From 145794dc09117b31b6730096558e52b673af7b84 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 20 Jun 2005 10:42:44 +0200 Subject: [ALSA] ak4114: removed duplicate wake_up() AK4114 receiver - wake_up(&runtime->sleep) is already called in snd_pcm_post_stop() Signed-off-by: Jaroslav Kysela --- sound/i2c/other/ak4114.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index f5e6018ea3f4..5adde308a00f 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -554,7 +554,6 @@ int snd_ak4114_check_rate_and_errors(ak4114_t *ak4114, unsigned int flags) if (snd_pcm_running(ak4114->capture_substream)) { // printk("rate changed (%i <- %i)\n", runtime->rate, res); snd_pcm_stop(ak4114->capture_substream, SNDRV_PCM_STATE_DRAINING); - wake_up(&runtime->sleep); res = 1; } snd_pcm_stream_unlock_irqrestore(ak4114->capture_substream, _flags); -- cgit v1.2.3 From b0af0de5cb57c96b0c3d739005172152b7de0ce8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Jun 2005 14:49:19 +0200 Subject: [ALSA] hda-codec - Fix oops with ALC880 HDA Codec driver - Fixed oops with ALC880 auto-config mode - Fixed a wrong config table entry for ALC880 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bab89843d850..ed16ce817dd8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -119,6 +119,7 @@ struct alc_spec { unsigned int num_kctl_alloc, num_kctl_used; snd_kcontrol_new_t *kctl_alloc; struct hda_input_mux private_imux; + hda_nid_t private_dac_nids[4]; }; @@ -1549,7 +1550,8 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1019, .pci_subdevice = 0xa880, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1695, .pci_subdevice = 0x400d, .config = ALC880_5ST_DIG }, - { .pci_subvendor = 0x0000, .pci_subdevice = 0x8086, .config = ALC880_5ST_DIG }, + /* note subvendor = 0 below */ + /* { .pci_subvendor = 0x0000, .pci_subdevice = 0x8086, .config = ALC880_5ST_DIG }, */ { .modelname = "w810", .config = ALC880_W810 }, { .pci_subvendor = 0x161f, .pci_subdevice = 0x203d, .config = ALC880_W810 }, @@ -1656,7 +1658,8 @@ static struct alc_config_preset alc880_presets[] = { }, [ALC880_W810] = { .mixers = { alc880_w810_base_mixer }, - .init_verbs = { alc880_volume_init_verbs, alc880_pin_w810_init_verbs }, + .init_verbs = { alc880_volume_init_verbs, alc880_pin_w810_init_verbs, + alc880_gpio2_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_w810_dac_nids), .dac_nids = alc880_w810_dac_nids, .dig_out_nid = ALC880_DIGOUT_NID, @@ -1666,8 +1669,7 @@ static struct alc_config_preset alc880_presets[] = { }, [ALC880_Z71V] = { .mixers = { alc880_z71v_mixer }, - .init_verbs = { alc880_volume_init_verbs, alc880_pin_z71v_init_verbs, - alc880_gpio2_init_verbs }, + .init_verbs = { alc880_volume_init_verbs, alc880_pin_z71v_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids), .dac_nids = alc880_z71v_dac_nids, .dig_out_nid = ALC880_DIGOUT_NID, @@ -1809,6 +1811,7 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pi int i, j; memset(assigned, 0, sizeof(assigned)); + spec->multiout.dac_nids = spec->private_dac_nids; /* check the pins hardwired to audio widget */ for (i = 0; i < cfg->line_outs; i++) { -- cgit v1.2.3 From 573567e07bb4470ff177f17d1adca3f3bd310221 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 27 Jun 2005 08:17:30 +0200 Subject: [ALSA] usb-audio - high speed audio support USB generic driver Add support for endpoints with bInterval > 1, and decoding of the wMaxPacketSize field of high-speed endpoints. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 37 +++++++++++++++++++++++++------------ 1 file changed, 25 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index b5e734d975e0..facd9fc11c3c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -153,6 +153,7 @@ struct snd_usb_substream { unsigned int format; /* USB data format */ unsigned int datapipe; /* the data i/o pipe */ unsigned int syncpipe; /* 1 - async out or adaptive in */ + unsigned int datainterval; /* log_2 of data packet interval */ unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ @@ -518,7 +519,8 @@ static int prepare_playback_urb(snd_usb_substream_t *subs, if (subs->fill_max) counts = subs->maxframesize; /* fixed */ else { - subs->phase = (subs->phase & 0xffff) + subs->freqm; + subs->phase = (subs->phase & 0xffff) + + (subs->freqm << subs->datainterval); counts = subs->phase >> 16; if (counts > subs->maxframesize) counts = subs->maxframesize; @@ -899,16 +901,19 @@ static int init_substream_urbs(snd_usb_substream_t *subs, unsigned int period_by else subs->freqn = get_usb_high_speed_rate(rate); subs->freqm = subs->freqn; - subs->freqmax = subs->freqn + (subs->freqn >> 2); /* max. allowed frequency */ - subs->phase = 0; - - /* calculate the max. size of packet */ - maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) >> 16; - if (subs->maxpacksize && maxsize > subs->maxpacksize) { - //snd_printd(KERN_DEBUG "maxsize %d is greater than defined size %d\n", - // maxsize, subs->maxpacksize); + /* calculate max. frequency */ + if (subs->maxpacksize) { + /* whatever fits into a max. size packet */ maxsize = subs->maxpacksize; + subs->freqmax = (maxsize / (frame_bits >> 3)) + << (16 - subs->datainterval); + } else { + /* no max. packet size: just take 25% higher than nominal */ + subs->freqmax = subs->freqn + (subs->freqn >> 2); + maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - subs->datainterval); } + subs->phase = 0; if (subs->fill_max) subs->curpacksize = subs->maxpacksize; @@ -918,7 +923,7 @@ static int init_substream_urbs(snd_usb_substream_t *subs, unsigned int period_by if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) urb_packs = nrpacks; else - urb_packs = nrpacks * 8; + urb_packs = (nrpacks * 8) >> subs->datainterval; /* allocate a temporary buffer for playback */ if (is_playback) { @@ -991,7 +996,7 @@ static int init_substream_urbs(snd_usb_substream_t *subs, unsigned int period_by u->urb->pipe = subs->datapipe; u->urb->transfer_flags = URB_ISO_ASAP; u->urb->number_of_packets = u->packets; - u->urb->interval = 1; + u->urb->interval = 1 << subs->datainterval; u->urb->context = u; u->urb->complete = snd_usb_complete_callback(snd_complete_urb); } @@ -1195,6 +1200,12 @@ static int set_format(snd_usb_substream_t *subs, struct audioformat *fmt) subs->datapipe = usb_sndisocpipe(dev, ep); else subs->datapipe = usb_rcvisocpipe(dev, ep); + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH && + get_endpoint(alts, 0)->bInterval >= 1 && + get_endpoint(alts, 0)->bInterval <= 4) + subs->datainterval = get_endpoint(alts, 0)->bInterval - 1; + else + subs->datainterval = 0; subs->syncpipe = subs->syncinterval = 0; subs->maxpacksize = fmt->maxpacksize; subs->fill_max = 0; @@ -2492,8 +2503,10 @@ static int parse_audio_endpoints(snd_usb_audio_t *chip, int iface_no) fp->altset_idx = i; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - /* FIXME: decode wMaxPacketSize of high bandwith endpoints */ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) + fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) + * (fp->maxpacksize & 0x7ff); fp->attributes = csep[3]; /* some quirks for attributes here */ -- cgit v1.2.3 From b4d3f9d452ec574e0ffb292267427f69bb470631 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 27 Jun 2005 08:18:27 +0200 Subject: [ALSA] usb-audio - fix capture of non-48k sample rates on Audigy 2 NX USB generic driver On the SB Audigy 2 NX, capturing with sample rates that are not a multiple of 48 kHz does not seem to work, so disable it. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index facd9fc11c3c..c57b44511b54 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2408,10 +2408,9 @@ static int parse_audio_format(snd_usb_audio_t *chip, struct audioformat *fp, if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020)) { if (fmt[3] == USB_FORMAT_TYPE_I && - stream == SNDRV_PCM_STREAM_PLAYBACK && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) - return -1; /* use 48k only */ + return -1; } #endif return 0; -- cgit v1.2.3 From c7d4b2fa3169a1206450bc445d1997a17479644f Mon Sep 17 00:00:00 2001 From: Matt Date: Mon, 27 Jun 2005 14:59:41 +0200 Subject: [ALSA] hda-codec - SigmaTel HDA multichannel support HDA Codec driver Adds 6/8 channel support to the SigmaTel HDA patch. Please apply. Signed-off-by: Matt Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 733 ++++++++++++++++++++++++++++------------- 1 file changed, 496 insertions(+), 237 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 013be2ea513a..fad825677e7c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -30,32 +30,33 @@ #include #include #include +#include #include "hda_codec.h" #include "hda_local.h" #undef STAC_TEST struct sigmatel_spec { + snd_kcontrol_new_t *mixers[4]; + unsigned int num_mixers; + + unsigned int surr_switch: 1; + /* playback */ struct hda_multi_out multiout; - hda_nid_t playback_nid; + hda_nid_t dac_nids[4]; /* capture */ hda_nid_t *adc_nids; unsigned int num_adcs; hda_nid_t *mux_nids; unsigned int num_muxes; - hda_nid_t capture_nid; hda_nid_t dig_in_nid; - /* power management*/ - hda_nid_t *pstate_nids; - unsigned int num_pstates; - +#ifdef STAC_TEST /* pin widgets */ hda_nid_t *pin_nids; unsigned int num_pins; -#ifdef STAC_TEST unsigned int *pin_configs; #endif @@ -64,16 +65,20 @@ struct sigmatel_spec { snd_kcontrol_new_t *mixer; /* capture source */ - struct hda_input_mux input_mux; - char input_labels[HDA_MAX_NUM_INPUTS][16]; + struct hda_input_mux *input_mux; unsigned int cur_mux[2]; /* channel mode */ unsigned int num_ch_modes; unsigned int cur_ch_mode; - const struct sigmatel_channel_mode *channel_modes; - struct hda_pcm pcm_rec[1]; /* PCM information */ + struct hda_pcm pcm_rec[2]; /* PCM information */ + + /* dynamic controls and input_mux */ + struct auto_pin_cfg autocfg; + unsigned int num_kctl_alloc, num_kctl_used; + snd_kcontrol_new_t *kctl_alloc; + struct hda_input_mux private_imux; }; static hda_nid_t stac9200_adc_nids[1] = { @@ -88,14 +93,6 @@ static hda_nid_t stac9200_dac_nids[1] = { 0x02, }; -static hda_nid_t stac9200_pstate_nids[3] = { - 0x01, 0x02, 0x03, -}; - -static hda_nid_t stac9200_pin_nids[8] = { - 0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, -}; - static hda_nid_t stac922x_adc_nids[2] = { 0x06, 0x07, }; @@ -104,24 +101,22 @@ static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; -static hda_nid_t stac922x_dac_nids[4] = { - 0x02, 0x03, 0x04, 0x05, -}; - -static hda_nid_t stac922x_pstate_nids[8] = { - 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x11, +#ifdef STAC_TEST +static hda_nid_t stac9200_pin_nids[8] = { + 0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, }; static hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, }; +#endif static int stac92xx_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(&spec->input_mux, uinfo); + return