODE_1X 0x00 +#define CS4270_MODE_2X 0x10 +#define CS4270_MODE_4X 0x20 +#define CS4270_MODE_SLAVE 0x30 +#define CS4270_MODE_DIV_MASK 0x0E +#define CS4270_MODE_DIV1 0x00 +#define CS4270_MODE_DIV15 0x02 +#define CS4270_MODE_DIV2 0x04 +#define CS4270_MODE_DIV3 0x06 +#define CS4270_MODE_DIV4 0x08 +#define CS4270_MODE_POPGUARD 0x01 +#define CS4270_FORMAT_FREEZE_A 0x80 +#define CS4270_FORMAT_FREEZE_B 0x40 +#define CS4270_FORMAT_LOOPBACK 0x20 +#define CS4270_FORMAT_DAC_MASK 0x18 +#define CS4270_FORMAT_DAC_LJ 0x00 +#define CS4270_FORMAT_DAC_I2S 0x08 +#define CS4270_FORMAT_DAC_RJ16 0x18 +#define CS4270_FORMAT_DAC_RJ24 0x10 +#define CS4270_FORMAT_ADC_MASK 0x01 +#define CS4270_FORMAT_ADC_LJ 0x00 +#define CS4270_FORMAT_ADC_I2S 0x01 +#define CS4270_TRANS_ONE_VOL 0x80 +#define CS4270_TRANS_SOFT 0x40 +#define CS4270_TRANS_ZERO 0x20 +#define CS4270_TRANS_INV_ADC_A 0x08 +#define CS4270_TRANS_INV_ADC_B 0x10 +#define CS4270_TRANS_INV_DAC_A 0x02 +#define CS4270_TRANS_INV_DAC_B 0x04 +#define CS4270_TRANS_DEEMPH 0x01 +#define CS4270_MUTE_AUTO 0x20 +#define CS4270_MUTE_ADC_A 0x08 +#define CS4270_MUTE_ADC_B 0x10 +#define CS4270_MUTE_POLARITY 0x04 +#define CS4270_MUTE_DAC_A 0x01 +#define CS4270_MUTE_DAC_B 0x02 + +/* + * Clock Ratio Selection for Master Mode with I2C enabled + * + * The data for this chart is taken from Table 5 of the CS4270 reference + * manual. + * + * This table is used to determine how to program the Mode Control register. + * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + * + * Each element in this array corresponds to the ratios in mclk_ratios[]. + * These two arrays need to be in sync. + * + * 'speed_mode' is the corresponding bit pattern to be written to the + * MODE bits of the Mode Control Register + * + * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of + * the Mode Control Register. + * + * In situations where a single ratio is represented by multiple speed + * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick + * double-speed instead of quad-speed. However, the CS4270 errata states + * that Divide-By-1.5 can cause failures, so we avoid that mode where + * possible. + * + * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not + * work if VD = 3.3V. If this effects you, select the + * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will + * never select any sample rates that require divide-by-1.5. + */ +static struct { + unsigned int ratio; + u8 speed_mode; + u8 mclk; +} cs4270_mode_ratios[] = { + {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, +#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA + {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, +#endif + {128, CS4270_MODE_2X, CS4270_MODE_DIV1}, + {192, CS4270_MODE_4X, CS4270_MODE_DIV3}, + {256, CS4270_MODE_1X, CS4270_MODE_DIV1}, + {384, CS4270_MODE_2X, CS4270_MODE_DIV3}, + {512, CS4270_MODE_1X, CS4270_MODE_DIV2}, + {768, CS4270_MODE_1X, CS4270_MODE_DIV3}, + {1024, CS4270_MODE_1X, CS4270_MODE_DIV4} +}; + +/* The number of MCLK/LRCK ratios supported by the CS4270 */ +#define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) /* * Determine the CS4270 samples rates. @@ -97,7 +215,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, cs4270->mclk = freq; for (i = 0; i < NUM_MCLK_RATIOS; i++) { - unsigned int rate = freq / mclk_ratios[i]; + unsigned int rate = freq / cs4270_mode_ratios[i].ratio; rates |= snd_pcm_rate_to_rate_bit(rate); if (rate < rate_min) rate_min = rate; @@ -155,80 +273,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, } /* - * The codec isn't really big-endian or little-endian, since the I2S - * interface requires data to be sent serially with the MSbit first. - * However, to support BE and LE I2S devices, we specify both here. That - * way, ALSA will always match the bit patterns. - */ -#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ - SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) - -#ifdef USE_I2C - -/* CS4270 registers addresses */ -#define CS4270_CHIPID 0x01 /* Chip ID */ -#define CS4270_PWRCTL 0x02 /* Power Control */ -#define CS4270_MODE 0x03 /* Mode Control */ -#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */ -#define CS4270_TRANS 0x05 /* Transition Control */ -#define CS4270_MUTE 0x06 /* Mute Control */ -#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */ -#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */ - -#define CS4270_FIRSTREG 0x01 -#define CS4270_LASTREG 0x08 -#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) - -/* Bit masks for the CS4270 registers */ -#define CS4270_CHIPID_ID 0xF0 -#define CS4270_CHIPID_REV 0x0F -#define CS4270_PWRCTL_FREEZE 0x80 -#define CS4270_PWRCTL_PDN_ADC 0x20 -#define CS4270_PWRCTL_PDN_DAC 0x02 -#define CS4270_PWRCTL_PDN 0x01 -#define CS4270_MODE_SPEED_MASK 0x30 -#define CS4270_MODE_1X 0x00 -#define CS4270_MODE_2X 0x10 -#define CS4270_MODE_4X 0x20 -#define CS4270_MODE_SLAVE 0x30 -#define CS4270_MODE_DIV_MASK 0x0E -#define CS4270_MODE_DIV1 0x00 -#define CS4270_MODE_DIV15 0x02 -#define CS4270_MODE_DIV2 0x04 -#define CS4270_MODE_DIV3 0x06 -#define CS4270_MODE_DIV4 0x08 -#define CS4270_MODE_POPGUARD 0x01 -#define CS4270_FORMAT_FREEZE_A 0x80 -#define CS4270_FORMAT_FREEZE_B 0x40 -#define CS4270_FORMAT_LOOPBACK 0x20 -#define CS4270_FORMAT_DAC_MASK 0x18 -#define CS4270_FORMAT_DAC_LJ 0x00 -#define CS4270_FORMAT_DAC_I2S 0x08 -#define CS4270_FORMAT_DAC_RJ16 0x18 -#define CS4270_FORMAT_DAC_RJ24 0x10 -#define CS4270_FORMAT_ADC_MASK 0x01 -#define CS4270_FORMAT_ADC_LJ 0x00 -#define CS4270_FORMAT_ADC_I2S 0x01 -#define CS4270_TRANS_ONE_VOL 0x80 -#define CS4270_TRANS_SOFT 0x40 -#define CS4270_TRANS_ZERO 0x20 -#define CS4270_TRANS_INV_ADC_A 0x08 -#define CS4270_TRANS_INV_ADC_B 0x10 -#define CS4270_TRANS_INV_DAC_A 0x02 -#define CS4270_TRANS_INV_DAC_B 0x04 -#define CS4270_TRANS_DEEMPH 0x01 -#define CS4270_MUTE_AUTO 0x20 -#define CS4270_MUTE_ADC_A 0x08 -#define CS4270_MUTE_ADC_B 0x10 -#define CS4270_MUTE_POLARITY 0x04 -#define CS4270_MUTE_DAC_A 0x01 -#define CS4270_MUTE_DAC_B 0x02 - -/* * A list of addresses on which this CS4270 could use. I2C addresses are * 7 bits. For the CS4270, the upper four bits are always 1001, and the * lower three bits are determined via the AD2, AD1, and AD0 pins @@ -315,53 +359,6 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, } /* - * Clock Ratio Selection for Master Mode with I2C enabled - * - * The data for this chart is taken from Table 5 of the CS4270 reference - * manual. - * - * This table is used to determine how to program the Mode Control register. - * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling - * rates the CS4270 currently supports. - * - * Each element in this array corresponds to the ratios in mclk_ratios[]. - * These two arrays need to be in sync. - * - * 'speed_mode' is the corresponding bit pattern to be written to the - * MODE bits of the Mode Control Register - * - * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of - * the Mode Control Register. - * - * In situations where a single ratio is represented by multiple speed - * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick - * double-speed instead of quad-speed. However, the CS4270 errata states - * that Divide-By-1.5 can cause failures, so we avoid that mode where - * possible. - * - * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not - * work if VD = 3.3V. If this effects you, select the - * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will - * never select any sample rates that require divide-by-1.5. - */ -static struct { - u8 speed_mode; - u8 mclk; -} cs4270_mode_ratios[NUM_MCLK_RATIOS] = { - {CS4270_MODE_4X, CS4270_MODE_DIV1}, /* 64 */ -#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA - {CS4270_MODE_4X, CS4270_MODE_DIV15}, /* 96 */ -#endif - {CS4270_MODE_2X, CS4270_MODE_DIV1}, /* 128 */ - {CS4270_MODE_4X, CS4270_MODE_DIV3}, /* 192 */ - {CS4270_MODE_1X, CS4270_MODE_DIV1}, /* 256 */ - {CS4270_MODE_2X, CS4270_MODE_DIV3}, /* 384 */ - {CS4270_MODE_1X, CS4270_MODE_DIV2}, /* 512 */ - {CS4270_MODE_1X, CS4270_MODE_DIV3}, /* 768 */ - {CS4270_MODE_1X, CS4270_MODE_DIV4} /* 1024 */ -}; - -/* * Program the CS4270 with the given hardware parameters. * * The .dai_ops functions are used to provide board-specific data, like @@ -388,7 +385,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ratio = cs4270->mclk / rate; /* MCLK/LRCK ratio */ for (i = 0; i < NUM_MCLK_RATIOS; i++) { - if (mclk_ratios[i] == ratio) + if (cs4270_mode_ratios[i].ratio == ratio) break; } @@ -669,7 +666,7 @@ error: return ret; } -#endif +#endif /* USE_I2C*/ struct snd_soc_codec_dai cs4270_dai = { .name = "CS4270", @@ -687,10 +684,6 @@ struct snd_soc_codec_dai cs4270_dai = { .rates = 0, .formats = CS4270_FORMATS, }, - .dai_ops = { - .set_sysclk = cs4270_set_dai_sysclk, - .set_fmt = cs4270_set_dai_fmt, - } }; EXPORT_SYMBOL_GPL(cs4270_dai); @@ -752,6 +745,8 @@ static int cs4270_probe(struct platform_device *pdev) if (codec->control_data) { /* Initialize codec ops */ cs4270_dai.ops.hw_params = cs4270_hw_params; + cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk; + cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt; #ifdef CONFIG_SND_SOC_CS4270_HWMUTE cs4270_dai.dai_ops.digital_mute = cs4270_mute; #endif -- 1.5.2.4

2 1

[alsa-devel] rfe: keep SND_PCI_QUIRK groups are sorted by vendor and product id pair
by Andy Shevchenko 17 Dec '07

17 Dec '07

Hi! Some to be discussed. I found the SND_PCI_QUIRK group (lines with this macro under one structure) is unsorted. Almost in all cases no sense to make so. I wrote python script to keep lists sorted. Please, comment this. P.S. Known bug is a deletion of the empty lines under group. #!/usr/bin/python # -*- coding: UTF-8 -*- # vim: ts=4 sw=4 et: """ The sript sorts SND_PCI_QUIRK groups by vendor and product id pair. """ __author__ = "Andy Shevchenko <andy(a)smile.org.ua>" __revision__ = "$Id$" import sys import os import re def read_file(fname): """ Read given file """ fdes = open(fname, 'rt') result = fdes.readlines() fdes.close() return result def save_file(fname, lines): """ Write lines to the given file """ fdes = open(fname, 'wt') fdes.writelines(lines) fdes.flush() fdes.close() def parse_quirks(quirks): """ Parse buffer as SND_PCI_QUIRK group """ rexp_begin = re.compile("^\s*[/\*]*\s*SND_PCI_QUIRK$(?P<vid>[^,]+),\s*(?P<pid>[^,]+)") rexp_end = re.compile("$,\s*(/\*.+\*/)?\s*$") xbuffer = {} # vendor, product: prefix lines, line, postfix lines flag = False vpid = 0 prefix = [] postfix = [] for line in quirks: if line.strip() == '': continue rmatch = rexp_begin.match(line) if rmatch: if flag == True: xbuffer[vpid][2].extend(postfix) postfix = [] flag = False vpid = (int(rmatch.group('vid'), 16) << 16) + int(rmatch.group('pid'), 16) if xbuffer.has_key(vpid): print "Warning: the line '%s' is duplicate!" % line.strip() xbuffer[vpid][2].append(line) else: xbuffer[vpid] = [prefix, line, []] prefix = [] if not rexp_end.search(line): flag = True else: if flag == False: prefix.append(line) else: postfix.append(line) return xbuffer def sort_quirks(quirks): """ Sort SND_PCI_QUIRK group by vendor and product id pair """ xbuffer = [] keys = quirks.keys() keys.sort() for key in keys: quirk = quirks[key] xbuffer.extend(quirk[0]) xbuffer.append(quirk[1]) xbuffer.extend(quirk[2]) return xbuffer def catch_quirk_lines(lines): """ Catch SND_PCI_QUIRK group from lines and sort it """ rexp_begin = re.compile("\s*struct\s+snd_pci_quirk\s+.+\s*=\s*{\s*$") flag = False xbuffer = [] quirks = [] for line in lines: if flag == False and rexp_begin.search(line): xbuffer.append(line) quirks.extend(xbuffer) flag = True xbuffer = [] continue if flag == True and line.strip() == '{}': quirks.extend(sort_quirks(parse_quirks(xbuffer))) flag = False xbuffer = [] xbuffer.append(line) quirks.extend(xbuffer) return quirks def main(): """ MAIN """ try: fname = sys.argv[1] except IndexError: print "Usage: %s <input file>" % os.path.basename(sys.argv[0]) sys.exit(1) lines = read_file(fname) quirks = catch_quirk_lines(lines) save_file('%s.sorted' % fname, quirks) if __name__ == '__main__': main() # End of file -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

2 2

[alsa-devel] Is the mixer interface in the kernel protect against concurrent access?
by Hans-Christian Egtvedt 17 Dec '07

17 Dec '07

Hi, I am solving a locking bug in the at73c213 SPI sound driver, and wondered if the mixer API is protected against concurrent access in ALSA? Or should I implement a mutex to protect my hardware against concurrent access? -- Mvh Hans-Christian Egtvedt

2 2

[alsa-devel] Notebook Toshiba satellite x200-21g with realtek snd-hda-intel audio device still not supported (codec ALC268)
by Daniele (Mastro) 17 Dec '07

17 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi all i'm now to this mailing list i've bought a laptop and i would like to help you supporting my sound card what works: 2 speaker, low volume (it should be a 5+1 audio card) what doesn't - -) low volume - -) inoperating headphone jack - -) not working mic - -) no subwoofer - -) may be some other stuff not tested ( SPDIF? hardware midi?) with speaker-test -t wav -c 6 only front left/right channel works my kernel is a debian kernel Linux Luffy 2.6.23-1-686-bigmem #1 SMP Wed Dec 5 02:42:05 UTC 2007 i686 GNU/Linux some info about the audio card: lspci -n 00:1b.0 0403: 8086:284b (rev 03) lspci -vvv 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Toshiba America Info Systems Unknown device ff00 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f6600000 (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 Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) cat /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff0a Revision Id: 0x100003 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 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x0281103e: [Jack] Line In at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4016852d: [N/A] Speaker at Ext N/A Conn = Digital, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x8a 0x8a] Connection: 7 0x18 0x19 0x1a* 0x1c 0x14 0x15 0x13 many of this think doesn't mean a thing to me but i will be glad to help you if you ask.. i don't mind compiling some other driver/kernel/tool for this just ask me one last note: searching the internet i've found this page: https://wiki.ubuntu.com/Gutsy_Intel_HD_Audio_Controller if you go down you find that the Toshiba Satellite x205-S7483 has the exact same problem here. I think they have the same card searching the Realtek web site i didn't find any 6 channel audio card with the ALC268 codec may be this mean the ALC268 is not the right codec for my audio card or this is a really new model/heavily Toshiba modified device please ask me if you need any more info or want me to do some test thanks you all for your work - -- Daniele -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHZTelgSF24OYDe4YRAm9cAJ0cWzwxUT4QZWAuTYi0ajyteb8dUgCguWlz 9rvQrAqdJBZRfHQ5yRU5PaU= =4wBW -----END PGP SIGNATURE----- - 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