snd_hda_input_mux_info(spec->input_mux, uinfo); } static int stac92xx_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) @@ -140,26 +135,64 @@ static int stac92xx_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t struct sigmatel_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); } -static struct hda_verb stac9200_ch2_init[] = { +static struct hda_verb stac9200_core_init[] = { /* set dac0mux for dac converter */ - { 0x07, 0x701, 0x00}, + { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, {} }; -static struct hda_verb stac922x_ch2_init[] = { +static struct hda_verb stac922x_core_init[] = { /* set master volume and direct control */ - { 0x16, 0x70f, 0xff}, + { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, {} }; -struct sigmatel_channel_mode { - unsigned int channels; - const struct hda_verb *sequence; -}; +static int stac922x_channel_modes[3] = {2, 6, 8}; + +static int stac922x_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = spec->num_ch_modes; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; + sprintf(uinfo->value.enumerated.name, "%dch", + stac922x_channel_modes[uinfo->value.enumerated.item]); + return 0; +} + +static int stac922x_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_ch_mode; + return 0; +} + +static int stac922x_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + if (ucontrol->value.enumerated.item[0] >= spec->num_ch_modes) + ucontrol->value.enumerated.item[0] = spec->num_ch_modes; + if (ucontrol->value.enumerated.item[0] == spec->cur_ch_mode && + ! codec->in_resume) + return 0; + + spec->cur_ch_mode = ucontrol->value.enumerated.item[0]; + spec->multiout.max_channels = stac922x_channel_modes[spec->cur_ch_mode]; + + return 1; +} static snd_kcontrol_new_t stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), @@ -174,13 +207,12 @@ static snd_kcontrol_new_t stac9200_mixer[] = { }, HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Input Mux Volume", 0x0c, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT), { } /* end */ }; +/* This needs to be generated dynamically based on sequence */ static snd_kcontrol_new_t stac922x_mixer[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x2, 0x0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -195,14 +227,38 @@ static snd_kcontrol_new_t stac922x_mixer[] = { { } /* end */ }; +static snd_kcontrol_new_t stac922x_ch_mode_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = stac922x_ch_mode_info, + .get = stac922x_ch_mode_get, + .put = stac922x_ch_mode_put, + }, + { } /* end */ +}; + static int stac92xx_build_controls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; int err; + int i; err = snd_hda_add_new_ctls(codec, spec->mixer); 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->surr_switch) { + err = snd_hda_add_new_ctls(codec, stac922x_ch_mode_mixer); + if (err < 0) + return err; + } if (spec->multiout.dig_out_nid) { err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) @@ -222,9 +278,9 @@ static unsigned int stac9200_pin_configs[8] = { 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; -static unsigned int stac922x_pin_configs[14] = { - 0x40000100, 0x40000100, 0x40000100, 0x01114010, - 0x01813122, 0x40000100, 0x01447010, 0x01c47010, +static unsigned int stac922x_pin_configs[10] = { + 0x01014010, 0x01014011, 0x01014012, 0x0221401f, + 0x01813122, 0x01014014, 0x01441030, 0x01c41030, 0x40000100, 0x40000100, }; @@ -255,180 +311,66 @@ static void stac92xx_set_config_regs(struct hda_codec *codec) } #endif -static int stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, unsigned int value) -{ - unsigned int pin_ctl; - - pin_ctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, - 0x00); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl | value); - - return 0; -} - -static int stac92xx_set_vref(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int vref_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) >> AC_PINCAP_VREF_SHIFT; - unsigned int vref_ctl = AC_PINCTL_VREF_HIZ; - - if (vref_caps & AC_PINCAP_VREF_100) - vref_ctl = AC_PINCTL_VREF_100; - else if (vref_caps & AC_PINCAP_VREF_80) - vref_ctl = AC_PINCTL_VREF_80; - else if (vref_caps & AC_PINCAP_VREF_50) - vref_ctl = AC_PINCTL_VREF_50; - else if (vref_caps & AC_PINCAP_VREF_GRD) - vref_ctl = AC_PINCTL_VREF_GRD; - - stac92xx_set_pinctl(codec, nid, vref_ctl); - - return 0; -} - /* - * retrieve the default device type from the default config value + * Analog playback callbacks */ -#define get_defcfg_type(cfg) ((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT) -#define get_defcfg_location(cfg) ((cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT) - -static int stac92xx_config_pin(struct hda_codec *codec, hda_nid_t nid, unsigned int pin_cfg) +static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + snd_pcm_substream_t *substream) { struct sigmatel_spec *spec = codec->spec; - u32 location = get_defcfg_location(pin_cfg); - char *label; - const char *type = NULL; - int ainput = 0; - - switch(get_defcfg_type(pin_cfg)) { - case AC_JACK_HP_OUT: - /* Enable HP amp */ - stac92xx_set_pinctl(codec, nid, AC_PINCTL_HP_EN); - /* Fall through */ - case AC_JACK_SPDIF_OUT: - case AC_JACK_LINE_OUT: - case AC_JACK_SPEAKER: - /* Enable output */ - stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); - break; - case AC_JACK_SPDIF_IN: - stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN); - break; - case AC_JACK_MIC_IN: - if ((location & 0x0f) == AC_JACK_LOC_FRONT) - type = "Front Mic"; - else - type = "Mic"; - ainput = 1; - /* Set vref */ - stac92xx_set_vref(codec, nid); - stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN); - break; - case AC_JACK_CD: - type = "CD"; - ainput = 1; - stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN); - break; - case AC_JACK_LINE_IN: - if ((location & 0x0f) == AC_JACK_LOC_FRONT) - type = "Front Line"; - else - type = "Line"; - ainput = 1; - stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN); - break; - case AC_JACK_AUX: - if ((location & 0x0f) == AC_JACK_LOC_FRONT) - type = "Front Aux"; - else - type = "Aux"; - ainput = 1; - stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN); - break; - } - - if (ainput) { - hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; - int i, j, num_cons, index = -1; - if (!type) - type = "Input"; - label = spec->input_labels[spec->input_mux.num_items]; - strcpy(label, type); - spec->input_mux.items[spec->input_mux.num_items].label = label; - for (i=0; inum_muxes; i++) { - num_cons = snd_hda_get_connections(codec, spec->mux_nids[i], con_lst, HDA_MAX_NUM_INPUTS); - for (j=0; j= 0) - break; - } - spec->input_mux.items[spec->input_mux.num_items].index = index; - spec->input_mux.num_items++; - } - - return 0; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); } -static int stac92xx_config_pins(struct hda_codec *codec) +/* + * set up the i/o for analog out + * when the digital out is available, copy the front out to digital out, too. + */ +static int stac92xx_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + snd_pcm_substream_t *substream) { - struct sigmatel_spec *spec = codec->spec; + hda_nid_t *nids = mout->dac_nids; + int chs = substream->runtime->channels; int i; - unsigned int pin_cfg; - - for (i=0; i < spec->num_pins; i++) { - /* Default to disabled */ - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - 0x00); - - pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - if (((pin_cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT) == AC_JACK_PORT_NONE) - continue; /* Move on */ - stac92xx_config_pin(codec, spec->pin_nids[i], pin_cfg); + down(&codec->spdif_mutex); + if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { + if (chs == 2 && + snd_hda_is_supported_format(codec, mout->dig_out_nid, format) && + ! (codec->spdif_status & IEC958_AES0_NONAUDIO)) { + mout->dig_out_used = HDA_DIG_ANALOG_DUP; + /* setup digital receiver */ + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, + stream_tag, 0, format); + } else { + mout->dig_out_used = 0; + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); + } } - - return 0; -} - -static int stac92xx_init(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - - for (i=0; i < spec->num_pstates; i++) - snd_hda_codec_write(codec, spec->pstate_nids[i], 0, - AC_VERB_SET_POWER_STATE, 0x00); - - mdelay(100); - - snd_hda_sequence_write(codec, spec->init); - -#ifdef STAC_TEST - stac92xx_set_config_regs(codec); -#endif - - stac92xx_config_pins(codec); - + up(&codec->spdif_mutex); + + /* front */ + snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); + if (mout->hp_nid) + /* headphone out will just decode front left/right (stereo) */ + snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); + /* surrounds */ + if (mout->max_channels > 2) + for (i = 1; i < mout->num_dacs; i++) { + if ((mout->max_channels == 6) && (i == 3)) + break; + if (chs >= (i + 1) * 2) /* independent out */ + snd_hda_codec_setup_stream(codec, nids[i], stream_tag, i * 2, + format); + else /* copy front */ + snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, + format); + } return 0; } -/* - * Analog playback callbacks - */ -static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - snd_pcm_substream_t *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream); -} static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, @@ -437,7 +379,7 @@ static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, snd_pcm_substream_t *substream) { struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, + return stac92xx_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream); } @@ -516,7 +458,7 @@ static struct hda_pcm_stream stac92xx_pcm_digital_capture = { static struct hda_pcm_stream stac92xx_pcm_analog_playback = { .substreams = 1, .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .nid = 0x02, /* NID to query formats and rates */ .ops = { .open = stac92xx_playback_pcm_open, @@ -544,11 +486,9 @@ static int stac92xx_build_pcms(struct hda_codec *codec) codec->num_pcms = 1; codec->pcm_info = info; - info->name = "STAC92xx"; + info->name = "STAC92xx Analog"; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->playback_nid; info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->capture_nid; if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms++; @@ -567,9 +507,325 @@ static int stac92xx_build_pcms(struct hda_codec *codec) return 0; } +#define NUM_CONTROL_ALLOC 32 + +enum { + STAC_CTL_WIDGET_VOL, + STAC_CTL_WIDGET_MUTE, +}; + +static snd_kcontrol_new_t stac92xx_control_templates[] = { + HDA_CODEC_VOLUME(NULL, 0, 0, 0), + HDA_CODEC_MUTE(NULL, 0, 0, 0), +}; + +/* add dynamic controls */ +static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char *name, unsigned long val) +{ + snd_kcontrol_new_t *knew; + + if (spec->num_kctl_used >= spec->num_kctl_alloc) { + int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC; + + knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */ + if (! knew) + return -ENOMEM; + if (spec->kctl_alloc) { + memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc); + kfree(spec->kctl_alloc); + } + spec->kctl_alloc = knew; + spec->num_kctl_alloc = num; + } + + knew = &spec->kctl_alloc[spec->num_kctl_used]; + *knew = stac92xx_control_templates[type]; + knew->name = snd_kmalloc_strdup(name, GFP_KERNEL); + if (! knew->name) + return -ENOMEM; + knew->private_value = val; + spec->num_kctl_used++; + return 0; +} + +/* fill in the dac_nids table from the parsed pin configuration */ +static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, const struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid; + int i; + + /* check the pins hardwired to audio widget */ + for (i = 0; i < cfg->line_outs; i++) { + nid = cfg->line_out_pins[i]; + spec->multiout.dac_nids[i] = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + } + + spec->multiout.num_dacs = cfg->line_outs; + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; + hda_nid_t nid; + int i, err; + + for (i = 0; i < cfg->line_outs; i++) { + if (! spec->multiout.dac_nids[i]) + continue; + + nid = spec->multiout.dac_nids[i]; + + if (i == 2) { + /* Center/LFE */ + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0) + return err; + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) + return err; + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0) + return err; + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + return err; + } + } + + return 0; +} + +/* add playback controls for HP output */ +static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t pin = cfg->hp_pin; + hda_nid_t nid; + int i, err; + + if (! pin) + return 0; + + nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + for (i = 0; i < cfg->line_outs; i++) { + if (! spec->multiout.dac_nids[i]) + continue; + if (spec->multiout.dac_nids[i] == nid) + return 0; + } + + spec->multiout.hp_nid = nid; + + /* control HP volume/switch on the output mixer amp */ + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + return err; + if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + return err; + + return 0; +} + +/* create playback/capture controls for input pins */ +static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + static char *labels[AUTO_PIN_LAST] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" + }; + struct hda_input_mux *imux = &spec->private_imux; + hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; + int i, j, k; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + int index = -1; + if (cfg->input_pins[i]) { + imux->items[imux->num_items].label = labels[i]; + + for (j=0; jnum_muxes; j++) { + int num_cons = snd_hda_get_connections(codec, spec->mux_nids[j], con_lst, HDA_MAX_NUM_INPUTS); + for (k=0; kinput_pins[i]) { + index = k; + break; + } + if (index >= 0) + break; + } + imux->items[imux->num_items].index = index; + imux->num_items++; + } + } + + return 0; +} + +static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type) + +{ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); +} + +static void stac92xx_auto_init_multi_out(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->autocfg.line_outs; i++) { + hda_nid_t nid = spec->autocfg.line_out_pins[i]; + stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); + } +} + +static void stac92xx_auto_init_hp_out(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t pin; + + pin = spec->autocfg.hp_pin; + if (pin) /* connect to front */ + stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); +} + +static int stac922x_parse_auto_config(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int err; + + if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0) + return err; + if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) + return err; + if (! spec->autocfg.line_outs && ! spec->autocfg.hp_pin) + return 0; /* can't find valid pin config */ + + if ((err = stac92xx_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || + (err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg)) < 0 || + (err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + if (spec->multiout.max_channels > 2) { + spec->surr_switch = 1; + spec->cur_ch_mode = 1; + spec->num_ch_modes = 2; + if (spec->multiout.max_channels == 8) { + spec->cur_ch_mode++; + spec->num_ch_modes++; + } + } + + if (spec->autocfg.dig_out_pin) { + spec->multiout.dig_out_nid = 0x08; + stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_out_pin, AC_PINCTL_OUT_EN); + } + if (spec->autocfg.dig_in_pin) { + spec->dig_in_nid = 0x09; + stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_in_pin, AC_PINCTL_IN_EN); + } + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->input_mux = &spec->private_imux; + + return 1; +} + +static int stac9200_parse_auto_config(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int err; + + if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0) + return err; + + if ((err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0) + return err; + + if (spec->autocfg.dig_out_pin) { + spec->multiout.dig_out_nid = 0x05; + stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_out_pin, AC_PINCTL_OUT_EN); + } + if (spec->autocfg.dig_in_pin) { + spec->dig_in_nid = 0x04; + stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_in_pin, AC_PINCTL_IN_EN); + } + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->input_mux = &spec->private_imux; + + return 1; +} + +static int stac92xx_init_pstate(struct hda_codec *codec) +{ + hda_nid_t nid, nid_start; + int nodes; + + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_POWER_STATE, 0x00); + + nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid_start); + for (nid = nid_start; nid < nodes + nid_start; nid++) { + unsigned int wid_caps = snd_hda_param_read(codec, nid, + AC_PAR_AUDIO_WIDGET_CAP); + if (wid_caps & AC_WCAP_POWER) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_POWER_STATE, 0x00); + } + + mdelay(100); + + return 0; +} + +static int stac92xx_init(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + stac92xx_init_pstate(codec); + + snd_hda_sequence_write(codec, spec->init); + + stac92xx_auto_init_multi_out(codec); + stac92xx_auto_init_hp_out(codec); + + return 0; +} + static void stac92xx_free(struct hda_codec *codec) { - kfree(codec->spec); + struct sigmatel_spec *spec = codec->spec; + int i; + + if (! spec) + return; + + if (spec->kctl_alloc) { + for (i = 0; i < spec->num_kctl_used; i++) + kfree(spec->kctl_alloc[i].name); + kfree(spec->kctl_alloc); + } + + kfree(spec); } static struct hda_codec_ops stac92xx_patch_ops = { @@ -582,6 +838,7 @@ static struct hda_codec_ops stac92xx_patch_ops = { static int patch_stac9200(struct hda_codec *codec) { struct sigmatel_spec *spec; + int err; spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -589,26 +846,27 @@ static int patch_stac9200(struct hda_codec *codec) codec->spec = spec; +#ifdef STAC_TEST + spec->pin_nids = stac9200_pin_nids; + spec->num_pins = 8; + spec->pin_configs = stac9200_pin_configs; + stac92xx_set_config_regs(codec); +#endif spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; spec->multiout.dac_nids = stac9200_dac_nids; - spec->multiout.dig_out_nid = 0x05; - spec->dig_in_nid = 0x04; spec->adc_nids = stac9200_adc_nids; spec->mux_nids = stac9200_mux_nids; spec->num_muxes = 1; - spec->input_mux.num_items = 0; - spec->pstate_nids = stac9200_pstate_nids; - spec->num_pstates = 3; - spec->pin_nids = stac9200_pin_nids; -#ifdef STAC_TEST - spec->pin_configs = stac9200_pin_configs; -#endif - spec->num_pins = 8; - spec->init = stac9200_ch2_init; + + spec->init = stac9200_core_init; spec->mixer = stac9200_mixer; - spec->playback_nid = 0x02; - spec->capture_nid = 0x03; + + err = stac9200_parse_auto_config(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } codec->patch_ops = stac92xx_patch_ops; @@ -618,6 +876,7 @@ static int patch_stac9200(struct hda_codec *codec) static int patch_stac922x(struct hda_codec *codec) { struct sigmatel_spec *spec; + int err; spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -625,26 +884,26 @@ static int patch_stac922x(struct hda_codec *codec) codec->spec = spec; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 4; - spec->multiout.dac_nids = stac922x_dac_nids; - spec->multiout.dig_out_nid = 0x08; - spec->dig_in_nid = 0x09; - spec->adc_nids = stac922x_adc_nids; - spec->mux_nids = stac922x_mux_nids; - spec->num_muxes = 2; - spec->input_mux.num_items = 0; - spec->pstate_nids = stac922x_pstate_nids; - spec->num_pstates = 8; - spec->pin_nids = stac922x_pin_nids; #ifdef STAC_TEST + spec->num_pins = 10; + spec->pin_nids = stac922x_pin_nids; spec->pin_configs = stac922x_pin_configs; + stac92xx_set_config_regs(codec); #endif - spec->num_pins = 10; - spec->init = stac922x_ch2_init; + spec->adc_nids = stac922x_adc_nids; + spec->mux_nids = stac922x_mux_nids; + spec->num_muxes = 2; + + spec->init = stac922x_core_init; spec->mixer = stac922x_mixer; - spec->playback_nid = 0x02; - spec->capture_nid = 0x06; + + spec->multiout.dac_nids = spec->dac_nids; + + err = stac922x_parse_auto_config(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } codec->patch_ops = stac92xx_patch_ops; -- cgit v1.2.3 From ff6fdc37fbe66e24ef9ad7c23a278ff757480dda Mon Sep 17 00:00:00 2001 From: Matt Date: Mon, 27 Jun 2005 15:06:52 +0200 Subject: [ALSA] hda-codec - SigmaTel HDA resume support HDA Codec driver Adds resume support to the SigmaTel HDA patch. Please apply. Signed-off-by: Matt Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index fad825677e7c..01cc58a247c1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -828,11 +828,32 @@ static void stac92xx_free(struct hda_codec *codec) kfree(spec); } +#ifdef CONFIG_PM +static int stac92xx_resume(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + + stac92xx_init(codec); + for (i = 0; i < spec->num_mixers; i++) + snd_hda_resume_ctls(codec, spec->mixers[i]); + if (spec->multiout.dig_out_nid) + snd_hda_resume_spdif_out(codec); + if (spec->dig_in_nid) + snd_hda_resume_spdif_in(codec); + + return 0; +} +#endif + static struct hda_codec_ops stac92xx_patch_ops = { .build_controls = stac92xx_build_controls, .build_pcms = stac92xx_build_pcms, .init = stac92xx_init, .free = stac92xx_free, +#ifdef CONFIG_PM + .resume = stac92xx_resume, +#endif }; static int patch_stac9200(struct hda_codec *codec) -- cgit v1.2.3 From b95eed7cde4a44476fa12e776e090fc494059458 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Jun 2005 15:07:33 +0200 Subject: [ALSA] trident - Shut up compile warnings Trident driver Shut up compile warnings about uninitialized variables. Signed-off-by: Takashi Iwai --- sound/pci/trident/trident_main.