2 2

[alsa-devel] [RFC PATCH] alsactl: skip inactive controls
by Dave Dillow 17 Dec '07

17 Dec '07

When using alsactl to save or restore the card settings, it currently skips over controls that don't have the appropriate read/write permissions. It should also skip over inactive controls, otherwise it will get an error when it tries to access that control, and will fail to save the card state (or fully restore it.) --- I noticed this while adding surround support to SiS7019. I think the problem affects other cards, as I am following the pattern used by trident, rme*, cmipci, emu10k1 and others. This is only for discussion, since I've not tested this (nor even compiled it) but I think it will fix the issue. diff -r d3ba62cd2776 alsactl/state.c --- a/alsactl/state.c Fri Dec 14 15:29:51 2007 +0100 +++ b/alsactl/state.c Sat Dec 15 14:06:35 2007 -0500 @@ -250,7 +250,8 @@ static int get_control(snd_ctl_t *handle return err; } - if (!snd_ctl_elem_info_is_readable(info)) + if (snd_ctl_elem_info_is_inactive(info) || + !snd_ctl_elem_info_is_readable(info)) return 0; snd_ctl_elem_value_set_id(ctl, id); err = snd_ctl_elem_read(handle, ctl); @@ -1306,7 +1307,8 @@ static int set_control(snd_ctl_t *handle } } - if (!snd_ctl_elem_info_is_writable(info)) + if (snd_ctl_elem_info_is_inactive(info) || + !snd_ctl_elem_info_is_writable(info)) return 0; snd_ctl_elem_value_set_numid(ctl, numid1);

2 1

[alsa-devel] [PATCH] hda-codec - Fix definition of AC_KNBCAP_DELTA to match spec
by Andrew Paprocki 17 Dec '07

17 Dec '07

AC_KNBCAP_DELTA is incorrectly defined as (1<<8). According to the Intel HDA spec, this is bit 7 after AC_KNBCAP_NUM_STEPS which is a 0x7f mask. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r 4a204306b295 -r 3681a8bf780e pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Fri Dec 14 14:42:41 2007 +0100 +++ b/pci/hda/hda_codec.h Fri Dec 14 12:27:35 2007 -0500 @@ -234,7 +234,7 @@ enum { /* Volume knobs capabilities */ #define AC_KNBCAP_NUM_STEPS (0x7f<<0) -#define AC_KNBCAP_DELTA (1<<8) +#define AC_KNBCAP_DELTA (1<<7) /* * Control Parameters

2 1

[alsa-devel] [PATCH] alsa-lib: Broken SNDRV_PCM_IOCTL_TTSTAMP entry
by Matthew Ranostay 17 Dec '07

17 Dec '07

Fixed a IOCTL entry that was broken with a recent update to alsa-driver. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 0b2df0817fb6 include/sound/asound.h --- a/include/sound/asound.h Fri Dec 14 15:19:03 2007 +0100 +++ b/include/sound/asound.h Sun Dec 16 19:38:55 2007 -0700 @@ -463,7 +463,7 @@ enum { enum { SNDRV_PCM_IOCTL_PVERSION = _IOR('A', 0x00, int), SNDRV_PCM_IOCTL_INFO = _IOR('A', 0x01, struct sndrv_pcm_info), - SNDRV_PCM_IOCTL_TSTAMP = _IOW('A', 0x02, int), + SNDRV_PCM_IOCTL_TTSTAMP = _IOW('A', 0x03, int), SNDRV_PCM_IOCTL_HW_REFINE = _IOWR('A', 0x10, struct sndrv_pcm_hw_params), SNDRV_PCM_IOCTL_HW_PARAMS = _IOWR('A', 0x11, struct sndrv_pcm_hw_params), SNDRV_PCM_IOCTL_HW_FREE = _IO('A', 0x12), diff -r 0b2df0817fb6 src/pcm/pcm_hw.c --- a/src/pcm/pcm_hw.c Fri Dec 14 15:19:03 2007 +0100 +++ b/src/pcm/pcm_hw.c Sun Dec 16 19:38:55 2007 -0700 @@ -978,9 +978,9 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp, if (SNDRV_PROTOCOL_VERSION(2, 0, 5) <= ver) { int on = 1; - if (ioctl(fd, SNDRV_PCM_IOCTL_TSTAMP, &on) < 0) { + if (ioctl(fd, SNDRV_PCM_IOCTL_TTSTAMP, &on) < 0) { ret = -errno; - SNDMSG("TSTAMP failed\n"); + SNDMSG("TTSTAMP failed\n"); return ret; } }

2 1

[alsa-devel] Altec Lansing FX5051 USB Speakers
by Jeremy Nickurak 16 Dec '07

16 Dec '07

Some number of months ago I posted a request here for any suggestions in getting a set of USB speakers from Altec Lansing, model# FX5051, working with ALSA. I wasn't able to follow up in detail at the time, so I'm hoping try again. They're a set of 5.1 speakers with a built in USB sound card. Sound does appear to work, but the mixer is locked at 100%. There is no analog manual mixer, and the digital control dial exhibits the same problem as alsamixer, so I have to find a way to adjust the usb device's mixer volume in order to make the speakers usable. The speakers work fine in Windows XP, without installing any drivers, so it would seem that this device is operating based on some standard. In windows, adjusting the mixer via the digital control dial, or via the windows mixer control both result in the LED display moving in sync to indicate the new volume setting. In alsa, adjusting the digital control dial doesn't appear to have any effect, other then waking the LED display from its low-light mode. Attempting to adjust volume from alsamixer likewise wakes the display from low-light mode, but also fails to significantly change the mixer setting. When I say significantly, I mean that the mixer flips between about 98% and 100%, when using the Up/Down keys (or PgUp or PgDown). End appears to drop the volume to 0%, but Home doesn't restore it. Pressing the device's 'Power' button also behaves strangely. I can turn the device off, but it is turned back on shortly afterward. No usb connect/disconnect events appear here, instead, the power button seems to function more like a mute. amixer -c 2 contents yields: atrus@kedri:~$ amixer -c 2 contents numid=3,iface=MIXER,name='Tone Control - Bass' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=1 numid=4,iface=MIXER,name='Tone Control - Treble' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=3 numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=49,step=0 : values=48,48 | dBscale-min=-49.00dB,step=1.00dB,mute=0 I'm attaching the contents of 'lsusb -v'. Any suggestions would be appreciated, and any requests for further information about this device will be accomodated ASAP. Thanks in advance. -- Jeremy Nickurak -= Email/Jabber: atrus(a)rifetech.com =-

1 1

[alsa-devel] [PATCH v3 1/2] sis7019: support the SiS 7019 Audio Accelerator
by Dave Dillow 14 Dec '07

14 Dec '07

Basic audio support for the SiS 7019 Audio Accelerator as found in the SiS 55x SoC. There is currently no synth support at the moment, but audio playback and capture with two periods per buffer has seen extensive use. Arbitrary period and buffer sizes (with multiple periods per buffer) have seen light testing, but are believed to be production ready. Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-kernel hg repo. Expect surround sound and S/PDIF support soon. I plan on doing that as a separate patch, as that gives a bisection point for bug-hunting. I'm not tied to that, though. Changes since last posting: * Fix the use of the capture timing channel (multiple periods per buffer) * Fix checkpatch warnings * Remove space from the name pci/sis7019.c | 1461 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ pci/sis7019.h | 342 +++++++++++++ pci/Kconfig | 10 pci/Makefile | 2 4 files changed, 1815 insertions(+) diff -r eb09731e73d7 pci/Kconfig --- a/pci/Kconfig Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/Kconfig Thu Dec 13 21:28:56 2007 -0500 @@ -802,6 +802,16 @@ config SND_RME9652 To compile this driver as a module, choose M here: the module will be called snd-rme9652. +config SND_SIS7019 + tristate "SiS 7019 Audio Accelerator" + depends on SND && X86 && !X86_64 + select SND_AC97_CODEC + help + Say Y here to include support for the SiS 7019 Audio Accelerator. + + To compile this driver as a module, choose M here: the module + will be called snd-sis7019. + config SND_SONICVIBES tristate "S3 SonicVibes" depends on SND diff -r eb09731e73d7 pci/Makefile --- a/pci/Makefile Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/Makefile Thu Dec 13 21:28:56 2007 -0500 @@ -23,6 +23,7 @@ snd-maestro3-objs := maestro3.o snd-maestro3-objs := maestro3.o snd-rme32-objs := rme32.o snd-rme96-objs := rme96.o +snd-sis7019-objs := sis7019.o snd-sonicvibes-objs := sonicvibes.o snd-via82xx-objs := via82xx.o snd-via82xx-modem-objs := via82xx_modem.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_MAESTRO3) += snd-maestr obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o obj-$(CONFIG_SND_RME32) += snd-rme32.o obj-$(CONFIG_SND_RME96) += snd-rme96.o +obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o diff -r eb09731e73d7 pci/sis7019.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.c Thu Dec 13 21:28:56 2007 -0500 @@ -0,0 +1,1461 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * 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. + * + * 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 + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/time.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include "sis7019.h" + +MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); +MODULE_DESCRIPTION("SiS7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = 1; + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is one or two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the SSO on the playback + * channel to clock out virtual periods, and adjust the virtual period length + * to maintain synchronization. This algorithm came from the Trident driver. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u16 sync_period_size; + u16 sync_buffer_size; + u32 sso; + u32 vperiod; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +/* We need four pages to store our wave parameters during a suspend. If + * we're not doing power management, we still need to allocate a page + * for the silence buffer. + */ +#ifdef CONFIG_PM +#define SIS_SUSPEND_PAGES 4 +#else +#define SIS_SUSPEND_PAGES 1 +#endif + +struct sis7019 { + unsigned long ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* Protect against more than one thread hitting the AC97 + * registers (in a more polite manner than pounding the hardware + * semaphore) + */ + struct mutex ac97_mutex; + + /* voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; + + /* Allocate pages to store the internal wave state during + * suspends. When we're operating, this can be used as a silence + * buffer for a timing channel. + */ + void *suspend_state[SIS_SUSPEND_PAGES]; + + int silence_users; + dma_addr_t silence_dma_addr; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +/* The HW offset parameters (Loop End, Stop Sample, End Sample) have a + * documented range of 8-0xfff8 samples. Given that they are 0-based, + * that places our period/buffer range at 9-0xfff9 samples. That makes the + * max buffer size 0xfff9 samples * 2 channels * 2 bytes per sample, and + * max samples / min samples gives us the max periods in a buffer. + * + * We'll add a constraint upon open that limits the period and buffer sample + * size to values that are legal for the hardware. + */ +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static void sis_update_sso(struct voice *voice, u16 period) +{ + void __iomem *base = voice->ctrl_base; + + voice->sso += period; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* Enforce the documented hardware minimum offset */ + if (voice->sso < 8) + voice->sso = 8; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); +} + +static void sis_update_voice(struct voice *voice) +{ + if (voice->flags & VOICE_SSO_TIMING) { + sis_update_sso(voice, voice->period_size); + } else if (voice->flags & VOICE_SYNC_TIMING) { + int sync; + + /* If we've not hit the end of the virtual period, update + * our records and keep going. + */ + if (voice->vperiod > voice->period_size) { + voice->vperiod -= voice->period_size; + if (voice->vperiod < voice->period_size) + sis_update_sso(voice, voice->vperiod); + else + sis_update_sso(voice, voice->period_size); + return; + } + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = voice->sync_cso; + sync -= readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + if (sync > (voice->sync_buffer_size / 2)) + sync -= voice->sync_buffer_size; + + /* If sync is positive, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + */ + if (sync > 0) { + if (sync < 16) + sync = 16; + sis_update_sso(voice, sync); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * a minimum wait time, which makes us quite late here, + * eating into the user's time to refresh the buffer, esp. + * if using small periods. + * + * If we're less than 9 samples behind, we're on target. + */ + if (sync > -9) + voice->vperiod = voice->sync_period_size + 1; + else + voice->vperiod = voice->sync_period_size - 4; + + if (voice->vperiod < voice->buffer_size) { + sis_update_sso(voice, voice->vperiod); + voice->vperiod = 0; + } else + sis_update_sso(voice, voice->period_size); + + sync = voice->sync_cso + voice->sync_period_size; + if (sync >= voice->sync_buffer_size) + sync -= voice->sync_buffer_size; + voice->sync_cso = sync; + } + + snd_pcm_period_elapsed(voice->substream); +} + +static void sis_voice_irq(u32 status, struct voice *voice) +{ + int bit; + + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); + voice++; + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + do { + status = inl(io + SIS_PISR_A); + if (status) { + sis_voice_irq(status, sis->voices); + outl(status, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + sis_voice_irq(status, &sis->voices[32]); + outl(status, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void __sis_map_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + if (!sis->silence_users) + sis->silence_dma_addr = pci_map_single(sis->pci, + sis->suspend_state[0], + 4096, PCI_DMA_TODEVICE); + sis->silence_users++; +} + +static void __sis_unmap_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + sis->silence_users--; + if (!sis->silence_users) + pci_unmap_single(sis->pci, sis->silence_dma_addr, 4096, + PCI_DMA_TODEVICE); +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + __sis_unmap_silence(sis); + voice->timing->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *__sis_alloc_playback_voice(struct sis7019 *sis) +{ + /* Must hold the voice_lock on entry */ + struct voice *voice; + int i; + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + return voice; +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + voice = __sis_alloc_playback_voice(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + unsigned int period_size, buffer_size; + unsigned long flags; + int needed; + + /* If there are one or two periods per buffer, we don't need a + * timing voice, as we can use the capture channel's interrupts + * to clock out the periods. + */ + period_size = params_period_size(hw_params); + buffer_size = params_buffer_size(hw_params); + needed = (period_size != buffer_size && + period_size != (buffer_size / 2)); + + if (needed && !voice->timing) { + spin_lock_irqsave(&sis->voice_lock, flags); + voice->timing = __sis_alloc_playback_voice(sis); + if (voice->timing) + __sis_map_silence(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + /* The baseline setup is for a single period per buffer, and + * we add bells and whistles as needed from there. + */ + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + if (runtime->period_size == (runtime->buffer_size / 2)) { + control |= SIS_PLAY_DMA_INTR_AT_MLP; + } else if (runtime->period_size != runtime->buffer_size) { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size - 1; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control &= ~SIS_PLAY_DMA_INTR_AT_LEO; + control |= SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso |= (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + /* Ok, we're ready to go, set up the channel. + */ + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + /* No locks needed, as the PCM core will hold the locks on the + * substreams, and the HW will only start/stop the indicated voices + * without changing the state of the others. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static void sis_prepare_timing_voice(struct voice *voice, + struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *timing = voice->timing; + void __iomem *play_base = timing->ctrl_base; + void __iomem *wave_base = timing->wave_base; + u16 buffer_size, period_size; + u32 format, control, sso_eso, delta; + u32 vperiod, sso, reg; + + /* Set our initial buffer and period as large as we can given a + * single page of silence. + */ + buffer_size = 4096 / runtime->channels; + buffer_size /= snd_pcm_format_size(runtime->format, 1); + period_size = buffer_size; + + /* Initially, we want to interrupt just a bit behind the end of + * the period we're clocking out. 10 samples seems to give a good + * delay. + * + * We want to spread our interrupts throughout the virtual period, + * so that we don't end up with two interrupts back to back at the + * end -- this helps minimize the effects of any jitter. Adjust our + * clocking period size so that the last period is at least a fourth + * of a full period. + * + * This is all moot if we don't need to use virtual periods. + */ + vperiod = runtime->period_size + 10; + if (vperiod > period_size) { + u16 tail = vperiod % period_size; + u16 quarter_period = period_size / 4; + + if (tail && tail < quarter_period) { + u16 loops = vperiod / period_size; + + tail = quarter_period - tail; + tail += loops - 1; + tail /= loops; + period_size -= tail; + } + + sso = period_size - 1; + } else { + /* The initial period will fit inside the buffer, so we + * don't need to use virtual periods -- disable them. + */ + period_size = runtime->period_size; + sso = vperiod - 1; + vperiod = 0; + } + + /* The interrupt handler implements the timing syncronization, so + * setup its state. + */ + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_base = voice->ctrl_base; + timing->sync_cso = runtime->period_size - 1; + timing->sync_period_size = runtime->period_size; + timing->sync_buffer_size = runtime->buffer_size; + timing->period_size = period_size; + timing->buffer_size = buffer_size; + timing->sso = sso; + timing->vperiod = vperiod; + + /* Using unsigned samples with the all-zero silence buffer + * forces the output to the lower rail, killing playback. + * So ignore unsigned vs signed -- it doesn't change the timing. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + control = timing->buffer_size - 1; + control |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso = timing->buffer_size - 1; + sso_eso |= timing->sso << 16; + + delta = sis_rate_to_delta(runtime->rate); + + /* We've done the math, now configure the channel. + */ + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(sis->silence_dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(control, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *rec_base = voice->ctrl_base; + u32 format, dma_addr, control; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_CAPTURE_DMA_LOOP; + + /* If we've got more than two periods per buffer, then we have + * use a timing voice to clock out the periods. Otherwise, we can + * use the capture channel's interrupts. + */ + if (voice->timing) { + sis_prepare_timing_voice(voice, substream); + } else { + control |= SIS_CAPTURE_DMA_INTR_AT_LEO; + if (runtime->period_size != runtime->buffer_size) + control |= SIS_CAPTURE_DMA_INTR_AT_MLP; + } + + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(control, rec_base + SIS_CAPTURE_DMA_CONTROL); + + /* Force the writes to post. */ + readl(rec_base); + + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + /* We have 64 voices, and the driver currently records from + * only one channel, though that could change in the future. + */ + rc = snd_pcm_new(sis->card, "SiS7019", 0, 64, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + + /* Get the AC97 semaphore -- software first, so we don't spin + * pounding out IO reads on the hardware semaphore... + */ + mutex_lock(&sis->ac97_mutex); + + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + mutex_unlock(&sis->ac97_mutex); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static void sis_free_suspend(struct sis7019 *sis) +{ + int i; + + for (i = 0; i < SIS_SUSPEND_PAGES; i++) + kfree(sis->suspend_state[i]); +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + sis_free_suspend(sis); + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + /* All done, let go of the semaphore, and check for errors + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!sis->codecs_present || !count) + return -EIO; + + /* Let the hardware know that the audio driver is alive, + * and enable PCM slots on the AC-link for L/R playback (3 & 4) and + * record channels. We're going to want to use Variable Rate Audio + * for recording, to avoid needlessly resampling from 48kHZ. + */ + outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF); + outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE | + SIS_AC97_CONF_PCM_CAP_MIC_ENABLE | + SIS_AC97_CONF_PCM_CAP_LR_ENABLE | + SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF); + + /* All AC97 PCM slots should be sourced from sub-mixer 0. + */ + outl(0, io + SIS_AC97_PSR); + + /* There is only one valid DMA setup for a PCI environment. + */ + outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); + + /* Reset the syncronization groups for all of the channels + * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. + * we'll need to change how we handle these. Until then, we just + * assign sub-mixer 0 to all playback channels, and avoid any + * attenuation on the audio. + */ + outl(0, io + SIS_PLAY_SYNC_GROUP_A); + outl(0, io + SIS_PLAY_SYNC_GROUP_B); + outl(0, io + SIS_PLAY_SYNC_GROUP_C); + outl(0, io + SIS_PLAY_SYNC_GROUP_D); + outl(0, io + SIS_MIXER_SYNC_GROUP); + + for (i = 0; i < 64; i++) { + writel(i, SIS_MIXER_START_ADDR(ioaddr, i)); + writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN | + SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i)); + } + + /* Don't attenuate any audio set for the wave amplifier. + * + * FIXME: Maximum attenuation is set for the music amp, which will + * need to change if we start using the synth engine. + */ + outl(0xffff0000, io + SIS_WEVCR); + + /* Ensure that the wave engine is in normal operating mode. + */ + outl(0, io + SIS_WECCR); + + /* Go ahead and enable the DMA interrupts. They won't go live + * until we start a channel. + */ + outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE | + SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER); + + return 0; +} + +#ifdef CONFIG_PM +static int sis_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(sis->pcm); + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[2]); + + /* snd_pcm_suspend_all() stopped all channels, so we're quiescent. + */ + if (sis->irq >= 0) { + synchronize_irq(sis->irq); + free_irq(sis->irq, sis); + sis->irq = -1; + } + + /* Save the internal state away + */ + for (i = 0; i < 4; i++) { + memcpy_fromio(sis->suspend_state[i], ioaddr, 4096); + ioaddr += 4096; + } + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int sis_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "sis7019: unable to re-enable device\n"); + goto error; + } + + if (sis_chip_init(sis)) { + printk(KERN_ERR "sis7019: unable to re-init controller\n"); + goto error; + } + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); + goto error; + } + + /* Restore saved state, then clear out the page we use for the + * silence buffer. + */ + for (i = 0; i < 4; i++) { + memcpy_toio(ioaddr, sis->suspend_state[i], 4096); + ioaddr += 4096; + } + + memset(sis->suspend_state[0], 0, 4096); + + sis->irq = pci->irq; + pci_set_master(pci); + + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[2]); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + +error: + snd_card_disconnect(card); + return -EIO; +} +#endif /* CONFIG_PM */ + +static int sis_alloc_suspend(struct sis7019 *sis) +{ + int i; + + /* We need 16K to store the internal wave engine state during a + * suspend, but we don't need it to be contiguous, so play nice + * with the memory system. We'll also use this area for a silence + * buffer. + */ + for (i = 0; i < SIS_SUSPEND_PAGES; i++) { + sis->suspend_state[i] = kmalloc(4096, GFP_KERNEL); + if (!sis->suspend_state[i]) + return -ENOMEM; + } + memset(sis->suspend_state[0], 0, 4096); + + return 0; +} + +static int __devinit sis_chip_create(struct snd_card *card, + struct pci_dev *pci) +{ + struct sis7019 *sis = card->private_data; + struct voice *voice; + static struct snd_device_ops ops = { + .dev_free = sis_dev_free, + }; + int rc; + int i; + + rc = pci_enable_device(pci); + if (rc) + goto error_out; + + if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0) { + printk(KERN_ERR "sis7019: architecture does not support " + "30-bit PCI busmaster DMA"); + goto error_out_enabled; + } + + memset(sis, 0, sizeof(*sis)); + mutex_init(&sis->ac97_mutex); + spin_lock_init(&sis->voice_lock); + sis->card = card; + sis->pci = pci; + sis->irq = -1; + sis->ioport = pci_resource_start(pci, 0); + + rc = pci_request_regions(pci, "SiS7019"); + if (rc) { + printk(KERN_ERR "sis7019: unable request regions\n"); + goto error_out_enabled; + } + + rc = -EIO; + sis->ioaddr = ioremap_nocache(pci_resource_start(pci, 1), 0x4000); + if (!sis->ioaddr) { + printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + goto error_out_cleanup; + } + + rc = sis_alloc_suspend(sis); + if (rc < 0) { + printk(KERN_ERR "sis7019: unable to allocate state storage\n"); + goto error_out_cleanup; + } + + rc = sis_chip_init(sis); + if (rc) + goto error_out_cleanup; + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + goto error_out_cleanup; + } + + sis->irq = pci->irq; + pci_set_master(pci); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + voice->num = i; + voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i); + voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i); + } + + voice = &sis->capture_voice; + voice->flags = VOICE_CAPTURE; + voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN; + voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num); + + rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops); + if (rc) + goto error_out_cleanup; + + snd_card_set_dev(card, &pci->dev); + + return 0; + +error_out_cleanup: + sis_chip_free(sis); + +error_out_enabled: + pci_disable_device(pci); + +error_out: + return rc; +} + +static int __devinit snd_sis7019_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct sis7019 *sis; + int rc; + + rc = -ENOENT; + if (!enable) + goto error_out; + + rc = -ENOMEM; + card = snd_card_new(index, id, THIS_MODULE, sizeof(*sis)); + if (!card) + goto error_out; + + strcpy(card->driver, "SiS7019"); + strcpy(card->shortname, "SiS7019"); + rc = sis_chip_create(card, pci); + if (rc) + goto card_error_out; + + sis = card->private_data; + + rc = sis_mixer_create(sis); + if (rc) + goto card_error_out; + + rc = sis_pcm_create(sis); + if (rc) + goto card_error_out; + + snprintf(card->longname, sizeof(card->longname), + "%s Audio Accelerator with %s at 0x%lx, irq %d", + card->shortname, snd_ac97_get_short_name(sis->ac97[0]), + sis->ioport, sis->irq); + + rc = snd_card_register(card); + if (rc) + goto card_error_out; + + pci_set_drvdata(pci, card); + return 0; + +card_error_out: + snd_card_free(card); + +error_out: + return rc; +} + +static void __devexit snd_sis7019_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver sis7019_driver = { + .name = "SiS7019", + .id_table = snd_sis7019_ids, + .probe = snd_sis7019_probe, + .remove = __devexit_p(snd_sis7019_remove), + +#ifdef CONFIG_PM + .suspend = sis_suspend, + .resume = sis_resume, +#endif +}; + +static int __init sis7019_init(void) +{ + return pci_register_driver(&sis7019_driver); +} + +static void __exit sis7019_exit(void) +{ + pci_unregister_driver(&sis7019_driver); +} + +module_init(sis7019_init); +module_exit(sis7019_exit); diff -r eb09731e73d7 pci/sis7019.h --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.h Thu Dec 13 21:28:56 2007 -0500 @@ -0,0 +1,342 @@ +#ifndef __sis7019_h__ +#define __sis7019_h__ + +/* + * Definitions for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * 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. + * + * 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 + */ + + +/* General Control Register */ +#define SIS_GCR 0x00 +#define SIS_GCR_MACRO_POWER_DOWN 0x80000000 +#define SIS_GCR_MODEM_ENABLE 0x00010000 +#define SIS_GCR_SOFTWARE_RESET 0x00000001 + +/* General Interrupt Enable Register */ +#define SIS_GIER 0x04 +#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000 +#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000 +#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000 +#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000 +#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000 +#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010 +#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008 +#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004 +#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002 +#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001 + +/* General Interrupt Status Register */ +#define SIS_GISR 0x08 +#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000 +#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000 +#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000 +#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000 +#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000 +#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010 +#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008 +#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004 +#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002 +#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001 + +/* DMA Control Register */ +#define SIS_DMA_CSR 0x10 +#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d +#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200 +#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100 +#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010 +#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008 +#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004 +#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002 +#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001 + +/* Playback Channel Start Registers */ +#define SIS_PLAY_START_A_REG 0x14 +#define SIS_PLAY_START_B_REG 0x18 + +/* Playback Channel Stop Registers */ +#define SIS_PLAY_STOP_A_REG 0x1c +#define SIS_PLAY_STOP_B_REG 0x20 + +/* Recording Channel Start Register */ +#define SIS_RECORD_START_REG 0x24 + +/* Recording Channel Stop Register */ +#define SIS_RECORD_STOP_REG 0x28 + +/* Playback Interrupt Status Registers */ +#define SIS_PISR_A 0x2c +#define SIS_PISR_B 0x30 + +/* Recording Interrupt Status Register */ +#define SIS_RISR 0x34 + +/* AC97 AC-link Playback Source Register */ +#define SIS_AC97_PSR 0x40 +#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000 +#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000 +#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000 +#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000 +#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00 +#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0 +#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f + +/* AC97 AC-link Command Register */ +#define SIS_AC97_CMD 0x50 +#define SIS_AC97_CMD_DATA_MASK 0xffff0000 +#define SIS_AC97_CMD_REG_MASK 0x0000ff00 +#define SIS_AC97_CMD_CODEC3_READ 0x0000000d +#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c +#define SIS_AC97_CMD_CODEC2_READ 0x0000000b +#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a +#define SIS_AC97_CMD_CODEC_READ 0x00000009 +#define SIS_AC97_CMD_CODEC_WRITE 0x00000008 +#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005 +#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004 +#define SIS_AC97_CMD_DONE 0x00000000 + +/* AC97 AC-link Semaphore Register */ +#define SIS_AC97_SEMA 0x54 +#define SIS_AC97_SEMA_BUSY 0x00000001 +#define SIS_AC97_SEMA_RELEASE 0x00000000 + +/* AC97 AC-link Status Register */ +#define SIS_AC97_STATUS 0x58 +#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000 +#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000 +#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000 +#define SIS_AC97_STATUS_CODEC3_READY 0x00000400 +#define SIS_AC97_STATUS_CODEC2_READY 0x00000200 +#define SIS_AC97_STATUS_CODEC_READY 0x00000100 +#define SIS_AC97_STATUS_WARM_RESET 0x00000080 +#define SIS_AC97_STATUS_COLD_RESET 0x00000040 +#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020 +#define SIS_AC97_STATUS_NORMAL 0x00000010 +#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004 +#define SIS_AC97_STATUS_SEMAPHORE 0x00000002 +#define SIS_AC97_STATUS_BUSY 0x00000001 + +/* AC97 AC-link Audio Configuration Register */ +#define SIS_AC97_CONF 0x5c +#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000 +#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000 +#define SIS_AC97_CONF_PR6_ENABLE 0x20000000 +#define SIS_AC97_CONF_PR5_ENABLE 0x10000000 +#define SIS_AC97_CONF_PR4_ENABLE 0x08000000 +#define SIS_AC97_CONF_PR3_ENABLE 0x04000000 +#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000 +#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000 +#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000 +#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000 +#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000 +#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000 +#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000 +#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000 +#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000 +#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200 +#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100 +#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080 +#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040 +#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020 +#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010 +#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008 +#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004 +#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002 +#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001 + +/* Playback Channel Sync Group registers */ +#define SIS_PLAY_SYNC_GROUP_A 0x80 +#define SIS_PLAY_SYNC_GROUP_B 0x84 +#define SIS_PLAY_SYNC_GROUP_C 0x88 +#define SIS_PLAY_SYNC_GROUP_D 0x8c +#define SIS_MIXER_SYNC_GROUP 0x90 + +/* Wave Engine Config and Control Register */ +#define SIS_WECCR 0xa0 +#define SIS_WECCR_TESTMODE_MASK 0x00300000 +#define SIS_WECCR_TESTMODE_NORMAL 0x00000000 +#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000 +#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000 +#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000 +#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000 +#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000 +#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000 +#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00 +#define SIS_WECCR_COMMAND_MASK 0x00000007 +#define SIS_WECCR_COMMAND_NONE 0x00000000 +#define SIS_WECCR_COMMAND_DONE 0x00000000 +#define SIS_WECCR_COMMAND_PAUSE 0x00000001 +#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002 +#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003 +#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004 + +/* Wave Engine Volume Control Register */ +#define SIS_WEVCR 0xa4 +#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000 +#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000 +#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00 +#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff + +/* Wave Engine Interrupt Status Registers */ +#define SIS_WEISR_A 0xa8 +#define SIS_WEISR_B 0xac + + +/* Playback DMA parameters (paramter RAM) */ +#define SIS_PLAY_DMA_OFFSET 0x0000 +#define SIS_PLAY_DMA_SIZE 0x10 +#define SIS_PLAY_DMA_ADDR(addr, num) \ + ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET) + +#define SIS_PLAY_DMA_FORMAT_CSO 0x00 +#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000 +#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000 +#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff +#define SIS_PLAY_DMA_BASE 0x04 +#define SIS_PLAY_DMA_CONTROL 0x08 +#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000 +#define SIS_PLAY_DMA_RELEASE 0x02000000 +#define SIS_PLAY_DMA_LOOP 0x01000000 +#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000 +#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000 +#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000 +#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000 +#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff +#define SIS_PLAY_DMA_SSO_ESO 0x0c +#define SIS_PLAY_DMA_SSO_MASK 0xffff0000 +#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff + +/* Capture DMA parameters (paramter RAM) */ +#define SIS_CAPTURE_DMA_OFFSET 0x0800 +#define SIS_CAPTURE_DMA_SIZE 0x10 +#define SIS_CAPTURE_DMA_ADDR(addr, num) \ + ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET) + +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15 +#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16 +#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17 +#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18 +#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19 +#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20 + +#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00 +#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000 +#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000 +#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000 +#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000 +#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000 +#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000 +#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_BASE 0x04 +#define SIS_CAPTURE_DMA_CONTROL 0x08 +#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000 +#define SIS_CAPTURE_DMA_RELEASE 0x02000000 +#define SIS_CAPTURE_DMA_LOOP 0x01000000 +#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000 +#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000 +#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_RESERVED 0x0c + + +/* Mixer routing list start pointer (parameter RAM) */ +#define SIS_MIXER_START_OFFSET 0x1000 +#define SIS_MIXER_START_SIZE 0x04 +#define SIS_MIXER_START_ADDR(addr, num) \ + ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET) + +#define SIS_MIXER_START_MASK 0x0000007f + +/* Mixer routing table (parameter RAM) */ +#define SIS_MIXER_OFFSET 0x1400 +#define SIS_MIXER_SIZE 0x04 +#define SIS_MIXER_ADDR(addr, num) \ + ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET) + +#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000 +#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0x00ff0000 +#define SIS_MIXER_LEFT_NO_ATTEN 0x00ff0000 +#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00 +#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000 +#define SIS_MIXER_DEST_MASK 0x0000007f +#define SIS_MIXER_DEST_0 0x00000020 +#define SIS_MIXER_DEST_1 0x00000021 +#define SIS_MIXER_DEST_2 0x00000022 +#define SIS_MIXER_DEST_3 0x00000023 +#define SIS_MIXER_DEST_4 0x00000024 +#define SIS_MIXER_DEST_5 0x00000025 +#define SIS_MIXER_DEST_6 0x00000026 +#define SIS_MIXER_DEST_7 0x00000027 +#define SIS_MIXER_DEST_8 0x00000028 +#define SIS_MIXER_DEST_9 0x00000029 +#define SIS_MIXER_DEST_10 0x0000002a +#define SIS_MIXER_DEST_11 0x0000002b +#define SIS_MIXER_DEST_12 0x0000002c +#define SIS_MIXER_DEST_13 0x0000002d +#define SIS_MIXER_DEST_14 0x0000002e +#define SIS_MIXER_DEST_15 0x0000002f + +/* Wave Engine Control Parameters (parameter RAM) */ +#define SIS_WAVE_OFFSET 0x2000 +#define SIS_WAVE_SIZE 0x40 +#define SIS_WAVE_ADDR(addr, num) \ + ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET) + +#define SIS_WAVE_GENERAL 0x00 +#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000 +#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000 +#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000 +#define SIS_WAVE_GENERAL_ARTICULATION 0x04 +#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000 +#define SIS_WAVE_ARTICULATION 0x08 +#define SIS_WAVE_TIMER 0x0c +#define SIS_WAVE_GENERATOR 0x10 +#define SIS_WAVE_CHANNEL_CONTROL 0x14 +#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000 +#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000 +#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000 +#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000 +#define SIS_WAVE_LFO_EG_CONTROL 0x18 +#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c +#define SIS_WAVE_LFO_EG_CONTROL_3 0x20 +#define SIS_WAVE_LFO_EG_CONTROL_4 0x24 + +#endif /* __sis7019_h__ */