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index a09b0fb49e81..b01c91bb5f6a 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -472,6 +472,7 @@ void snd_trident_write_voice_regs(trident_t * trident, break; default: snd_BUG(); + return; } outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR)); -- cgit v1.2.3 From 548e7823bc33b8cde4de59dfafe0fd69d951d3b5 Mon Sep 17 00:00:00 2001 From: Harald Welte Date: Mon, 27 Jun 2005 15:10:56 +0200 Subject: [ALSA] Add new pci device id (SB400) to atiixp-modem ATIIXP-modem driver I didn't actually test whether the modem works, but at least the driver loads and initializes fine. Please consider inclusion. Signed-off-by: Takashi Iwai --- sound/pci/atiixp_modem.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index a6b4b8d589fd..8d2002951bd7 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -265,6 +265,7 @@ struct snd_atiixp { */ static struct pci_device_id snd_atiixp_ids[] = { { 0x1002, 0x434d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */ + { 0x1002, 0x4378, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */ { 0, } }; -- cgit v1.2.3 From b6482d48e536729829025262d6529df09ae20396 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Jun 2005 15:32:43 +0200 Subject: [ALSA] hda-codec - Add 6stack model for ALC880 Documentation,HDA Codec driver - Added a new '6stack' model for ALC880. - Fixed the typo in 6stack-digout model name. - Added description for missing models in ALSA-Configuration.txt. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ed16ce817dd8..9b4339a004fc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -40,6 +40,7 @@ enum { ALC880_W810, ALC880_Z71V, ALC880_AUTO, + ALC880_6ST, ALC880_6ST_DIG, ALC880_F1734, ALC880_ASUS, @@ -1559,7 +1560,9 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .modelname = "z71v", .config = ALC880_Z71V }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_Z71V }, - { .modelname = "6statack-digout", .config = ALC880_6ST_DIG }, + { .modelname = "6stack", .config = ALC880_6ST }, + + { .modelname = "6stack-digout", .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x2668, .pci_subdevice = 0x8086, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x8086, .pci_subdevice = 0x2668, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x1462, .pci_subdevice = 0x1150, .config = ALC880_6ST_DIG }, @@ -1646,6 +1649,15 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_fivestack_modes, .input_mux = &alc880_capture_source, }, + [ALC880_6ST] = { + .mixers = { alc880_six_stack_mixer }, + .init_verbs = { alc880_volume_init_verbs, alc880_pin_6stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids), + .dac_nids = alc880_6st_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes), + .channel_mode = alc880_sixstack_modes, + .input_mux = &alc880_6stack_capture_source, + }, [ALC880_6ST_DIG] = { .mixers = { alc880_six_stack_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_pin_6stack_init_verbs }, -- cgit v1.2.3 From 7a318a70a42057692f191ff49c289cd3e27e21f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Jun 2005 14:16:21 +0200 Subject: [ALSA] hda-codec - Add entry for Acer APFV HDA Codec driver Added the model entry for Acer APFV. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b4339a004fc..9b8569900787 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1549,7 +1549,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x8086, .pci_subdevice = 0xa100, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1565, .pci_subdevice = 0x8202, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1019, .pci_subdevice = 0xa880, .config = ALC880_5ST_DIG }, - { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG }, + /* { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG }, */ /* conflict with 6stack */ { .pci_subvendor = 0x1695, .pci_subdevice = 0x400d, .config = ALC880_5ST_DIG }, /* note subvendor = 0 below */ /* { .pci_subvendor = 0x0000, .pci_subdevice = 0x8086, .config = ALC880_5ST_DIG }, */ @@ -1561,6 +1561,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_Z71V }, { .modelname = "6stack", .config = ALC880_6ST }, + { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_6ST }, /* Acer APFV */ { .modelname = "6stack-digout", .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x2668, .pci_subdevice = 0x8086, .config = ALC880_6ST_DIG }, -- cgit v1.2.3 From a3352f01ea2d38b0d5b7b63de754e94b9aba0390 Mon Sep 17 00:00:00 2001 From: Henrik Kretzschmar Date: Wed, 29 Jun 2005 19:30:42 +0200 Subject: [ALSA] Fix two typos and changes on snd_assert() ALSA Core Both typos were in the kerneldocs. I splitted the snd_assert() calls in one-expression-per-call for better debugging. Signed-off-by: Henrik Kretzschmar Signed-off-by: Takashi Iwai --- sound/core/device.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/core/device.c b/sound/core/device.c index 18c71f913d2a..ca00ad7740c9 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -28,7 +28,7 @@ /** * snd_device_new - create an ALSA device component * @card: the card instance - * @type: the device type, SNDRV_DEV_TYPE_XXX + * @type: the device type, SNDRV_DEV_XXX * @device_data: the data pointer of this device * @ops: the operator table * @@ -46,7 +46,9 @@ int snd_device_new(snd_card_t *card, snd_device_type_t type, { snd_device_t *dev; - snd_assert(card != NULL && device_data != NULL && ops != NULL, return -ENXIO); + snd_assert(card != NULL, return -ENXIO); + snd_assert(device_data != NULL, return -ENXIO); + snd_assert(ops != NULL, return -ENXIO); dev = kcalloc(1, sizeof(*dev), GFP_KERNEL); if (dev == NULL) return -ENOMEM; @@ -102,7 +104,7 @@ int snd_device_free(snd_card_t *card, void *device_data) } /** - * snd_device_free - disconnect the device + * snd_device_disconnect - disconnect the device * @card: the card instance * @device_data: the data pointer to disconnect * @@ -118,7 +120,7 @@ int snd_device_disconnect(snd_card_t *card, void *device_data) { struct list_head *list; snd_device_t *dev; - + snd_assert(card != NULL, return -ENXIO); snd_assert(device_data != NULL, return -ENXIO); list_for_each(list, &card->devices) { @@ -154,8 +156,9 @@ int snd_device_register(snd_card_t *card, void *device_data) struct list_head *list; snd_device_t *dev; int err; - - snd_assert(card != NULL && device_data != NULL, return -ENXIO); + + snd_assert(card != NULL, return -ENXIO); + snd_assert(device_data != NULL, return -ENXIO); list_for_each(list, &card->devices) { dev = snd_device(list); if (dev->device_data != device_data) -- cgit v1.2.3 From 0884484762f731a1d5446c0b618a74c5957dea4f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 09:55:31 +0200 Subject: [ALSA] Fix dependency of GUS driver ALSA sequencer Add the missing snd-seq-midi-emul to SND_GUS_SYNTH list. Signed-off-by: Takashi Iwai --- sound/core/seq/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index 64cb50d7b589..402e2b4a34c6 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -38,7 +38,7 @@ obj-$(CONFIG_SND_VIRMIDI) += snd-seq-virmidi.o snd-seq-midi-event.o obj-$(call sequencer,$(CONFIG_SND_RAWMIDI)) += snd-seq-midi.o snd-seq-midi-event.o obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o snd-seq-instr.o obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o snd-seq-instr.o -obj-$(call sequencer,$(CONFIG_SND_GUS_SYNTH)) += snd-seq-instr.o +obj-$(call sequencer,$(CONFIG_SND_GUS_SYNTH)) += snd-seq-midi-emul.o snd-seq-instr.o obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-seq-midi-emul.o snd-seq-virmidi.o obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-seq-midi-emul.o snd-seq-virmidi.o obj-$(call sequencer,$(CONFIG_SND_TRIDENT)) += snd-seq-midi-emul.o snd-seq-instr.o -- cgit v1.2.3 From 82fe0c5803f4c77ffeb4c1c2367defb3dcedad45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 10:54:33 +0200 Subject: [ALSA] Use kstrdup HDA Codec driver Use the new kstrdup() function instead of in-house one. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 01cc58a247c1..07d06f7c4390 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -540,7 +540,7 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char knew = &spec->kctl_alloc[spec->num_kctl_used]; *knew = stac92xx_control_templates[type]; - knew->name = snd_kmalloc_strdup(name, GFP_KERNEL); + knew->name = kstrdup(name, GFP_KERNEL); if (! knew->name) return -ENOMEM; knew->private_value = val; -- cgit v1.2.3 From 6d00a3127972e7853d6296ffc1e72c5b1a23d937 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 13:40:51 +0200 Subject: [ALSA] Fix and clean-up of vxpocket driver Documentation,PCMCIA Kconfig,Digigram VX Pocket driver - Fixed Oops with request_firmware() - Detect the card type in runtime (vxpoocket v2 or 440) - snd-vxp440 driver is merged to snd-vxpocket - Clean up the code Signed-off-by: Takashi Iwai --- sound/pcmcia/Kconfig | 15 +- sound/pcmcia/vx/Makefile | 7 +- sound/pcmcia/vx/vxpocket.c | 427 +++++++++++++++++++++++++++++++++++++++------ sound/pcmcia/vx/vxpocket.h | 27 +-- 4 files changed, 378 insertions(+), 98 deletions(-) (limited to 'sound') diff --git a/sound/pcmcia/Kconfig b/sound/pcmcia/Kconfig index 3611e298834f..5d1b0b762efa 100644 --- a/sound/pcmcia/Kconfig +++ b/sound/pcmcia/Kconfig @@ -8,23 +8,12 @@ config SND_VXPOCKET depends on SND && PCMCIA && ISA select SND_VX_LIB help - Say Y here to include support for Digigram VXpocket - soundcards. + Say Y here to include support for Digigram VXpocket and + VXpocket 440 soundcards. To compile this driver as a module, choose M here: the module will be called snd-vxpocket. -config SND_VXP440 - tristate "Digigram VXpocket 440" - depends on SND && PCMCIA && ISA - select SND_VX_LIB - help - Say Y here to include support for Digigram VXpocket 440 - soundcards. - - To compile this driver as a module, choose M here: the module - will be called snd-vxp440. - config SND_PDAUDIOCF tristate "Sound Core PDAudioCF" depends on SND && PCMCIA && ISA diff --git a/sound/pcmcia/vx/Makefile b/sound/pcmcia/vx/Makefile index f35dfa1af094..54971f01e968 100644 --- a/sound/pcmcia/vx/Makefile +++ b/sound/pcmcia/vx/Makefile @@ -3,9 +3,6 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-vx-cs-objs := vx_entry.o vxp_ops.o vxp_mixer.o -snd-vxpocket-objs := vxpocket.o -snd-vxp440-objs := vxp440.o +snd-vxpocket-objs := vxpocket.o vxp_ops.o vxp_mixer.o -obj-$(CONFIG_SND_VXPOCKET) += snd-vxpocket.o snd-vx-cs.o -obj-$(CONFIG_SND_VXP440) += snd-vxp440.o snd-vx-cs.o +obj-$(CONFIG_SND_VXPOCKET) += snd-vxpocket.o diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 62d6fa128148..3a82161d3b24 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -24,21 +24,17 @@ #include #include #include "vxpocket.h" +#include +#include #include /* */ -#ifdef COMPILE_VXP440 -#define CARD_NAME "VXPocket440" -#else -#define CARD_NAME "VXPocket" -#endif - MODULE_AUTHOR("Takashi Iwai "); -MODULE_DESCRIPTION("Digigram " CARD_NAME); +MODULE_DESCRIPTION("Digigram VXPocket"); MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}"); +MODULE_SUPPORTED_DEVICE("{{Digigram,VXPocket},{Digigram,VXPocket440}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -46,82 +42,405 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ static int ibl[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +MODULE_PARM_DESC(index, "Index value for VXPocket soundcard."); module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +MODULE_PARM_DESC(id, "ID string for VXPocket soundcard."); module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); +MODULE_PARM_DESC(enable, "Enable VXPocket soundcard."); module_param_array(ibl, int, NULL, 0444); -MODULE_PARM_DESC(ibl, "Capture IBL size for " CARD_NAME " soundcard."); +MODULE_PARM_DESC(ibl, "Capture IBL size for VXPocket soundcard."); /* */ -#ifdef COMPILE_VXP440 +static unsigned int card_alloc; +static dev_link_t *dev_list; /* Linked list of devices */ +static dev_info_t dev_info = "snd-vxpocket"; -/* 1 DSP, 1 sync UER, 1 sync World Clock (NIY) */ -/* SMPTE (NIY) */ -/* 2 stereo analog input (line/micro) */ -/* 2 stereo analog output */ -/* Only output levels can be modified */ -/* UER, but only for the first two inputs and outputs. */ -#define NUM_CODECS 2 -#define CARD_TYPE VX_TYPE_VXP440 -#define DEV_INFO "snd-vxp440" +static int vxpocket_event(event_t event, int priority, event_callback_args_t *args); -#else -/* 1 DSP, 1 sync UER */ -/* 1 programmable clock (NIY) */ -/* 1 stereo analog input (line/micro) */ -/* 1 stereo analog output */ -/* Only output levels can be modified */ +/* + */ +static void vxpocket_release(dev_link_t *link) +{ + if (link->state & DEV_CONFIG) { + /* release cs resources */ + pcmcia_release_configuration(link->handle); + pcmcia_release_io(link->handle, &link->io); + pcmcia_release_irq(link->handle, &link->irq); + link->state &= ~DEV_CONFIG; + } + if (link->handle) { + /* Break the link with Card Services */ + pcmcia_deregister_client(link->handle); + link->handle = NULL; + } +} -#define NUM_CODECS 1 -#define CARD_TYPE VX_TYPE_VXPOCKET -#define DEV_INFO "snd-vxpocket" +/* + * destructor, called from snd_card_free_in_thread() + */ +static int snd_vxpocket_dev_free(snd_device_t *device) +{ + vx_core_t *chip = device->device_data; -#endif + snd_vx_free_firmware(chip); + kfree(chip); + return 0; +} -static dev_info_t dev_info = DEV_INFO; -static struct snd_vx_hardware vxp_hw = { - .name = CARD_NAME, - .type = CARD_TYPE, +/* + * Hardware information + */ + +/* VX-pocket V2 + * + * 1 DSP, 1 sync UER + * 1 programmable clock (NIY) + * 1 stereo analog input (line/micro) + * 1 stereo analog output + * Only output levels can be modified + */ + +static struct snd_vx_hardware vxpocket_hw = { + .name = "VXPocket", + .type = VX_TYPE_VXPOCKET, /* hardware specs */ - .num_codecs = NUM_CODECS, - .num_ins = NUM_CODECS, - .num_outs = NUM_CODECS, + .num_codecs = 1, + .num_ins = 1, + .num_outs = 1, .output_level_max = VX_ANALOG_OUT_LEVEL_MAX, }; -static struct snd_vxp_entry hw_entry = { - .dev_info = &dev_info, +/* VX-pocket 440 + * + * 1 DSP, 1 sync UER, 1 sync World Clock (NIY) + * SMPTE (NIY) + * 2 stereo analog input (line/micro) + * 2 stereo analog output + * Only output levels can be modified + * UER, but only for the first two inputs and outputs. + */ - /* module parameters */ - .index_table = index, - .id_table = id, - .enable_table = enable, - .ibl = ibl, +static struct snd_vx_hardware vxp440_hw = { + .name = "VXPocket440", + .type = VX_TYPE_VXP440, + + /* hardware specs */ + .num_codecs = 2, + .num_ins = 2, + .num_outs = 2, + .output_level_max = VX_ANALOG_OUT_LEVEL_MAX, +}; + + +/* + * create vxpocket instance + */ +static struct snd_vxpocket *snd_vxpocket_new(snd_card_t *card, int ibl) +{ + client_reg_t client_reg; /* Register with cardmgr */ + dev_link_t *link; /* Info for cardmgr */ + vx_core_t *chip; + struct snd_vxpocket *vxp; + int ret; + static snd_device_ops_t ops = { + .dev_free = snd_vxpocket_dev_free, + }; + + chip = snd_vx_create(card, &vxpocket_hw, &snd_vxpocket_ops, + sizeof(struct snd_vxpocket) - sizeof(vx_core_t)); + if (! chip) + return NULL; + + if (snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops) < 0) { + kfree(chip); + return NULL; + } + chip->ibl.size = ibl; + + vxp = (struct snd_vxpocket *)chip; + + link = &vxp->link; + link->priv = chip; + + link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; + link->io.NumPorts1 = 16; + + link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT; + + link->irq.IRQInfo1 = IRQ_LEVEL_ID; + link->irq.Handler = &snd_vx_irq_handler; + link->irq.Instance = chip; + + link->conf.Attributes = CONF_ENABLE_IRQ; + link->conf.Vcc = 50; + link->conf.IntType = INT_MEMORY_AND_IO; + link->conf.ConfigIndex = 1; + link->conf.Present = PRESENT_OPTION; + + /* Register with Card Services */ + memset(&client_reg, 0, sizeof(client_reg)); + client_reg.dev_info = &dev_info; + client_reg.EventMask = + CS_EVENT_CARD_INSERTION | CS_EVENT_CARD_REMOVAL +#ifdef CONFIG_PM + | CS_EVENT_RESET_PHYSICAL | CS_EVENT_CARD_RESET + | CS_EVENT_PM_SUSPEND | CS_EVENT_PM_RESUME +#endif + ; + client_reg.event_handler = &vxpocket_event; + client_reg.Version = 0x0210; + client_reg.event_callback_args.client_data = link; + + ret = pcmcia_register_client(&link->handle, &client_reg); + if (ret != CS_SUCCESS) { + cs_error(link->handle, RegisterClient, ret); + return NULL; + } + + return vxp; +} + + +/** + * snd_vxpocket_assign_resources - initialize the hardware and card instance. + * @port: i/o port for the card + * @irq: irq number for the card + * + * this function assigns the specified port and irq, boot the card, + * create pcm and control instances, and initialize the rest hardware. + * + * returns 0 if successful, or a negative error code. + */ +static int snd_vxpocket_assign_resources(vx_core_t *chip, int port, int irq) +{ + int err; + snd_card_t *card = chip->card; + struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; + + snd_printdd(KERN_DEBUG "vxpocket assign resources: port = 0x%x, irq = %d\n", port, irq); + vxp->port = port; + + sprintf(card->shortname, "Digigram %s", card->driver); + sprintf(card->longname, "%s at 0x%x, irq %i", + card->shortname, port, irq); + + chip->irq = irq; + + if ((err = snd_vx_setup_firmware(chip)) < 0) + return err; + + return 0; +} + + +/* + * configuration callback + */ + +#define CS_CHECK(fn, ret) \ +do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) + +static void vxpocket_config(dev_link_t *link) +{ + client_handle_t handle = link->handle; + vx_core_t *chip = link->priv; + struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; + tuple_t tuple; + cisparse_t *parse; + u_short buf[32]; + int last_fn, last_ret; + + snd_printdd(KERN_DEBUG "vxpocket_config called\n"); + parse = kmalloc(sizeof(*parse), GFP_KERNEL); + if (! parse) { + snd_printk(KERN_ERR "vx: cannot allocate\n"); + return; + } + tuple.Attributes = 0; + tuple.TupleData = (cisdata_t *)buf; + tuple.TupleDataMax = sizeof(buf); + tuple.TupleOffset = 0; + tuple.DesiredTuple = CISTPL_CONFIG; + CS_CHECK(GetFirstTuple, pcmcia_get_first_tuple(handle, &tuple)); + CS_CHECK(GetTupleData, pcmcia_get_tuple_data(handle, &tuple)); + CS_CHECK(ParseTuple, pcmcia_parse_tuple(handle, &tuple, parse)); + link->conf.ConfigBase = parse->config.base; + link->conf.Present = parse->config.rmask[0]; + + /* redefine hardware record according to the VERSION1 string */ + tuple.DesiredTuple = CISTPL_VERS_1; + CS_CHECK(GetFirstTuple, pcmcia_get_first_tuple(handle, &tuple)); + CS_CHECK(GetTupleData, pcmcia_get_tuple_data(handle, &tuple)); + CS_CHECK(ParseTuple, pcmcia_parse_tuple(handle, &tuple, parse)); + if (! strcmp(parse->version_1.str + parse->version_1.ofs[1], "VX-POCKET")) { + snd_printdd("VX-pocket is detected\n"); + } else { + snd_printdd("VX-pocket 440 is detected\n"); + /* overwrite the hardware information */ + chip->hw = &vxp440_hw; + chip->type = vxp440_hw.type; + strcpy(chip->card->driver, vxp440_hw.name); + } + + /* Configure card */ + link->state |= DEV_CONFIG; + + CS_CHECK(RequestIO, pcmcia_request_io(handle, &link->io)); + CS_CHECK(RequestIRQ, pcmcia_request_irq(link->handle, &link->irq)); + CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link->handle, &link->conf)); + + chip->dev = &handle_to_dev(link->handle); + + if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq.AssignedIRQ) < 0) + goto failed; + + link->dev = &vxp->node; + link->state &= ~DEV_CONFIG_PENDING; + kfree(parse); + return; + +cs_failed: + cs_error(link->handle, last_fn, last_ret); +failed: + pcmcia_release_configuration(link->handle); + pcmcia_release_io(link->handle, &link->io); + pcmcia_release_irq(link->handle, &link->irq); + link->state &= ~DEV_CONFIG; + kfree(parse); +} + + +/* + * event callback + */ +static int vxpocket_event(event_t event, int priority, event_callback_args_t *args) +{ + dev_link_t *link = args->client_data; + vx_core_t *chip = link->priv; + + switch (event) { + case CS_EVENT_CARD_REMOVAL: + snd_printdd(KERN_DEBUG "CARD_REMOVAL..\n"); + link->state &= ~DEV_PRESENT; + if (link->state & DEV_CONFIG) + chip->chip_status |= VX_STAT_IS_STALE; + break; + case CS_EVENT_CARD_INSERTION: + snd_printdd(KERN_DEBUG "CARD_INSERTION..\n"); + link->state |= DEV_PRESENT | DEV_CONFIG_PENDING; + vxpocket_config(link); + break; +#ifdef CONFIG_PM + case CS_EVENT_PM_SUSPEND: + snd_printdd(KERN_DEBUG "SUSPEND\n"); + link->state |= DEV_SUSPEND; + if (chip && chip->card->pm_suspend) { + snd_printdd(KERN_DEBUG "snd_vx_suspend calling\n"); + chip->card->pm_suspend(chip->card, PMSG_SUSPEND); + } + /* Fall through... */ + case CS_EVENT_RESET_PHYSICAL: + snd_printdd(KERN_DEBUG "RESET_PHYSICAL\n"); + if (link->state & DEV_CONFIG) + pcmcia_release_configuration(link->handle); + break; + case CS_EVENT_PM_RESUME: + snd_printdd(KERN_DEBUG "RESUME\n"); + link->state &= ~DEV_SUSPEND; + /* Fall through... */ + case CS_EVENT_CARD_RESET: + snd_printdd(KERN_DEBUG "CARD_RESET\n"); + if (DEV_OK(link)) { + //struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; + snd_printdd(KERN_DEBUG "requestconfig...\n"); + pcmcia_request_configuration(link->handle, &link->conf); + if (chip && chip->card->pm_resume) { + snd_printdd(KERN_DEBUG "calling snd_vx_resume\n"); + chip->card->pm_resume(chip->card); + } + } + snd_printdd(KERN_DEBUG "resume done!