2 2

Re: [alsa-devel] [Pkg-alsa-devel] Bug#438118: alsa-utils: aplay non-blocking mode isn't working
by Elimar Riesebieter 14 Dec '07

14 Dec '07

forwarded 438118 alsa-devel(a)alsa-project.org stop Hi all, can one please have a lokk at this? aplay -N isn't working for the OP. On Thu, 16 Aug 2007 the mental interface of Anders Boström told: > >>>>> "ER" == Elimar Riesebieter <riesebie(a)lxtec.de> writes: > > >> strace aplay -N : > >> > >> ... > >> open("/usr/lib/alsa-lib/libasound_module_rate_speexrate.so", O_RDONLY) = -1 ENOENT (No such file or directory) > > ER> This file is provided by > ER> libasound2-plugins: /usr/lib/alsa-lib/libasound_module_rate_speexrate.so > > ER> Please install libasound2-plugins and try again. > > Thanks for the suggestion. > > Unfortunately, it didn't solve my problem, no sound when non-blocking > mode is used. New strace: > > strace aplay -N : > ... > open("/usr/lib/alsa-lib/libasound_module_rate_speexrate.so", O_RDONLY) = 5 > read(5, "\177ELF\2\1\1\0\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0\20\22\0"..., 832) = 832 > fstat(5, {st_mode=S_IFREG|0644, st_size=20848, ...}) = 0 > mmap(NULL, 2116136, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 5, 0) = 0x2b9f792fe000 > mprotect(0x2b9f79302000, 2097152, PROT_NONE) = 0 > mmap(0x2b9f79502000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 5, 0x4000) = 0x2b9f79502000 > close(5) = 0 > ioctl(4, 0xc2604110, 0x7fff3269dec0) = 0 > ioctl(4, 0xc2604110, 0x7fff3269dec0) = 0 > ioctl(4, 0xc2604110, 0x7fff3269de00) = 0 > ioctl(4, 0xc2604110, 0x7fff3269de00) = 0 > ioctl(4, 0xc2604110, 0x7fff3269ddd0) = 0 > ioctl(4, 0xc2604110, 0x7fff3269ddc0) = 0 > ioctl(4, 0xc2604111, 0x7fff3269ddc0) = 0 > ioctl(4, 0xc0884113, 0x7fff3269dd10) = 0 > ioctl(4, 0x80184132, 0x7fff3269dc70) = 0 > mmap(NULL, 49152, PROT_READ|PROT_WRITE, MAP_SHARED, 4, 0) = 0x2b9f79503000 > ioctl(4, 0xc0884113, 0x624e88) = 0 > ioctl(4, 0x4140, 0) = 0 > ioctl(4, 0xc0884113, 0x624e88) = 0 > read(3, "\377eO\333\337\370\344}\335_\377\330^l\356\361`j\336_l"..., 1001) = 1001 > read(3, "n{u^\342zl~e~_[\365_y\362_\363\350\337n[\347nmi`\367y\344"..., 604) = 604 > ioctl(4, 0x4144, 0x2b9f785d8000) = -1 EAGAIN (Resource temporarily unavailable) > close(3) = 0 > ioctl(4, 0x4143, 0x3) = 0 > munmap(0x2b9f79503000, 49152) = 0 > ioctl(4, 0x4112, 0) = 0 > close(4) = 0 > munmap(0x2b9f785d7000, 4096) = 0 > munmap(0x2b9f785d8000, 4096) = 0 > munmap(0x2b9f792fe000, 2116136) = 0 > exit_group(0) = ? I have no idea, so forwarded to alsa-dev then. THX in advance Elimar -- Experience is something you don't get until just after you need it!

5 8

[alsa-devel] [PATCH] hda: Added more 92HD71 codecs
by Matthew Ranostay 14 Dec '07

14 Dec '07

Added more codecs to the 92HD71 family, as well as support for several that don't have an analog mixer. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 51415b2a4cbb pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Dec 13 17:51:00 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Dec 13 14:10:59 2007 -0500 @@ -532,9 +532,24 @@ 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}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* 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)}, + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* unmute mono out node */ + { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, +}; + +static struct hda_verb stac92hd71bxx_analog_core_init[] = { + /* set master volume and direct control */ + { 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}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ @@ -714,9 +729,31 @@ static struct snd_kcontrol_new stac92hd7 { } /* end */ }; +static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), + STAC_INPUT_SOURCE(2), + + /* hardware gain controls */ + 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 */ +}; + static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), /* hardware gain controls */ HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), @@ -729,9 +766,6 @@ static struct snd_kcontrol_new stac92hd7 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 */ }; @@ -3088,11 +3122,24 @@ again: stac92xx_set_config_regs(codec); } + switch (codec->vendor_id) { + case 0x111d76b6: /* 4 Port without Analog Mixer */ + case 0x111d76b7: + case 0x111d76b4: /* 6 Port without Analog Mixer */ + case 0x111d76b5: + spec->mixer = stac92hd71bxx_mixer; + spec->init = stac92hd71bxx_core_init; + break; + default: + spec->mixer = stac92hd71bxx_analog_mixer; + spec->init = stac92hd71bxx_analog_core_init; + } + + spec->aloopback_mask = 0x20; + spec->aloopback_shift = 0; + 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; @@ -3702,8 +3749,17 @@ 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 = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, + { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7675, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d76b0, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, + { .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b3, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b4, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, {} /* terminator */ };

2 1

[alsa-devel] [BUG] Kernel Bugzilla: ALSA bugs
by Natalie Protasevich 14 Dec '07

14 Dec '07

Hello, There are couple of open ALSA bus on bugzilla.kernel.org, seems like they weren't noticed yet. Can someone take look at those please. Thanks! hp_only quirk needed for lifebook p7010 http://bugzilla.kernel.org/show_bug.cgi?id=9100 Kernel: 2.6.22.X on lifebook p7010 I need to add options snd-intel8x0 ac97_quirk=hp_only to have the master volume work properly snd-au8830 has strange I/O problems http://bugzilla.kernel.org/show_bug.cgi?id=9367 Kernel:2.6.22 jack only works in 'playback' mode, none 'duplex' or 'capture' works. snd_hda_intel doesn't mute front speakers when headset is plugged in http://bugzilla.kernel.org/show_bug.cgi?id=9338 Kernel: 2.6.23+ Vaio VGN-FZ19VN

1 0

[alsa-devel] [PATCH v3 2/2] sis7019: add support to driver package
by Dave Dillow 14 Dec '07

14 Dec '07

The sis7019 driver uses __ffs(), which first became available in kernel 2.5.2.6. This adds that compatibility function for x86 to adriver.h, and adds a more useful error message for other platforms that try to use it. Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-driver hg repo. It builds against 2.6.23, but I've not tested against earlier versions. pci/sis7019.c | 2 ++ include/adriver.h | 18 ++++++++++++++++++ 2 files changed, 20 insertions(+) diff -r 9eda190b55c3 include/adriver.h --- a/include/adriver.h Tue Dec 04 12:46:46 2007 +0100 +++ b/include/adriver.h Thu Dec 13 22:50:16 2007 -0500 @@ -1445,4 +1445,22 @@ static inline unsigned char snd_pci_revi #endif #endif +#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 3) +#if defined(__i386__) +static inline unsigned long __ffs(unsigned long word) +{ + __asm__("bsfl %1,%0" + :"=r" (word) + :"rm" (word)); + return word; +} +#else +static inline unsigned long __ffs(unsigned long word) +{ + __asm__("need_asm_for_your_arch_in_adriver.h"); + return word; +} +#endif +#endif + #endif /* __SOUND_LOCAL_DRIVER_H */ diff -r 9eda190b55c3 pci/sis7019.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.c Thu Dec 13 22:50:16 2007 -0500 @@ -0,0 +1,2 @@ +#include "../alsa-kernel/pci/sis7019.c" +EXPORT_NO_SYMBOLS;

1 0

[alsa-devel] Alsa library for the arm-linux architecture
by Felipe Uderman 13 Dec '07

13 Dec '07

Hello, I need to cross compile a project to the arm-linux platform that depends on the Alsa Library. My problem is that my cross development environment does not include the Alsa Library. I have downloaded the Alsa sources, and I checked that there are some instructions on how to port the library to other architectures. I would like to know if there are ports of the Alsa Library for the arm-linux architecture availible for download, or if I really have to cross compile it. Thank you for your help! Felipe

3 3

[alsa-devel] Dell-830 Intel Core2 Duo
by Alex Garin 13 Dec '07

13 Dec '07

Hi, I have this error when I installed Ubuntu 7.10 on this computer Dell-830 Intel Core2 Duo : No volume control GStreamer plugins and/or devices found. Find some things on the web, as I installed and compiled alsa-<driver, lib, utils> but still do not works. Any help please ? Thank you ! -- Alex Garin - CGI / GMA Gestion Intégrée des Technologies / Integrated Technology Management 1350 Boulevard Rene-Levesque Ouest, Montréal, Québec, H3G 1T4 (514) 415-3000 ext. 5194, (514) 415-3962 fax

1 0

[alsa-devel] [PATCH] hda: 92HD73 DMIC Amps
by Matthew Ranostay 13 Dec '07

13 Dec '07

Changed hardware gain mixers for the digital mic's from HDA_OUTPUT to HDA_INPUT. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 51415b2a4cbb pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Dec 13 17:51:00 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Dec 13 13:25:46 2007 -0500 @@ -626,8 +626,8 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -657,8 +657,8 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -688,8 +688,8 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),

1 0

[alsa-devel] [PATCH 1/2] sis7019: support the SiS 7019 Audio Accelerator
by Dave Dillow 13 Dec '07

13 Dec '07

Basic audio support for the SiS 7019 Audio Accelerator as found in the SiS 55x SoC. There is currently no synth support at the moment, but audio playback and capture with two periods per buffer has seen extensive use. Arbitrary period and buffer sizes (with multiple periods per buffer) have seen light testing, but are believed to be production ready. Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-kernel hg repo. Changes since last posting: * Cleanups as suggested * Fix a few mistaken hardware limits (off-by-one) pci/sis7019.c | 1462 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ pci/sis7019.h | 342 +++++++++++++ pci/Kconfig | 10 pci/Makefile | 2 4 files changed, 1816 insertions(+) diff -r eb09731e73d7 pci/Kconfig --- a/pci/Kconfig Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/Kconfig Sat Dec 08 18:46:18 2007 -0500 @@ -802,6 +802,16 @@ config SND_RME9652 To compile this driver as a module, choose M here: the module will be called snd-rme9652. +config SND_SIS7019 + tristate "SiS 7019 Audio Accelerator" + depends on SND && X86 && !X86_64 + select SND_AC97_CODEC + help + Say Y here to include support for the SiS 7019 Audio Accelerator. + + To compile this driver as a module, choose M here: the module + will be called snd-sis7019. + config SND_SONICVIBES tristate "S3 SonicVibes" depends on SND diff -r eb09731e73d7 pci/Makefile --- a/pci/Makefile Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/Makefile Sat Dec 08 18:46:18 2007 -0500 @@ -23,6 +23,7 @@ snd-maestro3-objs := maestro3.o snd-maestro3-objs := maestro3.o snd-rme32-objs := rme32.o snd-rme96-objs := rme96.o +snd-sis7019-objs := sis7019.o snd-sonicvibes-objs := sonicvibes.o snd-via82xx-objs := via82xx.o snd-via82xx-modem-objs := via82xx_modem.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_MAESTRO3) += snd-maestr obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o obj-$(CONFIG_SND_RME32) += snd-rme32.o obj-$(CONFIG_SND_RME96) += snd-rme96.o +obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o diff -r eb09731e73d7 pci/sis7019.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.c Sat Dec 08 18:46:18 2007 -0500 @@ -0,0 +1,1462 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * 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. + * + * 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 + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/time.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include "sis7019.h" + +MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); +MODULE_DESCRIPTION("SiS 7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = 1; + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is one or two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the SSO on the playback + * channel to clock out virtual periods, and adjust the virtual period length + * to maintain synchronization. This algorithm came from the Trident driver. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u16 sync_period_size; + u16 sync_buffer_size; + u32 sso; + u32 vperiod; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +/* We need four pages to store our wave parameters during a suspend. If + * we're not doing power management, we still need to allocate a page + * for the silence buffer. + */ +#ifdef CONFIG_PM +#define SIS_SUSPEND_PAGES 4 +#else +#define SIS_SUSPEND_PAGES 1 +#endif + +struct sis7019 { + unsigned long ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* Protect against more than one thread hitting the AC97 + * registers (in a more polite manner than pounding the hardware + * semaphore) + */ + struct mutex ac97_mutex; + + /* voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; + + /* Allocate pages to store the internal wave state during + * suspends. When we're operating, this can be used as a silence + * buffer for a timing channel. + */ + void *suspend_state[SIS_SUSPEND_PAGES]; + + int silence_users; + dma_addr_t silence_dma_addr; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +/* The HW offset parameters (Loop End, Stop Sample, End Sample) have a + * documented range of 8-0xfff8 samples. Given that they are 0-based, + * that places our period/buffer range at 9-0xfff9 samples. That makes the + * max buffer size 0xfff9 samples * 2 channels * 2 bytes per sample, and + * max samples / min samples gives us the max periods in a buffer. + * + * We'll add a constraint upon open that limits the period and buffer sample + * size to values that are legal for the hardware. + */ +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static void sis_update_sso(struct voice *voice, u16 period) +{ + void __iomem *base = voice->ctrl_base; + + voice->sso += period; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* Enforce the documented hardware minimum offset */ + if (voice->sso < 8) + voice->sso = 8; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); +} + +static void sis_update_voice(struct voice *voice) +{ + if (voice->flags & VOICE_SSO_TIMING) { + sis_update_sso(voice, voice->period_size); + } else if (voice->flags & VOICE_SYNC_TIMING) { + int sync; + + /* If we've not hit the end of the virtual period, update + * our records and keep going. + */ + if (voice->vperiod > voice->period_size) { + voice->vperiod -= voice->period_size; + if (voice->vperiod < voice->period_size) + sis_update_sso(voice, voice->vperiod); + else + sis_update_sso(voice, voice->period_size); + return; + } + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = voice->sync_cso; + sync -= readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + if (sync > (voice->sync_buffer_size / 2)) + sync -= voice->sync_buffer_size; + + /* If sync is positive, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + */ + if (sync > 0) { + if (sync < 16) + sync = 16; + sis_update_sso(voice, sync); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * a minimum wait time, which makes us quite late here, + * eating into the user's time to refresh the buffer, esp. + * if using small periods. + * + * If we're less than 9 samples behind, we're on target. + */ + if (sync > -9) + voice->vperiod = voice->sync_period_size + 1; + else + voice->vperiod = voice->sync_period_size - 4; + + if (voice->vperiod < voice->buffer_size) { + sis_update_sso(voice, voice->vperiod); + voice->vperiod = 0; + } else + sis_update_sso(voice, voice->period_size); + + sync = voice->sync_cso + voice->sync_period_size; + if (sync >= voice->sync_buffer_size) + sync -= voice->sync_buffer_size; + voice->sync_cso = sync; + } + + snd_pcm_period_elapsed(voice->substream); +} + +static void sis_voice_irq(u32 status, struct voice *voice) +{ + int bit; + + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); + voice++; + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + do { + status = inl(io + SIS_PISR_A); + if (status) { + sis_voice_irq(status, sis->voices); + outl(status, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + sis_voice_irq(status, &sis->voices[32]); + outl(status, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void __sis_map_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + if (!sis->silence_users) + sis->silence_dma_addr = pci_map_single(sis->pci, + sis->suspend_state[0], + 4096, PCI_DMA_TODEVICE); + sis->silence_users++; +} + +static void __sis_unmap_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + sis->silence_users--; + if (!sis->silence_users) + pci_unmap_single(sis->pci, sis->silence_dma_addr, 4096, + PCI_DMA_TODEVICE); +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + __sis_unmap_silence(sis); + voice->timing->flags = ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *__sis_alloc_playback_voice(struct sis7019 *sis) +{ + /* Must hold the voice_lock on entry */ + struct voice *voice; + int i; + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + return voice; +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + voice = __sis_alloc_playback_voice(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + unsigned int period_size, buffer_size; + unsigned long flags; + int needed; + + /* If there are one or two periods per buffer, we don't need a + * timing voice, as we can use the capture channel's interrupts + * to clock out the periods. + */ + period_size = params_period_size(hw_params); + buffer_size = params_buffer_size(hw_params); + needed = (period_size != buffer_size && + period_size != (buffer_size / 2)); + + if (needed && !voice->timing) { + spin_lock_irqsave(&sis->voice_lock, flags); + voice->timing = __sis_alloc_playback_voice(sis); + if (voice->timing) + __sis_map_silence(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + /* The baseline setup is for a single period per buffer, and + * we add bells and whistles as needed from there. + */ + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + if (runtime->period_size == (runtime->buffer_size / 2)) { + control |= SIS_PLAY_DMA_INTR_AT_MLP; + } else if (runtime->period_size != runtime->buffer_size) { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size - 1; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control &= ~SIS_PLAY_DMA_INTR_AT_LEO; + control |= SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso |= (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + /* Ok, we're ready to go, set up the channel. + */ + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + /* No locks needed, as the PCM core will hold the locks on the + * substreams, and the HW will only start/stop the indicated voices + * without changing the state of the others. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static void sis_prepare_timing_voice(struct voice *timing, struct voice *cap, + struct snd_pcm_runtime *runtime) +{ + u16 buffer_size, period_size; + u32 vperiod, sso; + + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_base = cap->ctrl_base; + timing->sync_cso = runtime->period_size - 1; + timing->sync_period_size = runtime->period_size; + timing->sync_buffer_size = runtime->buffer_size; + + /* Set our initial buffer and period as large as we can given a + * single page of silence. + */ + buffer_size = 4096 / runtime->channels; + buffer_size /= snd_pcm_format_size(runtime->format, 1); + period_size = buffer_size; + + /* Initially, we want to interrupt just a bit behind the end of + * the period we're clocking out. 10 samples seems to give a good + * delay. + * + * We want to spread our interrupts throughout the virtual period, + * so that we don't end up with two interrupts back to back at the + * end -- this helps minimize the effects of any jitter. Adjust our + * clocking period size so that the last period is at least a fourth + * of a full period. + * + * This is all moot if we don't need to use virtual periods. + */ + vperiod = runtime->period_size + 10; + if (vperiod > period_size) { + u16 tail = vperiod % period_size; + u16 quarter_period = period_size / 4; + + if (tail && tail < quarter_period) { + u16 loops = vperiod / period_size; + + tail = quarter_period - tail; + tail += loops - 1; + tail /= loops; + period_size -= tail; + } + + sso = period_size - 1; + } else { + /* The initial period will fit inside the buffer, so we + * don't need to use virtual periods -- disable them. + */ + period_size = runtime->period_size; + sso = vperiod - 1; + vperiod = 0; + } + + timing->period_size = period_size; + timing->buffer_size = buffer_size; + timing->sso = sso; + timing->vperiod = vperiod; +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + struct voice *timing = voice->timing; + void __iomem *rec_base = voice->ctrl_base; + void __iomem *play_base = NULL; + void __iomem *wave_base = NULL; + u32 format, dma_addr, rec_ctrl; + u16 leo; + + /* The following variables are only used if there is a timing channel. + */ + u32 uninitialized_var(timing_ctrl); + u32 uninitialized_var(sso_eso); + u32 uninitialized_var(delta); + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + rec_ctrl = leo | SIS_CAPTURE_DMA_LOOP; + if (timing) { + /* We're using a timing voice, so the capture voice will + * not need to generate interrupts. + */ + sis_prepare_timing_voice(timing, voice, runtime); + + timing_ctrl = timing->buffer_size - 1; + timing_ctrl |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso = timing->buffer_size - 1;; + sso_eso |= timing->sso << 16; + + delta = sis_rate_to_delta(runtime->rate); + + play_base = timing->ctrl_base; + wave_base = timing->wave_base; + } else { + /* We've just got one or two periods, so we can use the + * capture voice's interrupt generation. + */ + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_LEO; + if (runtime->period_size != runtime->buffer_size) + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_MLP; + } + + /* We're ready to program the channel(s) + */ + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(rec_ctrl, rec_base + SIS_CAPTURE_DMA_CONTROL); + + if (play_base) { + u32 reg; + + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(sis->silence_dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(timing_ctrl, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, + wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + } + + /* Force the writes to post. */ + readl(rec_base); + + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + /* We have 64 voices, and the driver currently records from + * only one channel, though that could change in the future. + */ + rc = snd_pcm_new(sis->card, "SiS 7019", 0, 64, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS 7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + + /* Get the AC97 semaphore -- software first, so we don't spin + * pounding out IO reads on the hardware semaphore... + */ + mutex_lock(&sis->ac97_mutex); + + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + mutex_unlock(&sis->ac97_mutex); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static void sis_free_suspend(struct sis7019 *sis) +{ + int i; + + for (i = 0; i < SIS_SUSPEND_PAGES; i++) + kfree(sis->suspend_state[i]); +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + sis_free_suspend(sis); + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + /* All done, let go of the semaphore, and check for errors + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!sis->codecs_present || !count) + return -EIO; + + /* Let the hardware know that the audio driver is alive, + * and enable PCM slots on the AC-link for L/R playback (3 & 4) and + * record channels. We're going to want to use Variable Rate Audio + * for recording, to avoid needlessly resampling from 48kHZ. + */ + outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF); + outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE | + SIS_AC97_CONF_PCM_CAP_MIC_ENABLE | + SIS_AC97_CONF_PCM_CAP_LR_ENABLE | + SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF); + + /* All AC97 PCM slots should be sourced from sub-mixer 0. + */ + outl(0, io + SIS_AC97_PSR); + + /* There is only one valid DMA setup for a PCI environment. + */ + outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); + + /* Reset the syncronization groups for all of the channels + * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. + * we'll need to change how we handle these. Until then, we just + * assign sub-mixer 0 to all playback channels, and avoid any + * attenuation on the audio. + */ + outl(0, io + SIS_PLAY_SYNC_GROUP_A); + outl(0, io + SIS_PLAY_SYNC_GROUP_B); + outl(0, io + SIS_PLAY_SYNC_GROUP_C); + outl(0, io + SIS_PLAY_SYNC_GROUP_D); + outl(0, io + SIS_MIXER_SYNC_GROUP); + + for (i = 0; i < 64; i++) { + writel(i, SIS_MIXER_START_ADDR(ioaddr, i)); + writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN | + SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i)); + } + + /* Don't attenuate any audio set for the wave amplifier. + * + * FIXME: Maximum attenuation is set for the music amp, which will + * need to change if we start using the synth engine. + */ + outl(0xffff0000, io + SIS_WEVCR); + + /* Ensure that the wave engine is in normal operating mode. + */ + outl(0, io + SIS_WECCR); + + /* Go ahead and enable the DMA interrupts. They won't go live + * until we start a channel. + */ + outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE | + SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER); + + return 0; +} + +#ifdef CONFIG_PM +static int sis_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(sis->pcm); + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[2]); + + /* snd_pcm_suspend_all() stopped all channels, so we're quiescent. + */ + if (sis->irq >= 0) { + synchronize_irq(sis->irq); + free_irq(sis->irq, sis); + sis->irq = -1; + } + + /* Save the internal state away + */ + for (i = 0; i < 4; i++ ) { + memcpy_fromio(sis->suspend_state[i], ioaddr, 4096); + ioaddr += 4096; + } + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int sis_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "sis7019: unable to re-enable device\n"); + goto error; + } + + if (sis_chip_init(sis)) { + printk(KERN_ERR "sis7019: unable to re-init controller\n"); + goto error; + } + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); + goto error; + } + + /* Restore saved state, then clear out the page we use for the + * silence buffer. + */ + for (i = 0; i < 4; i++ ) { + memcpy_toio(ioaddr, sis->suspend_state[i], 4096); + ioaddr += 4096; + } + + memset(sis->suspend_state[0], 0, 4096); + + sis->irq = pci->irq; + pci_set_master(pci); + + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[2]); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + +error: + snd_card_disconnect(card); + return -EIO; +} +#endif /* CONFIG_PM */ + +static int sis_alloc_suspend(struct sis7019 *sis) +{ + int i; + + /* We need 16K to store the internal wave engine state during a + * suspend, but we don't need it to be contiguous, so play nice + * with the memory system. We'll also use this area for a silence + * buffer. + */ + for (i = 0; i < SIS_SUSPEND_PAGES; i++) { + sis->suspend_state[i] = kmalloc(4096, GFP_KERNEL); + if (!sis->suspend_state[i]) + return -ENOMEM; + } + memset(sis->suspend_state[0], 0, 4096); + + return 0; +} + +static int __devinit sis_chip_create(struct snd_card *card, + struct pci_dev *pci) +{ + struct sis7019 *sis = card->private_data; + struct voice *voice; + static struct snd_device_ops ops = { + .dev_free = sis_dev_free, + }; + int rc; + int i; + + rc = pci_enable_device(pci); + if (rc) + goto error_out; + + if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0) { + printk(KERN_ERR "sis7019: architecture does not support " + "30-bit PCI busmaster DMA"); + goto error_out_enabled; + } + + memset(sis, 0, sizeof(*sis)); + mutex_init(&sis->ac97_mutex); + spin_lock_init(&sis->voice_lock); + sis->card = card; + sis->pci = pci; + sis->irq = -1; + sis->ioport = pci_resource_start(pci, 0); + + rc = pci_request_regions(pci, "SiS 7019"); + if (rc) { + printk(KERN_ERR "sis7019: unable request regions\n"); + goto error_out_enabled; + } + + rc = -EIO; + sis->ioaddr = ioremap_nocache(pci_resource_start(pci, 1), 0x4000); + if (!sis->ioaddr) { + printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + goto error_out_cleanup; + } + + rc = sis_alloc_suspend(sis); + if (rc < 0) { + printk(KERN_ERR "sis7019: unable to allocate state storage\n"); + goto error_out_cleanup; + } + + rc = sis_chip_init(sis); + if (rc) + goto error_out_cleanup; + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + goto error_out_cleanup; + } + + sis->irq = pci->irq; + pci_set_master(pci); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + voice->num = i; + voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i); + voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i); + } + + voice = &sis->capture_voice; + voice->flags = VOICE_CAPTURE; + voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN; + voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num); + + rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops); + if (rc) + goto error_out_cleanup; + + snd_card_set_dev(card, &pci->dev); + + return 0; + +error_out_cleanup: + sis_chip_free(sis); + +error_out_enabled: + pci_disable_device(pci); + +error_out: + return rc; +} + +static int __devinit snd_sis7019_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct sis7019 *sis; + int rc; + + rc = -ENOENT; + if (!enable) + goto error_out; + + rc = -ENOMEM; + card = snd_card_new(index, id, THIS_MODULE, sizeof(*sis)); + if (!card) + goto error_out; + + strcpy(card->driver, "SiS 7019"); + strcpy(card->shortname, "SiS 7019"); + rc = sis_chip_create(card, pci); + if (rc) + goto card_error_out; + + sis = card->private_data; + + rc = sis_mixer_create(sis); + if (rc) + goto card_error_out; + + rc = sis_pcm_create(sis); + if (rc) + goto card_error_out; + + snprintf(card->longname, sizeof(card->longname), + "%s Audio Accelerator with %s at 0x%lx, irq %d", + card->shortname, snd_ac97_get_short_name(sis->ac97[0]), + sis->ioport, sis->irq); + + rc = snd_card_register(card); + if (rc) + goto card_error_out; + + pci_set_drvdata(pci, card); + return 0; + +card_error_out: + snd_card_free(card); + +error_out: + return rc; +} + +static void __devexit snd_sis7019_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver sis7019_driver = { + .name = "SiS7019", + .id_table = snd_sis7019_ids, + .probe = snd_sis7019_probe, + .remove = __devexit_p(snd_sis7019_remove), + +#ifdef CONFIG_PM + .suspend = sis_suspend, + .resume = sis_resume, +#endif +}; + +static int __init sis7019_init(void) +{ + return pci_register_driver(&sis7019_driver); +} + +static void __exit sis7019_exit(void) +{ + pci_unregister_driver(&sis7019_driver); +} + +module_init(sis7019_init); +module_exit(sis7019_exit); diff -r eb09731e73d7 pci/sis7019.h --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.h Sat Dec 08 18:46:18 2007 -0500 @@ -0,0 +1,342 @@ +#ifndef __sis7019_h__ +#define __sis7019_h__ + +/* + * Definitions for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * 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. + * + * 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 + */ + + +/* General Control Register */ +#define SIS_GCR 0x00 +#define SIS_GCR_MACRO_POWER_DOWN 0x80000000 +#define SIS_GCR_MODEM_ENABLE 0x00010000 +#define SIS_GCR_SOFTWARE_RESET 0x00000001 + +/* General Interrupt Enable Register */ +#define SIS_GIER 0x04 +#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000 +#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000 +#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000 +#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000 +#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000 +#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010 +#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008 +#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004 +#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002 +#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001 + +/* General Interrupt Status Register */ +#define SIS_GISR 0x08 +#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000 +#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000 +#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000 +#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000 +#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000 +#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010 +#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008 +#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004 +#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002 +#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001 + +/* DMA Control Register */ +#define SIS_DMA_CSR 0x10 +#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d +#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200 +#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100 +#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010 +#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008 +#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004 +#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002 +#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001 + +/* Playback Channel Start Registers */ +#define SIS_PLAY_START_A_REG 0x14 +#define SIS_PLAY_START_B_REG 0x18 + +/* Playback Channel Stop Registers */ +#define SIS_PLAY_STOP_A_REG 0x1c +#define SIS_PLAY_STOP_B_REG 0x20 + +/* Recording Channel Start Register */ +#define SIS_RECORD_START_REG 0x24 + +/* Recording Channel Stop Register */ +#define SIS_RECORD_STOP_REG 0x28 + +/* Playback Interrupt Status Registers */ +#define SIS_PISR_A 0x2c +#define SIS_PISR_B 0x30 + +/* Recording Interrupt Status Register */ +#define SIS_RISR 0x34 + +/* AC97 AC-link Playback Source Register */ +#define SIS_AC97_PSR 0x40 +#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000 +#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000 +#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000 +#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000 +#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00 +#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0 +#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f + +/* AC97 AC-link Command Register */ +#define SIS_AC97_CMD 0x50 +#define SIS_AC97_CMD_DATA_MASK 0xffff0000 +#define SIS_AC97_CMD_REG_MASK 0x0000ff00 +#define SIS_AC97_CMD_CODEC3_READ 0x0000000d +#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c +#define SIS_AC97_CMD_CODEC2_READ 0x0000000b +#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a +#define SIS_AC97_CMD_CODEC_READ 0x00000009 +#define SIS_AC97_CMD_CODEC_WRITE 0x00000008 +#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005 +#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004 +#define SIS_AC97_CMD_DONE 0x00000000 + +/* AC97 AC-link Semaphore Register */ +#define SIS_AC97_SEMA 0x54 +#define SIS_AC97_SEMA_BUSY 0x00000001 +#define SIS_AC97_SEMA_RELEASE 0x00000000 + +/* AC97 AC-link Status Register */ +#define SIS_AC97_STATUS 0x58 +#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000 +#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000 +#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000 +#define SIS_AC97_STATUS_CODEC3_READY 0x00000400 +#define SIS_AC97_STATUS_CODEC2_READY 0x00000200 +#define SIS_AC97_STATUS_CODEC_READY 0x00000100 +#define SIS_AC97_STATUS_WARM_RESET 0x00000080 +#define SIS_AC97_STATUS_COLD_RESET 0x00000040 +#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020 +#define SIS_AC97_STATUS_NORMAL 0x00000010 +#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004 +#define SIS_AC97_STATUS_SEMAPHORE 0x00000002 +#define SIS_AC97_STATUS_BUSY 0x00000001 + +/* AC97 AC-link Audio Configuration Register */ +#define SIS_AC97_CONF 0x5c +#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000 +#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000 +#define SIS_AC97_CONF_PR6_ENABLE 0x20000000 +#define SIS_AC97_CONF_PR5_ENABLE 0x10000000 +#define SIS_AC97_CONF_PR4_ENABLE 0x08000000 +#define SIS_AC97_CONF_PR3_ENABLE 0x04000000 +#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000 +#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000 +#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000 +#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000 +#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000 +#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000 +#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000 +#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000 +#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000 +#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200 +#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100 +#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080 +#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040 +#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020 +#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010 +#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008 +#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004 +#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002 +#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001 + +/* Playback Channel Sync Group registers */ +#define SIS_PLAY_SYNC_GROUP_A 0x80 +#define SIS_PLAY_SYNC_GROUP_B 0x84 +#define SIS_PLAY_SYNC_GROUP_C 0x88 +#define SIS_PLAY_SYNC_GROUP_D 0x8c +#define SIS_MIXER_SYNC_GROUP 0x90 + +/* Wave Engine Config and Control Register */ +#define SIS_WECCR 0xa0 +#define SIS_WECCR_TESTMODE_MASK 0x00300000 +#define SIS_WECCR_TESTMODE_NORMAL 0x00000000 +#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000 +#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000 +#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000 +#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000 +#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000 +#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000 +#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00 +#define SIS_WECCR_COMMAND_MASK 0x00000007 +#define SIS_WECCR_COMMAND_NONE 0x00000000 +#define SIS_WECCR_COMMAND_DONE 0x00000000 +#define SIS_WECCR_COMMAND_PAUSE 0x00000001 +#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002 +#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003 +#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004 + +/* Wave Engine Volume Control Register */ +#define SIS_WEVCR 0xa4 +#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000 +#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000 +#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00 +#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff + +/* Wave Engine Interrupt Status Registers */ +#define SIS_WEISR_A 0xa8 +#define SIS_WEISR_B 0xac + + +/* Playback DMA parameters (paramter RAM) */ +#define SIS_PLAY_DMA_OFFSET 0x0000 +#define SIS_PLAY_DMA_SIZE 0x10 +#define SIS_PLAY_DMA_ADDR(addr, num) \ + ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET) + +#define SIS_PLAY_DMA_FORMAT_CSO 0x00 +#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000 +#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000 +#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff +#define SIS_PLAY_DMA_BASE 0x04 +#define SIS_PLAY_DMA_CONTROL 0x08 +#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000 +#define SIS_PLAY_DMA_RELEASE 0x02000000 +#define SIS_PLAY_DMA_LOOP 0x01000000 +#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000 +#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000 +#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000 +#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000 +#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff +#define SIS_PLAY_DMA_SSO_ESO 0x0c +#define SIS_PLAY_DMA_SSO_MASK 0xffff0000 +#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff + +/* Capture DMA parameters (paramter RAM) */ +#define SIS_CAPTURE_DMA_OFFSET 0x0800 +#define SIS_CAPTURE_DMA_SIZE 0x10 +#define SIS_CAPTURE_DMA_ADDR(addr, num) \ + ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET) + +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15 +#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16 +#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17 +#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18 +#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19 +#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20 + +#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00 +#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000 +#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000 +#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000 +#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000 +#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000 +#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000 +#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_BASE 0x04 +#define SIS_CAPTURE_DMA_CONTROL 0x08 +#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000 +#define SIS_CAPTURE_DMA_RELEASE 0x02000000 +#define SIS_CAPTURE_DMA_LOOP 0x01000000 +#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000 +#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000 +#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_RESERVED 0x0c + + +/* Mixer routing list start pointer (parameter RAM) */ +#define SIS_MIXER_START_OFFSET 0x1000 +#define SIS_MIXER_START_SIZE 0x04 +#define SIS_MIXER_START_ADDR(addr, num) \ + ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET) + +#define SIS_MIXER_START_MASK 0x0000007f + +/* Mixer routing table (parameter RAM) */ +#define SIS_MIXER_OFFSET 0x1400 +#define SIS_MIXER_SIZE 0x04 +#define SIS_MIXER_ADDR(addr, num) \ + ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET) + +#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000 +#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0x00ff0000 +#define SIS_MIXER_LEFT_NO_ATTEN 0x00ff0000 +#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00 +#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000 +#define SIS_MIXER_DEST_MASK 0x0000007f +#define SIS_MIXER_DEST_0 0x00000020 +#define SIS_MIXER_DEST_1 0x00000021 +#define SIS_MIXER_DEST_2 0x00000022 +#define SIS_MIXER_DEST_3 0x00000023 +#define SIS_MIXER_DEST_4 0x00000024 +#define SIS_MIXER_DEST_5 0x00000025 +#define SIS_MIXER_DEST_6 0x00000026 +#define SIS_MIXER_DEST_7 0x00000027 +#define SIS_MIXER_DEST_8 0x00000028 +#define SIS_MIXER_DEST_9 0x00000029 +#define SIS_MIXER_DEST_10 0x0000002a +#define SIS_MIXER_DEST_11 0x0000002b +#define SIS_MIXER_DEST_12 0x0000002c +#define SIS_MIXER_DEST_13 0x0000002d +#define SIS_MIXER_DEST_14 0x0000002e +#define SIS_MIXER_DEST_15 0x0000002f + +/* Wave Engine Control Parameters (parameter RAM) */ +#define SIS_WAVE_OFFSET 0x2000 +#define SIS_WAVE_SIZE 0x40 +#define SIS_WAVE_ADDR(addr, num) \ + ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET) + +#define SIS_WAVE_GENERAL 0x00 +#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000 +#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000 +#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000 +#define SIS_WAVE_GENERAL_ARTICULATION 0x04 +#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000 +#define SIS_WAVE_ARTICULATION 0x08 +#define SIS_WAVE_TIMER 0x0c +#define SIS_WAVE_GENERATOR 0x10 +#define SIS_WAVE_CHANNEL_CONTROL 0x14 +#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000 +#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000 +#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000 +#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000 +#define SIS_WAVE_LFO_EG_CONTROL 0x18 +#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c +#define SIS_WAVE_LFO_EG_CONTROL_3 0x20 +#define SIS_WAVE_LFO_EG_CONTROL_4 0x24 + +#endif /* __sis7019_h__ */