\n"); + break; +#endif + } + return 0; +} - /* h/w config */ - .hardware = &vxp_hw, - .ops = &snd_vxpocket_ops, -}; /* */ static dev_link_t *vxp_attach(void) { - return snd_vxpocket_attach(&hw_entry); + snd_card_t *card; + struct snd_vxpocket *vxp; + int i; + + /* find an empty slot from the card list */ + for (i = 0; i < SNDRV_CARDS; i++) { + if (! card_alloc & (1 << i)) + break; + } + if (i >= SNDRV_CARDS) { + snd_printk(KERN_ERR "vxpocket: too many cards found\n"); + return NULL; + } + if (! enable[i]) + return NULL; /* disabled explicitly */ + + /* ok, create a card instance */ + card = snd_card_new(index[i], id[i], THIS_MODULE, 0); + if (card == NULL) { + snd_printk(KERN_ERR "vxpocket: cannot create a card instance\n"); + return NULL; + } + + vxp = snd_vxpocket_new(card, ibl[i]); + if (! vxp) { + snd_card_free(card); + return NULL; + } + + vxp->index = index[i]; + card_alloc |= 1 << i; + + /* Chain drivers */ + vxp->link.next = dev_list; + dev_list = &vxp->link; + + return &vxp->link; } static void vxp_detach(dev_link_t *link) { - snd_vxpocket_detach(&hw_entry, link); + struct snd_vxpocket *vxp; + vx_core_t *chip; + dev_link_t **linkp; + + if (! link) + return; + + vxp = link->priv; + chip = (vx_core_t *)vxp; + card_alloc &= ~(1 << vxp->index); + + /* Remove the interface data from the linked list */ + for (linkp = &dev_list; *linkp; linkp = &(*linkp)->next) + if (*linkp == link) { + *linkp = link->next; + break; + } + + chip->chip_status |= VX_STAT_IS_STALE; /* to be sure */ + snd_card_disconnect(chip->card); + vxpocket_release(link); + snd_card_free_in_thread(chip->card); } /* @@ -137,7 +456,7 @@ MODULE_DEVICE_TABLE(pcmcia, vxp_ids); static struct pcmcia_driver vxp_cs_driver = { .owner = THIS_MODULE, .drv = { - .name = DEV_INFO, + .name = "snd-vxpocket", }, .attach = vxp_attach, .detach = vxp_detach, @@ -152,7 +471,7 @@ static int __init init_vxpocket(void) static void __exit exit_vxpocket(void) { pcmcia_unregister_driver(&vxp_cs_driver); - BUG_ON(hw_entry.dev_list != NULL); + BUG_ON(dev_list != NULL); } module_init(init_vxpocket); diff --git a/sound/pcmcia/vx/vxpocket.h b/sound/pcmcia/vx/vxpocket.h index 4462c04a4e8f..70754aa3dd11 100644 --- a/sound/pcmcia/vx/vxpocket.h +++ b/sound/pcmcia/vx/vxpocket.h @@ -28,24 +28,6 @@ #include #include -struct snd_vxp_entry { - dev_info_t *dev_info; - - /* module parameters */ - int *index_table; - char **id_table; - int *enable_table; - int *ibl; - - /* h/w config */ - struct snd_vx_hardware *hardware; - struct snd_vx_ops *ops; - - /* slots */ - vx_core_t *card_list[SNDRV_CARDS]; - dev_link_t *dev_list; /* Linked list of devices */ -}; - struct snd_vxpocket { vx_core_t core; @@ -57,8 +39,7 @@ struct snd_vxpocket { unsigned int regCDSP; /* current CDSP register */ unsigned int regDIALOG; /* current DIALOG register */ - int index; - struct snd_vxp_entry *hw_entry; + int index; /* card index */ /* pcmcia stuff */ dev_link_t link; @@ -70,12 +51,6 @@ extern struct snd_vx_ops snd_vxpocket_ops; void vx_set_mic_boost(vx_core_t *chip, int boost); void vx_set_mic_level(vx_core_t *chip, int level); -/* - * pcmcia stuff - */ -dev_link_t *snd_vxpocket_attach(struct snd_vxp_entry *hw); -void snd_vxpocket_detach(struct snd_vxp_entry *hw, dev_link_t *link); - int vxp_add_mic_controls(vx_core_t *chip); /* Constants used to access the CDSP register (0x08). */ -- cgit v1.2.3 From 174d52204500a63c19e6613e49db3fcfe4e9d35a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 13:42:39 +0200 Subject: [ALSA] vxpocket - Remove unused code Digigram VX Pocket driver Removed unused files (dropped by the last change). Signed-off-by: Takashi Iwai --- sound/pcmcia/vx/vx_entry.c | 375 --------------------------------------------- sound/pcmcia/vx/vxp440.c | 14 -- 2 files changed, 389 deletions(-) delete mode 100644 sound/pcmcia/vx/vx_entry.c delete mode 100644 sound/pcmcia/vx/vxp440.c (limited to 'sound') diff --git a/sound/pcmcia/vx/vx_entry.c b/sound/pcmcia/vx/vx_entry.c deleted file mode 100644 index df7a39ba9680..000000000000 --- a/sound/pcmcia/vx/vx_entry.c +++ /dev/null @@ -1,375 +0,0 @@ -/* - * Driver for Digigram VXpocket soundcards - * - * PCMCIA entry part - * - * Copyright (c) 2002 by Takashi Iwai - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * 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 -#include -#include "vxpocket.h" -#include -#include - - -MODULE_AUTHOR("Takashi Iwai "); -MODULE_DESCRIPTION("Common routines for Digigram PCMCIA VX drivers"); -MODULE_LICENSE("GPL"); - -/* - * prototypes - */ -static void vxpocket_config(dev_link_t *link); - - -static void vxpocket_release(dev_link_t *link) -{ - if (link->state & DEV_CONFIG) { - /* release cs resources */ - pcmcia_release_configuration(link->handle); - pcmcia_release_io(link->handle, &link->io); - pcmcia_release_irq(link->handle, &link->irq); - link->state &= ~DEV_CONFIG; - } -} - -/* - * destructor - */ -static int snd_vxpocket_free(vx_core_t *chip) -{ - struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - struct snd_vxp_entry *hw; - dev_link_t *link = &vxp->link; - - vxpocket_release(link); - - /* Break the link with Card Services */ - if (link->handle) - pcmcia_deregister_client(link->handle); - - hw = vxp->hw_entry; - if (hw) - hw->card_list[vxp->index] = NULL; - chip->card = NULL; - kfree(chip->dev); - - snd_vx_free_firmware(chip); - kfree(chip); - return 0; -} - -static int snd_vxpocket_dev_free(snd_device_t *device) -{ - vx_core_t *chip = device->device_data; - return snd_vxpocket_free(chip); -} - -/* - * snd_vxpocket_attach - attach callback for cs - * @hw: the hardware information - */ -dev_link_t *snd_vxpocket_attach(struct snd_vxp_entry *hw) -{ - client_reg_t client_reg; /* Register with cardmgr */ - dev_link_t *link; /* Info for cardmgr */ - int i, ret; - vx_core_t *chip; - struct snd_vxpocket *vxp; - snd_card_t *card; - static snd_device_ops_t ops = { - .dev_free = snd_vxpocket_dev_free, - }; - - snd_printdd(KERN_DEBUG "vxpocket_attach called\n"); - /* find an empty slot from the card list */ - for (i = 0; i < SNDRV_CARDS; i++) { - if (! hw->card_list[i]) - break; - } - if (i >= SNDRV_CARDS) { - snd_printk(KERN_ERR "vxpocket: too many cards found\n"); - return NULL; - } - if (! hw->enable_table[i]) - return NULL; /* disabled explicitly */ - - /* ok, create a card instance */ - card = snd_card_new(hw->index_table[i], hw->id_table[i], THIS_MODULE, 0); - if (card == NULL) { - snd_printk(KERN_ERR "vxpocket: cannot create a card instance\n"); - return NULL; - } - - chip = snd_vx_create(card, hw->hardware, hw->ops, - sizeof(struct snd_vxpocket) - sizeof(vx_core_t)); - if (! chip) - return NULL; - -#ifdef SND_VX_FW_LOADER - /* fake a device here since pcmcia doesn't give a valid device... */ - chip->dev = kcalloc(1, sizeof(*chip->dev), GFP_KERNEL); - if (! chip->dev) { - snd_printk(KERN_ERR "vxp: can't malloc chip->dev\n"); - kfree(chip); - snd_card_free(card); - return NULL; - } - device_initialize(chip->dev); - sprintf(chip->dev->bus_id, "vxpocket%d", i); -#endif - - if (snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops) < 0) { - kfree(chip); - snd_card_free(card); - return NULL; - } - - vxp = (struct snd_vxpocket *)chip; - vxp->index = i; - vxp->hw_entry = hw; - chip->ibl.size = hw->ibl[i]; - hw->card_list[i] = chip; - - link = &vxp->link; - link->priv = chip; - - link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; - link->io.NumPorts1 = 16; - - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT; - // link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING|IRQ_FIRST_SHARED; - - link->irq.IRQInfo1 = IRQ_LEVEL_ID; - link->irq.Handler = &snd_vx_irq_handler; - link->irq.Instance = chip; - - link->conf.Attributes = CONF_ENABLE_IRQ; - link->conf.Vcc = 50; - link->conf.IntType = INT_MEMORY_AND_IO; - link->conf.ConfigIndex = 1; - link->conf.Present = PRESENT_OPTION; - - /* Register with Card Services */ - memset(&client_reg, 0, sizeof(client_reg)); - client_reg.dev_info = hw->dev_info; - client_reg.Version = 0x0210; - client_reg.event_callback_args.client_data = link; - - ret = pcmcia_register_client(&link->handle, &client_reg); - if (ret != CS_SUCCESS) { - cs_error(link->handle, RegisterClient, ret); - snd_card_free(card); - return NULL; - } - - /* Chain drivers */ - link->next = hw->dev_list; - hw->dev_list = link; - - /* snd_card_set_pm_callback(card, snd_vxpocket_suspend, snd_vxpocket_resume, chip); */ - - return link; -} - - -/** - * snd_vxpocket_assign_resources - initialize the hardware and card instance. - * @port: i/o port for the card - * @irq: irq number for the card - * - * this function assigns the specified port and irq, boot the card, - * create pcm and control instances, and initialize the rest hardware. - * - * returns 0 if successful, or a negative error code. - */ -static int snd_vxpocket_assign_resources(vx_core_t *chip, int port, int irq) -{ - int err; - snd_card_t *card = chip->card; - struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - - snd_printdd(KERN_DEBUG "vxpocket assign resources: port = 0x%x, irq = %d\n", port, irq); - vxp->port = port; - - sprintf(card->shortname, "Digigram %s", card->driver); - sprintf(card->longname, "%s at 0x%x, irq %i", - card->shortname, port, irq); - - chip->irq = irq; - - if ((err = snd_vx_setup_firmware(chip)) < 0) - return err; - - return 0; -} - - -/* - * snd_vxpocket_detach - detach callback for cs - * @hw: the hardware information - */ -void snd_vxpocket_detach(struct snd_vxp_entry *hw, dev_link_t *link) -{ - vx_core_t *chip; - - if (! link) - return; - - chip = link->priv; - - snd_printdd(KERN_DEBUG "vxpocket_detach called\n"); - /* Remove the interface data from the linked list */ - if (hw) { - dev_link_t **linkp; - /* Locate device structure */ - for (linkp = &hw->dev_list; *linkp; linkp = &(*linkp)->next) - if (*linkp == link) { - *linkp = link->next; - break; - } - } - chip->chip_status |= VX_STAT_IS_STALE; /* to be sure */ - snd_card_disconnect(chip->card); - snd_card_free_in_thread(chip->card); -} - -/* - * configuration callback - */ - -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - -static void vxpocket_config(dev_link_t *link) -{ - client_handle_t handle = link->handle; - vx_core_t *chip = link->priv; - struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - tuple_t tuple; - cisparse_t *parse = NULL; - u_short buf[32]; - int last_fn, last_ret; - - snd_printdd(KERN_DEBUG "vxpocket_config called\n"); - parse = kmalloc(sizeof(*parse), GFP_KERNEL); - if (! parse) { - snd_printk(KERN_ERR "vx: cannot allocate\n"); - return; - } - tuple.Attributes = 0; - tuple.TupleData = (cisdata_t *)buf; - tuple.TupleDataMax = sizeof(buf); - tuple.TupleOffset = 0; - tuple.DesiredTuple = CISTPL_CONFIG; - CS_CHECK(GetFirstTuple, pcmcia_get_first_tuple(handle, &tuple)); - CS_CHECK(GetTupleData, pcmcia_get_tuple_data(handle, &tuple)); - CS_CHECK(ParseTuple, pcmcia_parse_tuple(handle, &tuple, parse)); - link->conf.ConfigBase = parse->config.base; - link->conf.Present = parse->config.rmask[0]; - - /* Configure card */ - link->state |= DEV_CONFIG; - - CS_CHECK(RequestIO, pcmcia_request_io(handle, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link->handle, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link->handle, &link->conf)); - - if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq.AssignedIRQ) < 0) - goto failed; - - link->dev = &vxp->node; - link->state &= ~DEV_CONFIG_PENDING; - kfree(parse); - return; - -cs_failed: - cs_error(link->handle, last_fn, last_ret); -failed: - pcmcia_release_configuration(link->handle); - pcmcia_release_io(link->handle, &link->io); - pcmcia_release_irq(link->handle, &link->irq); - link->state &= ~DEV_CONFIG; - kfree(parse); -} - - -/* - * event callback - */ -int vxpocket_event(event_t event, int priority, event_callback_args_t *args) -{ - dev_link_t *link = args->client_data; - vx_core_t *chip = link->priv; - - switch (event) { - case CS_EVENT_CARD_REMOVAL: - snd_printdd(KERN_DEBUG "CARD_REMOVAL..