2 3

[alsa-devel] [PATCH 2/2] sis7019: add support for pre-2.6 kernels
by Dave Dillow 13 Dec '07

13 Dec '07

The sis7019 driver uses __ffs(), which is available in 2.6 kernels, but not in most earlier ones. This patches the driver to use a local version assembly version, as the hardware is SiS 55x only (i486/Pentium-ish). Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-driver hg repo. It has been compiled and tested under 2.6.23. pci/sis7019.patch | 52 ++++++++++++++++++++++++++++++++++++++++++++++++++++ pci/Makefile | 3 ++- 2 files changed, 54 insertions(+), 1 deletion(-) diff -r 9eda190b55c3 pci/Makefile --- a/pci/Makefile Tue Dec 04 12:46:46 2007 +0100 +++ b/pci/Makefile Sat Dec 08 18:45:46 2007 -0500 @@ -10,7 +10,7 @@ include $(SND_TOPDIR)/Makefile.conf # clean-files := ad1889.c atiixp.c atiixp_modem.c bt87x.c cmipci.c ens1370.c \ - fm801.c intel8x0.c maestro3.c via82xx.c via82xx_modem.c + fm801.c intel8x0.c maestro3.c sis7019.c via82xx.c via82xx_modem.c obj-$(CONFIG_SND) += pdplus/ asihpi/ oxygen/ @@ -31,5 +31,6 @@ fm801.c: fm801.patch $(SND_TOPDIR)/alsa- fm801.c: fm801.patch $(SND_TOPDIR)/alsa-kernel/pci/fm801.c intel8x0.c: intel8x0.patch $(SND_TOPDIR)/alsa-kernel/pci/intel8x0.c maestro3.c: maestro3.patch $(SND_TOPDIR)/alsa-kernel/pci/maestro3.c +sis7019.c: sis7019.patch $(SND_TOPDIR)/alsa-kernel/pci/sis7019.c via82xx.c: via82xx.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx.c via82xx_modem.c: via82xx_modem.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx_modem.c diff -r 9eda190b55c3 pci/sis7019.patch --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.patch Sat Dec 08 18:45:46 2007 -0500 @@ -0,0 +1,52 @@ +--- sis7019.c.orig 2007-12-08 18:32:09.000000000 -0500 ++++ sis7019.c 2007-12-08 18:31:35.000000000 -0500 +@@ -31,7 +31,17 @@ + #include <sound/core.h> + #include <sound/ac97_codec.h> + #include <sound/initval.h> +-#include "sis7019.h" ++#include "../alsa-kernel/pci/sis7019.h" ++ ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2,6,0) ++static inline unsigned long sis_ffs(unsigned long word) ++{ ++ __asm__("bsfl %1,%0" ++ :"=r" (word) ++ :"rm" (word)); ++ return word; ++} ++#endif + + MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); + MODULE_DESCRIPTION("SiS 7019"); +@@ -290,7 +300,11 @@ + int bit; + + while (status) { ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2,6,0) ++ bit = sis_ffs(status); ++#else + bit = __ffs(status); ++#endif + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); +@@ -791,9 +805,9 @@ + + /* The following variables are only used if there is a timing channel. + */ +- u32 uninitialized_var(timing_ctrl); +- u32 uninitialized_var(sso_eso); +- u32 uninitialized_var(delta); ++ u32 timing_ctrl = 0; ++ u32 sso_eso = 0; ++ u32 delta = 0; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. +@@ -1460,3 +1474,5 @@ + + module_init(sis7019_init); + module_exit(sis7019_exit); ++ ++EXPORT_NO_SYMBOLS

3 5

[alsa-devel] ALSA external control plugin question
by Ben C 13 Dec '07

13 Dec '07

Hi, I have a newbie question. I'm trying to understand how the external control plugin fits into the picture. I have an external PCM plugin that does some audio processing before sending it to the slave audio device. And I want to control the behavior of the processing with this control plugin. So before any playback is happening, any setting changes via the control plugin need to be stored somewhere, so when playback happens, it could be used. Also during playback, the control plugin needs to find and talk to the PCM plugin so settings could be changed during playback. Is there some sort of internal mechanism that allows this type of setting storage/retrieval and communication between control & PCM plugins to happen, or do I have to roll my own outside of ALSA? I've tried to find examples, I've found the pulse control and maemo in the alsa-plugin tarball. But they both seem to use some sort of external server/device. What I'm looking for is ways to control an external PCM plugin that simply runs some audio processing algorithm, there's no additional server or device associated with it (well, just the slave, which would be specified in the config). Currently I have the PCM plugin working, but it can only read the settings from the alsa config file. Once playback starts it'll stick with the original settings. I also have the basics of the external control plugin working, but the controls currently don't do anything, since I don't know how to propagate the changes to the PCM plugin. I've been searching on the web but haven't found much. I'm willing to buy books too, but I can't seem to find books on ALSA. I would appreciate any explanation, pointers to documents, or even just pointing me to existing source code that has done this. Thank you for your time and your help. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[alsa-devel] [PATCH] hda-intel: Enable Analog CD Input from internal ATAPI connector on Asus M2N-SLI.
by Johannes Stezenbach 13 Dec '07

13 Dec '07

Hi, the internal CD connector on my Asus M2N-SLI Deluxe board doesn't work without the patch below, /proc/asound/card0/codec#0 shows that the pin is disabled (Pin-ctls: 0x00). I used kernel linux-2.6.24-rc4-190-g94545ba from Linus' git tree. During my experiments I found that loading the snd-hda-intel driver with model=auto also enables the CD connector. It seems that for my board (BIOS version 1102) the BIOS configuration is correct and the "Asus M2N-SLI" quirk shouldn't be necessary? I get the following debug output: ALSA sound/pci/hda/hda_intel.c:727: codec_mask = 0x1 ALSA sound/pci/hda/hda_codec.c:2140: hda_codec: model 'auto' is selected ALSA sound/pci/hda/hda_codec.c:2760: autoconfig: line_outs=4 (0x12/0x16/0x24/0x25/0x0) ALSA sound/pci/hda/hda_codec.c:2764: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:2768: hp_outs=1 (0x11/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:2776: inputs: mic=0x17, fmic=0x14, line=0x15, fline=0x0, cd=0x18, aux=0x0 (which matches the board's actual connections) Anyway, if you think the patch below is correct the please apply. It works fine for me. There is one thing that I don't get however: If I run aplay it creates an additional "PCM" mixer control, which however doesn't have any effect. I guess this is intended for softvol, but I don't need it. How do I stop aplay from creating this control? (If it matters, I use Debian unstable, alsa-lib 1.0.15.) Thanks, Johannes --- Enable Analog CD Input from internal ATAPI connector on Asus M2N-SLI. Signed-off-by: Johannes Stezenbach <js(a)sig21.net> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 196ad3c..8c2aff4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2079,6 +2079,8 @@ static struct hda_verb ad1988_6stack_init_verbs[] = { {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Analog CD Input */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, { } };

2 1

[alsa-devel] [PATCH] hda: Added STAC92HD73 support.
by Matthew Ranostay 13 Dec '07