\n"); - link->state &= ~DEV_PRESENT; - if (link->state & DEV_CONFIG) { - chip->chip_status |= VX_STAT_IS_STALE; - } - break; - case CS_EVENT_CARD_INSERTION: - snd_printdd(KERN_DEBUG "CARD_INSERTION..\n"); - link->state |= DEV_PRESENT | DEV_CONFIG_PENDING; - vxpocket_config(link); - break; -#ifdef CONFIG_PM - case CS_EVENT_PM_SUSPEND: - snd_printdd(KERN_DEBUG "SUSPEND\n"); - link->state |= DEV_SUSPEND; - if (chip && chip->card->pm_suspend) { - snd_printdd(KERN_DEBUG "snd_vx_suspend calling\n"); - chip->card->pm_suspend(chip->card, PMSG_SUSPEND); - } - /* Fall through... */ - case CS_EVENT_RESET_PHYSICAL: - snd_printdd(KERN_DEBUG "RESET_PHYSICAL\n"); - if (link->state & DEV_CONFIG) - pcmcia_release_configuration(link->handle); - break; - case CS_EVENT_PM_RESUME: - snd_printdd(KERN_DEBUG "RESUME\n"); - link->state &= ~DEV_SUSPEND; - /* Fall through... */ - case CS_EVENT_CARD_RESET: - snd_printdd(KERN_DEBUG "CARD_RESET\n"); - if (DEV_OK(link)) { - //struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - snd_printdd(KERN_DEBUG "requestconfig...\n"); - pcmcia_request_configuration(link->handle, &link->conf); - if (chip && chip->card->pm_resume) { - snd_printdd(KERN_DEBUG "calling snd_vx_resume\n"); - chip->card->pm_resume(chip->card); - } - } - snd_printdd(KERN_DEBUG "resume done!\n"); - break; -#endif - } - return 0; -} - -/* - * exported stuffs - */ -EXPORT_SYMBOL(snd_vxpocket_ops); -EXPORT_SYMBOL(snd_vxpocket_attach); -EXPORT_SYMBOL(vxpocket_event); -EXPORT_SYMBOL(snd_vxpocket_detach); diff --git a/sound/pcmcia/vx/vxp440.c b/sound/pcmcia/vx/vxp440.c deleted file mode 100644 index 59190a833001..000000000000 --- a/sound/pcmcia/vx/vxp440.c +++ /dev/null @@ -1,14 +0,0 @@ -#define COMPILE_VXP440 - -/* - add the following as /etc/pcmcia/vxp440.conf: - - device "snd-vxp440" - class "audio" module "snd-vxp440" - - card "Digigram VX-POCKET440" - manfid 0x01f1, 0x0100 - bind "snd-vxp440" -*/ - -#include "vxpocket.c" -- cgit v1.2.3 From 2eff7ec81eb586076974cb0918dffc5f4ad763d5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 13:45:20 +0200 Subject: [ALSA] cmipci - Add Mic Boost capture switch CMIPCI driver Added 'Mic Boost Capture Switch' and 'Phone' switches. The existing playback switch is renamed as 'Mic Boost Playback Switch'. Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 4725b4a010be..f5a4ac1ceef9 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -306,7 +306,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_REG_FM_PCI 0x50 /* - * for CMI-8338 .. this is not valid for CMI-8738. + * access from SB-mixer port */ #define CM_REG_EXTENT_IND 0xf0 #define CM_VPHONE_MASK 0xe0 /* Phone volume control (0-3) << 5 */ @@ -315,6 +315,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_VSPKM 0x08 /* Speaker mute control, default high */ #define CM_RLOOPREN 0x04 /* Rec. R-channel enable */ #define CM_RLOOPLEN 0x02 /* Rec. L-channel enable */ +#define CM_VADMIC3 0x01 /* Mic record boost */ /* * CMI-8338 spec ver 0.5 (this is not valid for CMI-8738): @@ -2135,8 +2136,12 @@ static snd_kcontrol_new_t snd_cmipci_mixers[] __devinitdata = { CMIPCI_MIXER_VOL_STEREO("Aux Playback Volume", CM_REG_AUX_VOL, 4, 0, 15), CMIPCI_MIXER_SW_STEREO("Aux Playback Switch", CM_REG_MIXER2, CM_VAUXLM_SHIFT, CM_VAUXRM_SHIFT, 0), CMIPCI_MIXER_SW_STEREO("Aux Capture Switch", CM_REG_MIXER2, CM_RAUXLEN_SHIFT, CM_RAUXREN_SHIFT, 0), - CMIPCI_MIXER_SW_MONO("Mic Boost", CM_REG_MIXER2, CM_MICGAINZ_SHIFT, 1), + CMIPCI_MIXER_SW_MONO("Mic Boost Playback Switch", CM_REG_MIXER2, CM_MICGAINZ_SHIFT, 1), CMIPCI_MIXER_VOL_MONO("Mic Capture Volume", CM_REG_MIXER2, CM_VADMIC_SHIFT, 7), + CMIPCI_SB_VOL_MONO("Phone Playback Volume", CM_REG_EXTENT_IND, 5, 7), + CMIPCI_DOUBLE("Phone Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 4, 4, 1, 0, 0), + CMIPCI_DOUBLE("PC Speaker Playnack Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0), + CMIPCI_DOUBLE("Mic Boost Capture Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 0, 0, 1, 0, 0), }; /* -- cgit v1.2.3 From 52b723888c1a55d34551f9b0b9d9296e0e3e8d3c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 13:47:06 +0200 Subject: [ALSA] Fix resume of intel8x0 Intel8x0 driver,AC97 Codec Fix resume of intel8x0 driver. The ac97 codec didn't restore some registers properly, and the restore of ICH4 SPDIF and SDIN settings was missing. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 2 ++ sound/pci/intel8x0.c | 13 +++++++++++++ 2 files changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index a4b72cd2eea0..0677d41239a9 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -367,6 +367,7 @@ int snd_ac97_update(ac97_t *ac97, unsigned short reg, unsigned short value) ac97->regs[reg] = value; ac97->bus->ops->write(ac97, reg, value); } + set_bit(reg, ac97->reg_accessed); up(&ac97->reg_mutex); return change; } @@ -410,6 +411,7 @@ int snd_ac97_update_bits_nolock(ac97_t *ac97, unsigned short reg, ac97->regs[reg] = new; ac97->bus->ops->write(ac97, reg, new); } + set_bit(reg, ac97->reg_accessed); return change; } diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index cc16f95f9cef..c3c3b68b4540 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -424,6 +424,7 @@ struct _snd_intel8x0 { unsigned xbox: 1; /* workaround for Xbox AC'97 detection */ int spdif_idx; /* SPDIF BAR index; *_SPBAR or -1 if use PCMOUT */ + unsigned int sdm_saved; /* SDM reg value */ ac97_bus_t *ac97_bus; ac97_t *ac97[3]; @@ -2373,6 +2374,8 @@ static int intel8x0_suspend(snd_card_t *card, pm_message_t state) for (i = 0; i < 3; i++) if (chip->ac97[i]) snd_ac97_suspend(chip->ac97[i]); + if (chip->device_type == DEVICE_INTEL_ICH4) + chip->sdm_saved = igetbyte(chip, ICHREG(SDM)); pci_disable_device(chip->pci); return 0; } @@ -2386,6 +2389,16 @@ static int intel8x0_resume(snd_card_t *card) pci_set_master(chip->pci); snd_intel8x0_chip_init(chip, 0); + /* re-initialize mixer stuff */ + if (chip->device_type == DEVICE_INTEL_ICH4) { + /* enable separate SDINs for ICH4 */ + iputbyte(chip, ICHREG(SDM), chip->sdm_saved); + /* use slot 10/11 for SPDIF */ + iputdword(chip, ICHREG(GLOB_CNT), + (igetdword(chip, ICHREG(GLOB_CNT)) & ~ICH_PCM_SPDIF_MASK) | + ICH_PCM_SPDIF_1011); + } + /* refill nocache */ if (chip->fix_nocache) fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1); -- cgit v1.2.3 From 5ba1e7b594db4d0e1f88ace87c1cb295761ca5c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 13:47:58 +0200 Subject: [ALSA] maestro3 - Clean up Maestro3 driver - Clean up maestro3 code - Use msleep() - Don't enable hw-vol irq when not defined Signed-off-by: Takashi Iwai --- sound/pci/maestro3.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 52c585901c54..39b5e7db1543 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -1050,11 +1050,6 @@ static struct m3_hv_quirk m3_hv_quirk_list[] = { * lowlevel functions */ -#define big_mdelay(msec) do {\ - set_current_state(TASK_UNINTERRUPTIBLE);\ - schedule_timeout(((msec) * HZ) / 1000);\ -} while (0) - static inline void snd_m3_outw(m3_t *chip, u16 value, unsigned long reg) { outw(value, chip->iobase + reg); @@ -1096,7 +1091,7 @@ static void snd_m3_assp_write(m3_t *chip, u16 region, u16 index, u16 data) static void snd_m3_assp_halt(m3_t *chip) { chip->reset_state = snd_m3_inb(chip, DSP_PORT_CONTROL_REG_B) & ~REGB_STOP_CLOCK; - big_mdelay(10); + msleep(10); snd_m3_outb(chip, chip->reset_state & ~REGB_ENABLE_RESET, DSP_PORT_CONTROL_REG_B); } @@ -2108,9 +2103,9 @@ static void snd_m3_ac97_reset(m3_t *chip) */ tmp = inw(io + RING_BUS_CTRL_A); outw(RAC_SDFS_ENABLE|LAC_SDFS_ENABLE, io + RING_BUS_CTRL_A); - big_mdelay(20); + msleep(20); outw(tmp, io + RING_BUS_CTRL_A); - big_mdelay(50); + msleep(50); #endif } @@ -2525,9 +2520,13 @@ static void snd_m3_enable_ints(m3_t *chip) { unsigned long io = chip->iobase; + unsigned short val; /* TODO: MPU401 not supported yet */ - outw(ASSP_INT_ENABLE | HV_INT_ENABLE /*| MPU401_INT_ENABLE*/, io + HOST_INT_CTRL); + val = ASSP_INT_ENABLE /*| MPU401_INT_ENABLE*/; + if (chip->hv_quirk && (chip->hv_quirk->config & HV_CTRL_ENABLE)) + val |= HV_INT_ENABLE; + outw(val, io + HOST_INT_CTRL); outb(inb(io + ASSP_CONTROL_C) | ASSP_HOST_INT_ENABLE, io + ASSP_CONTROL_C); } @@ -2589,7 +2588,7 @@ static int m3_suspend(snd_card_t *card, pm_message_t state) snd_pcm_suspend_all(chip->pcm); snd_ac97_suspend(chip->ac97); - big_mdelay(10); /* give the assp a chance to idle.. */ + msleep(10); /* give the assp a chance to idle.. */ snd_m3_assp_halt(chip); @@ -2697,6 +2696,8 @@ snd_m3_create(snd_card_t *card, struct pci_dev *pci, } spin_lock_init(&chip->reg_lock); + spin_lock_init(&chip->ac97_lock); + switch (pci->device) { case PCI_DEVICE_ID_ESS_ALLEGRO: case PCI_DEVICE_ID_ESS_ALLEGRO_1: @@ -2765,6 +2766,8 @@ snd_m3_create(snd_card_t *card, struct pci_dev *pci, snd_m3_assp_init(chip); snd_m3_amp_enable(chip, 1); + tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); + if (request_irq(pci->irq, snd_m3_interrupt, SA_INTERRUPT|SA_SHIRQ, card->driver, (void *)chip)) { snd_printk("unable to grab IRQ %d\n", pci->irq); @@ -2786,9 +2789,6 @@ snd_m3_create(snd_card_t *card, struct pci_dev *pci, return err; } - spin_lock_init(&chip->ac97_lock); - tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); - if ((err = snd_m3_mixer(chip)) < 0) return err; -- cgit v1.2.3 From ab79509a95b1d22c40d4a87823b6a48bc9a12af5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 18:13:54 +0200 Subject: [ALSA] Add help texts to Kconfig SPARC Added simple help texts to Kconfig of Sparc sound drivers. (Better texts are appreciated :) Signed-off-by: Takashi Iwai --- sound/sparc/Kconfig | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/sparc/Kconfig b/sound/sparc/Kconfig index 2358df1c45a9..f0fce1daf67d 100644 --- a/sound/sparc/Kconfig +++ b/sound/sparc/Kconfig @@ -7,12 +7,22 @@ config SND_SUN_AMD7930 tristate "Sun AMD7930" depends on SBUS && SND select SND_PCM + help + Say Y here to include support for AMD7930 sound device on Sun. + + To compile this driver as a module, choose M here: the module + will be called snd-sun-amd7930. # dep_tristate 'Sun