13 Dec '07

Added support for new STAC92HD73 family of codecs. Additionally added features for multiple analog loopbacks, and multiple dmux mixers. Regression testing for the analog loopback changes for STAC9205 and STAC9274D completed with any issues, as well for the dmux changes. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r eb09731e73d7 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Dec 04 11:28:40 2007 -0500 @@ -62,6 +62,11 @@ enum { }; enum { + STAC_92HD73XX_REF, + STAC_92HD73XX_MODELS +}; + +enum { STAC_92HD71BXX_REF, STAC_92HD71BXX_MODELS }; @@ -118,6 +123,7 @@ struct sigmatel_spec { unsigned int gpio_mute: 1; unsigned int gpio_mask, gpio_data; + unsigned int aloopback_mask; /* playback */ struct hda_multi_out multiout; @@ -130,7 +136,7 @@ struct sigmatel_spec { unsigned int num_muxes; hda_nid_t *dmic_nids; unsigned int num_dmics; - hda_nid_t dmux_nid; + hda_nid_t *dmux_nids; hda_nid_t dig_in_nid; /* pin widgets */ @@ -145,7 +151,7 @@ struct sigmatel_spec { /* capture source */ struct hda_input_mux *dinput_mux; - unsigned int cur_dmux; + unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; @@ -176,12 +182,38 @@ static hda_nid_t stac9200_dac_nids[1] = 0x02, }; +static hda_nid_t stac92hd73xx_adc_nids[2] = { + 0x1a, 0x1b +}; + +#define STAC92HD73XX_NUM_DMICS 2 +static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { + 0x13, 0x14, 0 +}; + +#define STAC92HD73_DAC_COUNT 5 +static hda_nid_t stac92hd73xx_dac_nids[STAC92HD73_DAC_COUNT] = { + 0x15, 0x16, 0x17, 0x18, 0x19, +}; + +static hda_nid_t stac92hd73xx_mux_nids[4] = { + 0x28, 0x29, 0x2a, 0x2b, +}; + +static hda_nid_t stac92hd73xx_dmux_nids[2] = { + 0x20, 0x21, +}; + 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_dmux_nids[1] = { + 0x1c, }; static hda_nid_t stac92hd71bxx_dac_nids[2] = { @@ -226,6 +258,10 @@ static hda_nid_t stac927x_mux_nids[3] = 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_dmux_nids[1] = { + 0x1b, +}; + #define STAC927X_NUM_DMICS 2 static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { 0x13, 0x14, 0 @@ -239,6 +275,10 @@ static hda_nid_t stac9205_mux_nids[2] = 0x19, 0x1a }; +static hda_nid_t stac9205_dmux_nids[1] = { + 0x1d, +}; + #define STAC9205_NUM_DMICS 2 static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { 0x17, 0x18, 0 @@ -257,6 +297,12 @@ 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 stac92hd73xx_pin_nids[12] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x10, 0x11, 0x12, 0x13, + 0x14, 0x22 }; static hda_nid_t stac92hd71bxx_pin_nids[10] = { @@ -289,8 +335,9 @@ static int stac92xx_dmux_enum_get(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.enumerated.item[0] = spec->cur_dmux; + unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + ucontrol->value.enumerated.item[0] = spec->cur_dmux[dmux_idx]; return 0; } @@ -299,9 +346,10 @@ static int stac92xx_dmux_enum_put(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); return snd_hda_input_mux_put(codec, spec->dinput_mux, ucontrol, - spec->dmux_nid, &spec->cur_dmux); + spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]); } static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -337,9 +385,11 @@ static int stac92xx_aloopback_get(struct struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.integer.value[0] = spec->aloopback; + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = spec->aloopback & + (spec->aloopback_mask << idx); return 0; } @@ -348,24 +398,29 @@ static int stac92xx_aloopback_put(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int dac_mode; - unsigned int val; - + unsigned int val, idx_val; + + idx_val = (spec->aloopback_mask & 0xFF) << idx; val = !!ucontrol->value.integer.value[0]; - if (spec->aloopback == val) + if ((spec->aloopback | idx) == val) return 0; - spec->aloopback = val; - + spec->aloopback ^= idx_val; + + /* Only return the bits defined by the shift value of the + * first two bytes of the mask */ dac_mode = snd_hda_codec_read(codec, codec->afg, 0, - kcontrol->private_value & 0xFFFF, 0x0); - - if (spec->aloopback) { + kcontrol->private_value & 0xFFFF, 0x0) >> + ((spec->aloopback_mask & 0xFF00) >> 8); + + if (spec->aloopback & idx_val) { snd_hda_power_up(codec); - dac_mode |= 0x40; + dac_mode |= idx_val; } else { snd_hda_power_down(codec); - dac_mode &= ~0x40; + dac_mode &= ~idx_val; } snd_hda_codec_write_cache(codec, codec->afg, 0, @@ -384,6 +439,86 @@ 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 stac92hd73xx_6ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + +static struct hda_verb stac92hd73xx_8ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* connect hp ports to dac3 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x03}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x03}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, + {} +}; + +static struct hda_verb stac92hd73xx_10ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02 }, + /* dac3 is connected to import3 mux */ + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f}, + /* connect hp ports to dac4 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x04}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x04}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, {} }; @@ -460,11 +595,11 @@ static struct hda_verb stac9205_core_ini .put = stac92xx_mux_enum_put, \ } -#define STAC_ANALOG_LOOPBACK(verb_read,verb_write) \ +#define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Analog Loopback", \ - .count = 1, \ + .count = cnt, \ .info = stac92xx_aloopback_info, \ .get = stac92xx_aloopback_get, \ .put = stac92xx_aloopback_put, \ @@ -481,6 +616,99 @@ static struct snd_kcontrol_new stac9200_ { } /* end */ }; +static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), @@ -513,7 +741,7 @@ static struct snd_kcontrol_new stac9205_ static struct snd_kcontrol_new stac9205_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -543,7 +771,7 @@ static struct snd_kcontrol_new stac927x_ static struct snd_kcontrol_new stac927x_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), - STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), @@ -851,6 +1079,27 @@ 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 ref92hd73xx_pin_configs[12] = { + 0x02214030, 0x02a19040, 0x01a19020, 0x02214030, + 0x0181302e, 0x01014010, 0x01014020, 0x01014030, + 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050, +}; + +static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, +}; + +static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_REF] = "ref", +}; + +static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { + /* SigmaTel reference board */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, + "DFI LanParty", STAC_92HD73XX_REF), {} /* terminator */ }; @@ -2030,7 +2279,7 @@ static int stac92xx_auto_create_dmic_inp continue; num_cons = snd_hda_get_connections(codec, - spec->dmux_nid, + spec->dmux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); for (j = 0; j < num_cons; j++) @@ -2211,7 +2460,8 @@ static int stac92xx_parse_auto_config(st spec->mixers[spec->num_mixers++] = spec->kctl_alloc; spec->input_mux = &spec->private_imux; - spec->dinput_mux = &spec->private_dimux; + if (!spec->dinput_mux) + spec->dinput_mux = &spec->private_dimux; return 1; } @@ -2686,6 +2936,111 @@ static int patch_stac925x(struct hda_cod } err = -EINVAL; } + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +} + +static struct hda_input_mux stac92hd73xx_dmux = { + .num_items = 4, + .items = { + { "Analog Inputs", 0x0b }, + { "CD", 0x08 }, + { "Digital Mic 1", 0x09 }, + { "Digital Mic 2", 0x0a }, + } +}; + +static int patch_stac92hd73xx(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + hda_nid_t conn[STAC92HD73_DAC_COUNT + 2]; + int err = 0; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); + spec->pin_nids = stac92hd73xx_pin_nids; + spec->board_config = snd_hda_check_board_config(codec, + STAC_92HD73XX_MODELS, + stac92hd73xx_models, + stac92hd73xx_cfg_tbl); +again: + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + " STAC92HD73XX, 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 = stac92hd73xx_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + spec->multiout.num_dacs = snd_hda_get_connections(codec, 0x0a, + conn, STAC92HD73_DAC_COUNT + 2) - 1; + + if (spec->multiout.num_dacs < 0) { + printk(KERN_WARNING "hda_codec: Could not determine " + "number of channels defaulting to DAC count\n"); + spec->multiout.num_dacs = STAC92HD73_DAC_COUNT; + } + + switch (spec->multiout.num_dacs) { + case 0x3: /* 6 Channel */ + spec->mixer = stac92hd73xx_6ch_mixer; + spec->init = stac92hd73xx_6ch_core_init; + break; + case 0x4: /* 8 Channel */ + spec->multiout.hp_nid = 0x18; + spec->mixer = stac92hd73xx_8ch_mixer; + spec->init = stac92hd73xx_8ch_core_init; + break; + case 0x5: /* 10 Channel */ + spec->multiout.hp_nid = 0x19; + spec->mixer = stac92hd73xx_10ch_mixer; + spec->init = stac92hd73xx_10ch_core_init; + }; + + spec->multiout.dac_nids = stac92hd73xx_dac_nids; + spec->aloopback_mask = 0x801; + + spec->mux_nids = stac92hd73xx_mux_nids; + spec->adc_nids = stac92hd73xx_adc_nids; + spec->dmic_nids = stac92hd73xx_dmic_nids; + spec->dmux_nids = stac92hd73xx_dmux_nids; + + spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); + spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->dinput_mux = &stac92hd73xx_dmux; + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x000001; + stac92xx_enable_gpio_mask(codec); + + err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + + 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_92HD73XX_REF; + goto again; + } + err = -EINVAL; + } + if (err < 0) { stac92xx_free(codec); return err; @@ -2736,7 +3091,7 @@ again: spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nid = 0x1c; + spec->dmux_nids = stac92hd71bxx_dmux_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); @@ -2931,7 +3286,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nid = 0x1b; + spec->dmux_nids = stac927x_dmux_nids; /* Enable DMIC0 */ stac92xx_set_config_reg(codec, 0x13, 0x90a60040); @@ -2947,6 +3302,7 @@ static int patch_stac927x(struct hda_cod } spec->multiout.dac_nids = spec->dac_nids; + spec->aloopback_mask = 0x40; stac92xx_enable_gpio_mask(codec); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); @@ -3004,11 +3360,12 @@ static int patch_stac9205(struct hda_cod spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); spec->dmic_nids = stac9205_dmic_nids; spec->num_dmics = STAC9205_NUM_DMICS; - spec->dmux_nid = 0x1d; + spec->dmux_nids = stac9205_dmux_nids; spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; + spec->aloopback_mask = 0x40; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config){ @@ -3337,6 +3694,8 @@ 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 = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, + { .id = 0x111d7675, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d76b0, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, {} /* terminator */ };

2 3

[alsa-devel] [patch 13/18] alsa: nopage
by npiggin＠suse.de 13 Dec '07

13 Dec '07

Convert ALSA from nopage to fault. Switch from OOM to SIGBUS if the resource is not available. Signed-off-by: Nick Piggin <npiggin(a)suse.de> Cc: perex(a)perex.cz Cc: tiwai(a)suse.de Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/core/pcm_native.c | 59 ++++++++++++++++++++---------------------------- 1 file changed, 25 insertions(+), 34 deletions(-) Index: linux-2.6/sound/core/pcm_native.c =================================================================== --- linux-2.6.orig/sound/core/pcm_native.c +++ linux-2.6/sound/core/pcm_native.c @@ -3018,26 +3018,23 @@ static unsigned int snd_pcm_capture_poll /* * mmap status record */ -static struct page * snd_pcm_mmap_status_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_status_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; - struct page * page; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - page = virt_to_page(runtime->status); - get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = virt_to_page(runtime->status); + get_page(vmf->page); + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_status = { - .nopage = snd_pcm_mmap_status_nopage, + .fault = snd_pcm_mmap_status_fault, }; static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file *file, @@ -3061,26 +3058,23 @@ static int snd_pcm_mmap_status(struct sn /* * mmap control record */ -static struct page * snd_pcm_mmap_control_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_control_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; - struct page * page; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - page = virt_to_page(runtime->control); - get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = virt_to_page(runtime->control); + get_page(vmf->page); + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_control = { - .nopage = snd_pcm_mmap_control_nopage, + .fault = snd_pcm_mmap_control_fault, }; static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file, @@ -3117,10 +3111,10 @@ static int snd_pcm_mmap_control(struct s #endif /* coherent mmap */ /* - * nopage callback for mmapping a RAM page + * fault callback for mmapping a RAM page */ -static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; @@ -3130,33 +3124,30 @@ static struct page *snd_pcm_mmap_data_no size_t dma_bytes; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - offset = area->vm_pgoff << PAGE_SHIFT; - offset += address - area->vm_start; - snd_assert((offset % PAGE_SIZE) == 0, return NOPAGE_SIGBUS); + offset = vmf->pgoff << PAGE_SHIFT; dma_bytes = PAGE_ALIGN(runtime->dma_bytes); if (offset > dma_bytes - PAGE_SIZE) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; if (substream->ops->page) { page = substream->ops->page(substream, offset); - if (! page) - return NOPAGE_OOM; /* XXX: is this really due to OOM? */ + if (!page) + return VM_FAULT_SIGBUS; } else { vaddr = runtime->dma_area + offset; page = virt_to_page(vaddr); } get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = page; + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_data = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, - .nopage = snd_pcm_mmap_data_nopage, + .fault = snd_pcm_mmap_data_fault, }; /* --

2 1

[alsa-devel] [PATCH] Initial support of the Mitac 8252D (based on ALC883)
by Andy Shevchenko 13 Dec '07

13 Dec '07

Hi! The attached patch adds initial support of the Mitac 8252D (http://www.mitac-mtc.com.tw/English/products/8252Dspec.htm). Working: - Front speakers (volume + mute) - Center/LFE speakers (volume + mute) - HP out (with Front Volume) - HP individual mute switch - HP Jack sense - Front Mic and its volume Not tested: - external mic and its volume Not working while now: - Mic Jack sense Questionable: - is Mic have Jack sense? - one or two Mic volume controls? - CD/Line-in: presense in the mixer Signed-off-by: Andy Shevchenko <andy(a)smile.org.ua> --- alsa-kernel/pci/hda/patch_realtek.c.orig 2007-12-05 17:44:28.000000000 +0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2007-12-05 20:16:46.000000000 +0300 @@ -185,6 +185,7 @@ ALC883_HAIER_W66, ALC888_6ST_HP, ALC888_3ST_HP, + ALC883_MITAC, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6342,6 +6343,36 @@ { } /* end */ }; +static struct snd_kcontrol_new alc883_mitac_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -6774,6 +6805,67 @@ { } }; +/* toggle speaker-output according to the hp-jack state */ +static void alc883_mitac_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +/* auto-toggle front mic */ +/* +static void alc883_mitac_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); +} +*/ + +static void alc883_mitac_automute(struct hda_codec *codec) +{ + alc883_mitac_hp_automute(codec); + /* alc883_mitac_mic_automute(codec); */ +} + +static void alc883_mitac_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_mitac_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + /* alc883_mitac_mic_automute(codec); */ + break; + } +} + +static struct hda_verb alc883_mitac_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Subwoofer */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, */ + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7156,6 +7246,7 @@ [ALC883_HAIER_W66] = "haier-w66", [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", + [ALC883_MITAC] = "mitac", [ALC883_AUTO] = "auto", }; @@ -7190,6 +7281,7 @@ SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), @@ -7437,6 +7529,19 @@ .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, + [ALC883_MITAC] = { + .mixers = { alc883_mitac_mixer }, + .init_verbs = { alc883_init_verbs, alc883_mitac_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_mitac_unsol_event, + .init_hook = alc883_mitac_automute, + }, }; -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

3 5

[alsa-devel] [patch] Device ID for MSI L745
by zhejiang 13 Dec '07

13 Dec '07

Hi, Please refer to the [0003641] on bugtracking system. The user found that "model=targa-2ch-dig" works and provided the ID. diff -Nur a/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c b/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c --- a/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c 2007-12-04 09:00:07.000000000 +0800 +++ b/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c 2007-12-11 15:30:20.000000000 +0800 @@ -7173,6 +7173,7 @@ SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),

2 1

[alsa-devel] [PATCH] 1616(M) cardbus improvements
by Ctirad Fertr 13 Dec '07

13 Dec '07

Hello, this patch improves E-Mu 1616(M) cardbus support. It adds definitions of the new Microdock and 1010 cardbus registers (thanks again for descriptions James) and improves mixer for this card. Now you can use S/PDIF and ADAT on Mirodock and also use headpohone output on host cardbus card as another independent output. Signed-off-by: Ctirad Fertr <c.fertr(a)gmail.com> ------ Ctirad

3 7

[alsa-devel] [PATCH] hda-codec - Fix definition of AC_KNBCAP_DELTA to match datasheet.
by Andrew Paprocki 13 Dec '07

13 Dec '07

hda-codec - Fix definition of AC_KNBCAP_DELTA to match datasheet. AC_KNBCAP_DELTA is incorrectly defined as (1<<8). According to the ALC883 datasheet rev 1.3, this is bit 7 after AC_KNBCAP_NUM_STEPS which is a 0x7f mask. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r da002fb740a42b4fd6dcbd45693f9bee3c828812 -r 65b577e690e95e4ace950c3f986270dd3e992192 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Mon Dec 03 22:52:58 2007 -0500 +++ b/pci/hda/hda_codec.h Mon Dec 03 23:02:51 2007 -0500 @@ -238,7 +238,7 @@ enum { /* Volume knobs capabilities */ #define AC_KNBCAP_NUM_STEPS (0x7f<<0) -#define AC_KNBCAP_DELTA (1<<8) +#define AC_KNBCAP_DELTA (1<<7) /* * Control Parameters

2 1

[alsa-devel] [PATCH] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h.
by Andrew Paprocki 13 Dec '07

13 Dec '07

hda-codec - Add missing #defines (and 1 rename) in hda_codec.h. Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet rev 1.3. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/hda_codec.c Mon Dec 03 22:52:58 2007 -0500 @@ -1434,8 +1434,9 @@ int snd_hda_create_spdif_out_ctls(struct if (err < 0) return err; } - codec->spdif_ctls = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + codec->spdif_ctls = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } @@ -1482,7 +1483,7 @@ static int snd_hda_spdif_in_status_get(s unsigned short val; unsigned int sbits; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); sbits = convert_to_spdif_status(val); ucontrol->value.iec958.status[0] = sbits; ucontrol->value.iec958.status[1] = sbits >> 8; @@ -1533,7 +1534,8 @@ int snd_hda_create_spdif_in_ctls(struct return err; } codec->spdif_in_enable = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0) & AC_DIG1_ENABLE; return 0; } diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/hda_codec.h Mon Dec 03 22:52:58 2007 -0500 @@ -77,12 +77,14 @@ enum { #define AC_VERB_GET_PIN_SENSE 0x0f09 #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#deifne AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -110,6 +112,7 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -190,6 +193,7 @@ enum { #define AC_PINCAP_OUT (1<<4) /* output capable */ #define AC_PINCAP_IN (1<<5) /* input capable */ #define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/patch_realtek.c Mon Dec 03 22:52:58 2007 -0500 @@ -564,7 +564,7 @@ static int alc_spdif_ctrl_get(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, 0x00); + AC_VERB_GET_DIGI_CONVERT_1, 0x00); *valp = (val & mask) != 0; return 0; @@ -578,7 +578,7 @@ static int alc_spdif_ctrl_put(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long val = *ucontrol->value.integer.value; unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, + AC_VERB_GET_DIGI_CONVERT_1, 0x00); /* Set/unset the masked control bit(s) as needed */

2 2

[alsa-devel] M-audio Delta Audiophile 192 drivers
by Manel Clos 13 Dec '07

13 Dec '07

Hi all! We purchased a M-audio Delta Audiophile 192, which uses the ICE1724 chipset. Using Ubuntu gutsy 7.10 and alsa 1.0.14. PCM is working, but as soon as we start Rosegarden the complete system hangs (SysRq doesn't even work). Also, the input of the card is not shown in gnome-mixer. Is there anything we can do to help the development of the driver? We are able to compile kernel + patches, do C programming, test PCM, Midi... Thanks in advance!

3 2

[alsa-devel] Question about pcm_formats[] array in sound/core/pcm_misc.c
by Timur Tabi 13 Dec '07

13 Dec '07

I'm confused about the entries for unsigned 24-bit sound in the pcm_formats[] array: [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, .silence = { 0x80, 0x00, 0x00 }, }, and [SNDRV_PCM_FORMAT_U24_3BE] = { .width = 24, .phys = 24, .le = 0, .signd = 0, .silence = { 0x80, 0x00, 0x00 }, }, SNDRV_PCM_FORMAT_U24_BE means a 24-bit sample packed into a 32-bit word. So I presume that if the 24-bit sample is: 0x123456 then when that sample is written to memory, it looks like this: X X + 1 X + 2 X + 3 0x00 0x12 0x34 0x56 (where X is some address in memory). If that's so, shouldn't the entry for SNDRV_PCM_FORMAT_U24_BE look like this: [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, .silence = { 0x00, 0x80, 0x00, 0x00 }, }, ^^^^^^ -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] Sound problems in PCCHIPS A13G+
by alsa-devel.20.pris＠spamgourmet.com 11 Dec '07

11 Dec '07

My motherboard: PCCHIPS A13G+ http://www.pcchips.com.tw/PCCWebSite...uID=44&LanID=0 I'm using Ubuntu 7.10 - the Gutsy Gibbon, I was using Feisty before and had the same problems. The interesting thing: the system sounds all work. Some example .ogg files that came with Ubuntu also work. There is one example file however "fables_01_01_aesop.spx" which only gives a very low sound, almost inaudible. The real strange thing happens with ".avi" and mp3 files. They sound distorted and very low, almost inaudible. Some have better/louder sound, but still distorted, but I can understand what is being spoken, others not. Thanks for any help! If you need more information just ask. Technical details follow(including the Alsa Audio Debug): The sound hardware of my Mobo is in the NVIDIA MCP61S Single chipset. This chipset consists of: GeForce(R) 6100 GPU and MCP nForce(R) 405 codec(less /proc/asound/card0/codec#0): Codec: Realtek ALC660-VD Address: 0 Vendor Id: 0x10ec0660 Subsystem Id: 0x1019c601 Revision Id: 0x100001 The aplay command lists another codec: ALC861VD aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC861VD Analog [ALC861VD Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 Here is the Alsa Audio Debug: ALSA Audio Debug v0.1.0 - Mon Nov 12 00:00:12 BRST 2007 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux 2.6.22-14-generic #1 SMP Sun Oct 14 23:05:12 GMT 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 263712 4 snd_pcm_oss 44672 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80388 3 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33152 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53232 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 3 snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54660 15 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 11400 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfe028000 irq 16 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control cat: /proc/asound/hwdep: No such file or directory 00-00: ALC861VD Analog : ALC861VD Analog : playback 1 : capture 2 Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- controlC0 pcmC0D0c pcmC0D0p seq timer CPU ------------------------------------------------------- model name : AMD Processor model unknown cpu MHz : 1900.000 RAM ------------------------------------------------------- MemTotal: 969524 kB SwapTotal: 1951888 kB Hardware -------------------------------------------------- 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control

2 2

[alsa-devel] Steinberg Midex 8 driver
by Jason Woodward 11 Dec '07

11 Dec '07

I am wondering if there is still work being done on a driver for the Steinberg Midex 8. I found a message from May 2007 by Norman Schleicher that midi out is working. I had given up hope on this long ago, and am excited at the possiblity of using my Midex in linux. I would love to have even a partially working driver to play with. Thanks, Jason Woodward

1 0

[alsa-devel] Capturing from Line and Mic simultaneously
by Christophe Osuna 11 Dec '07

11 Dec '07

Hi, I would like to capture sound from the Line and Mic inputs simultaneously. The sound chip seems capable to do so: inside alsamixer, in the "Capture" tab I can select up to three subdevices, and from the "Playback" tab I can set the input source one to "Mic" and the input source two to "Line" (and three to whatever). When trying to use several output ports at the same time I use four, six or eight channels. I tried the same thing for input (i.e. use four channels) but that failed. I guess I have to use subdevices but I have no idea on how to start working with them. Could anyone: * tell me if ALSA is limited to two capture channels, and in that case what should be done; * point me to some documentation showing what to do, or give me some general hints Thanks.

2 3

[alsa-devel] cmipci: have only surround40 on 6 ch card
by SeNS 11 Dec '07

11 Dec '07

Hello, All! Sorry for repost; probably, it's a driver problem. I have 6 channel CMI8738-based card (it works in 6 channel mode under win32; and works before under Linux with ALSA). =============================================== lspci output: "0000:00:0a.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10)" =============================================== my kernel version 2.4.34 alsa version 1.0.15 =============================================== cat /proc/asound/cards 0 [CMI8738 ]: CMI8738 - C-Media CMI8738 C-Media CMI8738 (model 37) at 0xe000, irq 10 =============================================== cat /proc/asound/card0/cmipci C-Media PCI CMI8738 (model 37) at 0xe000, irq 10 00: 02 00 01 00 04: 10 1c 00 00 08: 03 00 00 01 0c: 00 00 01 00 10: d4 00 00 40 14: 00 00 00 00 18: 00 80 80 08 1c: fa 4c 04 33 20: 10 01 00 00 24: 20 01 ff 00 28: 00 00 00 00 2c: 00 00 00 00 30: ff ff ff ff 34: 00 00 00 00 38: 00 00 00 00 3c: 00 00 00 00 =============================================== WBR, SeNS

2 5

Re: [alsa-devel] [alsa-cvslog] alsa-kernel: cmipci: disable "Modem" control on version 39 or newer chips
by Thierry Vignaud 10 Dec '07

10 Dec '07

Clemens Ladisch <clemens(a)ladisch.de> writes: > changeset: 5582:5082de4abb26 > tag: tip > user: cladisch > date: Mon Dec 10 12:28:52 2007 +0100 > files: pci/cmipci.c > description: > cmipci: disable "Modem" control on version 39 or newer chips > > On version 39 or newer chips, we better remove the "Modem" control > because this register bit now mutes the front channels of the > multichannel stream. shouldn't that deserve a comment in the code? > diff -r eb09731e73d7 -r 5082de4abb26 pci/cmipci.c > --- a/pci/cmipci.c Mon Dec 03 17:08:40 2007 +0100 > +++ b/pci/cmipci.c Mon Dec 10 12:28:52 2007 +0100 > @@ -2651,10 +2651,8 @@ static struct snd_kcontrol_new snd_cmipc > }; > > /* card control switches */ > -static struct snd_kcontrol_new snd_cmipci_control_switches[] __devinitdata = { > - // DEFINE_CARD_SWITCH("Joystick", joystick), /* now module option */ > - DEFINE_CARD_SWITCH("Modem", modem), > -}; > +static struct snd_kcontrol_new snd_cmipci_modem_switch __devinitdata = > +DEFINE_CARD_SWITCH("Modem", modem); > > > static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_device) > @@ -2735,9 +2733,9 @@ static int __devinit snd_cmipci_mixer_ne > } > > /* card switches */ > - sw = snd_cmipci_control_switches; > - for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_control_switches); idx++, sw++) { > - err = snd_ctl_add(cm->card, snd_ctl_new1(sw, cm)); > + if (cm->chip_version < 39) { > + err = snd_ctl_add(cm->card, > + snd_ctl_new1(&snd_cmipci_modem_switch, cm)); > if (err < 0) > return err; > } > > > ------------------------------------------------------------------------- > SF.Net email is sponsored by: > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog

2 2

[alsa-devel] Segfault in mix_areas3_smp with Edirol UA-25 in advance-mode
by Lasse Kärkkäinen 10 Dec '07

10 Dec '07

No asound.conf, just the default dmix configuration. Trying to play audio via this card randomly causes crashes. Might be a race condition between closing the PCM at the same time as opening a new one or something like that, but I didn't debug it further. I get this with both Xine and GStreamer, using ALSA as audio output. Using other sound cards or using UA-25 without advance mode¹ there are no segfaults. Advanced Linux Sound Architecture Driver Version 1.0.14 (Fri Jul 20 09:12:58 2007 UTC). alsa-lib version: 1.0.14a GStreamer: #0 0x00002b54714d58dc in mix_areas3_smp () from /usr/lib/libasound.so.2 #1 0x00002b54714d6477 in snd_pcm_dmix_sync_area () from /usr/lib/libasound.so.2 #2 0x00002b54714d6a4e in snd_pcm_dmix_start () from /usr/lib/libasound.so.2 #3 0x00002b54714bff60 in snd_pcm_rate_start () from /usr/lib/libasound.so.2 #4 0x00002b54714ac357 in snd_pcm_write_areas () from /usr/lib/libasound.so.2 #5 0x00002b54714b4ca2 in snd_pcm_mmap_writei () from /usr/lib/libasound.so.2 #6 0x00002aaaaf89dc1c in gst_alsasink_write (asink=0x1bf1620, data=0x1a51a40, length=3760) at gstalsasink.c:786 #7 0x00002aaaadf46ca0 in audioringbuffer_thread_func (buf=<value optimized out>) at gstaudiosink.c:226 #8 0x00002b5472bf72b4 in g_thread_create_proxy () from /usr/lib/libglib-2.0.so.0 #9 0x00002b547202bfe7 in start_thread () from /lib/libpthread.so.0 Xine: #0 0x00002b9c574d88dc in mix_areas3_smp () from /usr/lib/libasound.so.2 #1 0x00002b9c574d9477 in snd_pcm_dmix_sync_area () from /usr/lib/libasound.so.2 #2 0x00002b9c574d98fa in snd_pcm_dmix_mmap_commit () from /usr/lib/libasound.so.2 #3 0x00002b9c574b96f0 in snd_pcm_plugin_write_areas () from /usr/lib/libasound.so.2 #4 0x00002b9c574af29a in snd_pcm_write_areas () from /usr/lib/libasound.so.2 #5 0x00002b9c574b9052 in snd_pcm_plugin_writei () from /usr/lib/libasound.so.2 #6 0x00002aaaaefa8078 in ao_alsa_write () from /usr/lib64/xine/plugins/1.1.8/xineplug_ao_out_alsa.so #7 0x00002b9c542397a7 in ao_loop () from /usr/lib/libxine.so.1 #8 0x00002b9c5802efe7 in start_thread () from /lib/libpthread.so.0 ¹) There is a switch on this USB sound card for switching between modes. In advance mode the device is not standard USB Audio device, but it supports 24 bit packed samples and sample rates higher than 44.1 kHz. Otherwise it is a standard USB audio device, but only supports 16 bit 44.1 kHz.

2 2

[alsa-devel] Query regarding external alsa plugins
by Pharaoh . 10 Dec '07

10 Dec '07

Hi I am writing an external IO type plugin for my board since my audio driver is going to be a user space driver i.e. no alsa kernel driver because of some technical reasons. My pluging works okay and now I want to provide a mixer control interface like volume control and playback path selection etc. Is there anything like mixer plugin etc which can be intergrated with my existing plugin? Can I get a test code somewhere? Regards, Sandeep.

1 1

[alsa-devel] hdsp rpm support
by Roman Haefeli 10 Dec '07

10 Dec '07

hi all i just wanted to ask quickly what the current state is of support for the hdsp rpm. i found some mails in the alsa-user list archive, mentioning that adding support is not so difficult and is going to be done within a week. this was in january 2005. if i can help with something, i'd be glad to do so. i am not a coder, but i have a rpm box myself. roman ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

1 0

[alsa-devel] can't use dmixer
by feng_tian 08 Dec '07

08 Dec '07

Hi, all: I have a problem with dmix of alsa. I'm using alsa v1.0.11rc2 now. And make it running on ATMEL AT91SAM9260EK board. It runs ok without Dmix. But when I tried: bash> aplay -D dmixer test.wav The errors are: ALSA lib pcm_direct.c:790:(snd_pcm_direct_initialize_slave) slave plugin does not support mmap interleaved or mmap noninterleaved access ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to initialize slave aplay: main:547: audio open error: Invalid argument The configuration of my alsa in /etc/asound.con is: pcm.card0 { type hw card 0 } pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" format S16_LE period_time 0 period_size 1024 buffer_size 4096 rate 22050 } bindings { 0 0 1 1 } } ctl.dmixer { type hw card 0 device 0 } pcm.dsp { type plug slave.pcm "dmixer" #use our new PCM here } ctl.mixer{ type hw card 0 } bash> aplay -L gives: PCM list: hw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type hw card $CARD device $DEV subdevice $SUBDEV } plughw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type plug slave.pcm { type hw card $CARD device $DEV subdevice $SUBDEV } } plug { @args.0 SLAVE @args.SLAVE { type string } type plug slave.pcm $SLAVE } shm { @args.0 SOCKET @args.1 PCM @args.SOCKET { type string } @args.PCM { type string } type shm server $SOCKET pcm $PCM } tee { @args.0 SLAVE @args.1 FILE @args.2 FORMAT @args.SLAVE { type string } @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm $SLAVE file $FILE format $FORMAT } file { @args.0 FILE @args.1 FORMAT @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm null file $FILE format $FORMAT } null { type null } cards 'cards.pcm' front 'cards.pcm.front' rear 'cards.pcm.rear' center_lfe 'cards.pcm.center_lfe' side 'cards.pcm.side' surround40 'cards.pcm.surround40' surround41 'cards.pcm.surround41' surround50 'cards.pcm.surround50' surround51 'cards.pcm.surround51' surround71 'cards.pcm.surround71' iec958 'cards.pcm.iec958' spdif 'cards.pcm.iec958' modem 'cards.pcm.modem' phoneline 'cards.pcm.phoneline' dmix 'cards.pcm.dmix' dsnoop 'cards.pcm.dsnoop' card0 { type hw card 0 } default { type plug slave.pcm dmixer } dmixer { type dmix ipc_key 1024 slave { pcm 'hw:0,0' format S16_LE period_time 0 period_size 1024 buffer_size 4096 rate 22050 } bindings { 0 0 1 1 } } dsp { type plug slave.pcm dmixer } Is there anyone experienced in this kind of error? Any helps or advices is wecome. BR Feng

3 3

[alsa-devel] rme hdsp pcmcia (and probably pci) broken
by Fernando Lopez-Lezcano 08 Dec '07

08 Dec '07

Hi, anyone has a pcmcia rme card running? And working? With a reasonably recent alsa? (this probably applies also for the regular pci rme cards, I've had reports on the Planet CCRMA list about problems that sound similar to this one). My test setup is on fc7 running: alsa-lib 1.0.15 alsa-tools 1.0.15 alsa-firmware 1.0.15 After a sequence of reboots this is what I get: 2.6.22.1-1.rt5.1 + alsa driver modules from svn 20070702.165519 --> OK 2.6.22.5-1.rt9.2 + alsa driver modules from hg20070830 --> OK 2.6.22.6-1.rt9.5 + alsa driver modules from hg20070830 --> OK 2.6.22.10-1.rt9.1 + alsa driver modules from hg20071012 --> BROKEN 2.6.23.1-1.rt11.1 + alsa drive modules from hg20071124 --> BROKEN BROKEN means: after starting hdspmixer I get no sound from the analog outputs (all previous tries play sound correctly with the default preset in hdspmixer). Some channels do appear (using a Digiface II). In the "broken" drivers I can get sound with the default preset if I send a signal to channels 7 and 12, all other stay muted. Something broke drastically in the driver between 2007.08.30 and 2007.10.12. Any help appreciated... -- Fernando

4 9