DBRI' CONFIG_SND_SUN_DBRI $CONFIG_SND config SND_SUN_CS4231 tristate "Sun CS4231" depends on SND select SND_PCM + help + Say Y here to include support for CS4231 sound device on Sun. + + To compile this driver as a module, choose M here: the module + will be called snd-sun-cs4231. endmenu -- cgit v1.2.3 From 1bd9debf25b8a5f5029d7619f43e4a9a775973d3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 18:26:20 +0200 Subject: [ALSA] Add DBRI driver on Sparcs Documentation,SPARC,/sparc/Makefile Add the DBRI driver on Sparcs by Martin Habets (moved from alsa-driver tree). Signed-off-by: Takashi Iwai --- sound/sparc/Kconfig | 12 +- sound/sparc/Makefile | 4 +- sound/sparc/dbri.c | 2729 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 2741 insertions(+), 4 deletions(-) create mode 100644 sound/sparc/dbri.c (limited to 'sound') diff --git a/sound/sparc/Kconfig b/sound/sparc/Kconfig index f0fce1daf67d..25a8a558ef92 100644 --- a/sound/sparc/Kconfig +++ b/sound/sparc/Kconfig @@ -13,7 +13,6 @@ config SND_SUN_AMD7930 To compile this driver as a module, choose M here: the module will be called snd-sun-amd7930. -# dep_tristate 'Sun DBRI' CONFIG_SND_SUN_DBRI $CONFIG_SND config SND_SUN_CS4231 tristate "Sun CS4231" depends on SND @@ -24,5 +23,14 @@ config SND_SUN_CS4231 To compile this driver as a module, choose M here: the module will be called snd-sun-cs4231. -endmenu +config SND_SUN_DBRI + tristate "Sun DBRI" + depends on SND && SBUS + select SND_PCM + help + Say Y here to include support for DBRI sound device on Sun. + To compile this driver as a module, choose M here: the module + will be called snd-sun-dbri. + +endmenu diff --git a/sound/sparc/Makefile b/sound/sparc/Makefile index 6809cc92d276..3cd89c67c2f2 100644 --- a/sound/sparc/Makefile +++ b/sound/sparc/Makefile @@ -4,9 +4,9 @@ # snd-sun-amd7930-objs := amd7930.o -#snd-sun-dbri-objs := dbri.o snd-sun-cs4231-objs := cs4231.o +snd-sun-dbri-objs := dbri.o obj-$(CONFIG_SND_SUN_AMD7930) += snd-sun-amd7930.o -#obj-$(CONFIG_SND_SUN_DBRI) += snd-sun-dbri.o obj-$(CONFIG_SND_SUN_CS4231) += snd-sun-cs4231.o +obj-$(CONFIG_SND_SUN_DBRI) += snd-sun-dbri.o diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c new file mode 100644 index 000000000000..941c7b1e7ebb --- /dev/null +++ b/sound/sparc/dbri.c @@ -0,0 +1,2729 @@ +/* + * Driver for DBRI sound chip found on Sparcs. + * Copyright (C) 2004 Martin Habets (mhabets@users.sourceforge.net) + * + * Based entirely upon drivers/sbus/audio/dbri.c which is: + * Copyright (C) 1997 Rudolf Koenig (rfkoenig@immd4.informatik.uni-erlangen.de) + * Copyright (C) 1998, 1999 Brent Baccala (baccala@freesoft.org) + * + * This is the lowlevel driver for the DBRI & MMCODEC duo used for ISDN & AUDIO + * on Sun SPARCstation 10, 20, LX and Voyager models. + * + * - DBRI: AT&T T5900FX Dual Basic Rates ISDN Interface. It is a 32 channel + * data time multiplexer with ISDN support (aka T7259) + * Interfaces: SBus,ISDN NT & TE, CHI, 4 bits parallel. + * CHI: (spelled ki) Concentration Highway Interface (AT&T or Intel bus ?). + * Documentation: + * - "STP 4000SBus Dual Basic Rate ISDN (DBRI) Tranceiver" from + * Sparc Technology Business (courtesy of Sun Support) + * - Data sheet of the T7903, a newer but very similar ISA bus equivalent + * available from the Lucent (formarly AT&T microelectronics) home + * page. + * - http://www.freesoft.org/Linux/DBRI/ + * - MMCODEC: Crystal Semiconductor CS4215 16 bit Multimedia Audio Codec + * Interfaces: CHI, Audio In & Out, 2 bits parallel + * Documentation: from the Crystal Semiconductor home page. + * + * The DBRI is a 32 pipe machine, each pipe can transfer some bits between + * memory and a serial device (long pipes, nr 0-15) or between two serial + * devices (short pipes, nr 16-31), or simply send a fixed data to a serial + * device (short pipes). + * A timeslot defines the bit-offset and nr of bits read from a serial device. + * The timeslots are linked to 6 circular lists, one for each direction for + * each serial device (NT,TE,CHI). A timeslot is associated to 1 or 2 pipes + * (the second one is a monitor/tee pipe, valid only for serial input). + * + * The mmcodec is connected via the CHI bus and needs the data & some + * parameters (volume, balance, output selection) timemultiplexed in 8 byte + * chunks. It also has a control mode, which serves for audio format setting. + * + * Looking at the CS4215 data sheet it is easy to set up 2 or 4 codecs on + * the same CHI bus, so I thought perhaps it is possible to use the onboard + * & the speakerbox codec simultanously, giving 2 (not very independent :-) + * audio devices. But the SUN HW group decided against it, at least on my + * LX the speakerbox connector has at least 1 pin missing and 1 wrongly + * connected. + */ + +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +MODULE_AUTHOR("Rudolf Koenig, Brent Baccala and Martin Habets"); +MODULE_DESCRIPTION("Sun DBRI"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Sun,DBRI}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Sun DBRI soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Sun DBRI soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Sun DBRI soundcard."); + +#define DBRI_DEBUG + +#define D_INT (1<<0) +#define D_GEN (1<<1) +#define D_CMD (1<<2) +#define D_MM (1<<3) +#define D_USR (1<<4) +#define D_DESC (1<<5) + +static int dbri_debug = 0; +module_param(dbri_debug, int, 0444); +MODULE_PARM_DESC(dbri_debug, "Debug value for Sun DBRI soundcard."); + +#ifdef DBRI_DEBUG +static char *cmds[] = { + "WAIT", "PAUSE", "JUMP", "IIQ", "REX", "SDP", "CDP", "DTS", + "SSP", "CHI", "NT", "TE", "CDEC", "TEST", "CDM", "RESRV" +}; + +#define dprintk(a, x...) if(dbri_debug & a) printk(KERN_DEBUG x) + +#define DBRI_CMD(cmd, intr, value) ((cmd << 28) | \ + (1 << 27) | \ + value) +#else +#define dprintk(a, x...) + +#define DBRI_CMD(cmd, intr, value) ((cmd << 28) | \ + (intr << 27) | \ + value) +#endif /* DBRI_DEBUG */ + +/*************************************************************************** + CS4215 specific definitions and structures +****************************************************************************/ + +struct cs4215 { + __u8 data[4]; /* Data mode: Time slots 5-8 */ + __u8 ctrl[4]; /* Ctrl mode: Time slots 1-4 */ + __u8 onboard; + __u8 offset; /* Bit offset from frame sync to time slot 1 */ + volatile __u32 status; + volatile __u32 version; + __u8 precision; /* In bits, either 8 or 16 */ + __u8 channels; /* 1 or 2 */ +}; + +/* + * Control mode first + */ + +/* Time Slot 1, Status register */ +#define CS4215_CLB (1<<2) /* Control Latch Bit */ +#define CS4215_OLB (1<<3) /* 1: line: 2.0V, speaker 4V */ + /* 0: line: 2.8V, speaker 8V */ +#define CS4215_MLB (1<<4) /* 1: Microphone: 20dB gain disabled */ +#define CS4215_RSRVD_1 (1<<5) + +/* Time Slot 2, Data Format Register */ +#define CS4215_DFR_LINEAR16 0 +#define CS4215_DFR_ULAW 1 +#define CS4215_DFR_ALAW 2 +#define CS4215_DFR_LINEAR8 3 +#define CS4215_DFR_STEREO (1<<2) +static struct { + unsigned short freq; + unsigned char xtal; + unsigned char csval; +} CS4215_FREQ[] = { + { 8000, (1 << 4), (0 << 3) }, + { 16000, (1 << 4), (1 << 3) }, + { 27429, (1 << 4), (2 << 3) }, /* Actually 24428.57 */ + { 32000, (1 << 4), (3 << 3) }, + /* { NA, (1 << 4), (4 << 3) }, */ + /* { NA, (1 << 4), (5 << 3) }, */ + { 48000, (1 << 4), (6 << 3) }, + { 9600, (1 << 4), (7 << 3) }, + { 5513, (2 << 4), (0 << 3) }, /* Actually 5512.5 */ + { 11025, (2 << 4), (1 << 3) }, + { 18900, (2 << 4), (2 << 3) }, + { 22050, (2 << 4), (3 << 3) }, + { 37800, (2 << 4), (4 << 3) }, + { 44100, (2 << 4), (5 << 3) }, + { 33075, (2 << 4), (6 << 3) }, + { 6615, (2 << 4), (7 << 3) }, + { 0, 0, 0} +}; + +#define CS4215_HPF (1<<7) /* High Pass Filter, 1: Enabled */ + +#define CS4215_12_MASK 0xfcbf /* Mask off reserved bits in slot 1 & 2 */ + +/* Time Slot 3, Serial Port Control register */ +#define CS4215_XEN (1<<0) /* 0: Enable serial output */ +#define CS4215_XCLK (1<<1) /* 1: Master mode: Generate SCLK */ +#define CS4215_BSEL_64 (0<<2) /* Bitrate: 64 bits per frame */ +#define CS4215_BSEL_128 (1<<2) +#define CS4215_BSEL_256 (2<<2) +#define CS4215_MCK_MAST (0<<4) /* Master clock */ +#define CS4215_MCK_XTL1 (1<<4) /* 24.576 MHz clock source */ +#define CS4215_MCK_XTL2 (2<<4) /* 16.9344 MHz clock source */ +#define CS4215_MCK_CLK1 (3<<4) /* Clockin, 256 x Fs */ +#define CS4215_MCK_CLK2 (4<<4) /* Clockin, see DFR */ + +/* Time Slot 4, Test Register */ +#define CS4215_DAD (1<<0) /* 0:Digital-Dig loop, 1:Dig-Analog-Dig loop */ +#define CS4215_ENL (1<<1) /* Enable Loopback Testing */ + +/* Time Slot 5, Parallel Port Register */ +/* Read only here and the same as the in data mode */ + +/* Time Slot 6, Reserved */ + +/* Time Slot 7, Version Register */ +#define CS4215_VERSION_MASK 0xf /* Known versions 0/C, 1/D, 2/E */ + +/* Time Slot 8, Reserved */ + +/* + * Data mode + */ +/* Time Slot 1-2: Left Channel Data, 2-3: Right Channel Data */ + +/* Time Slot 5, Output Setting */ +#define CS4215_LO(v) v /* Left Output Attenuation 0x3f: -94.5 dB */ +#define CS4215_LE (1<<6) /* Line Out Enable */ +#define CS4215_HE (1<<7) /* Headphone Enable */ + +/* Time Slot 6, Output Setting */ +#define CS4215_RO(v) v /* Right Output Attenuation 0x3f: -94.5 dB */ +#define CS4215_SE (1<<6) /* Speaker Enable */ +#define CS4215_ADI (1<<7) /* A/D Data Invalid: Busy in calibration */ + +/* Time Slot 7, Input Setting */ +#define CS4215_LG(v) v /* Left Gain Setting 0xf: 22.5 dB */ +#define CS4215_IS (1<<4) /* Input Select: 1=Microphone, 0=Line */ +#define CS4215_OVR (1<<5) /* 1: Overrange condition occurred */ +#define CS4215_PIO0 (1<<6) /* Parallel I/O 0 */ +#define CS4215_PIO1 (1<<7) + +/* Time Slot 8, Input Setting */ +#define CS4215_RG(v) v /* Right Gain Setting 0xf: 22.5 dB */ +#define CS4215_MA(v) (v<<4) /* Monitor Path Attenuation 0xf: mute */ + +/*************************************************************************** + DBRI specific definitions and structures +****************************************************************************/ + +/* DBRI main registers */ +#define REG0 0x00UL /* Status and Control */ +#define REG1 0x04UL /* Mode and Interrupt */ +#define REG2 0x08UL /* Parallel IO */ +#define REG3 0x0cUL /* Test */ +#define REG8 0x20UL /* Command Queue Pointer */ +#define REG9 0x24UL /* Interrupt Queue Pointer */ + +#define DBRI_NO_CMDS 64 +#define DBRI_NO_INTS 1 /* Note: the value of this define was + * originally 2. The ringbuffer to store + * interrupts in dma is currently broken. + * This is a temporary fix until the ringbuffer + * is fixed. + */ +#define DBRI_INT_BLK 64 +#define DBRI_NO_DESCS 64 +#define DBRI_NO_PIPES 32 + +#define DBRI_MM_ONB 1 +#define DBRI_MM_SB 2 + +#define DBRI_REC 0 +#define DBRI_PLAY 1 +#define DBRI_NO_STREAMS 2 + +/* One transmit/receive descriptor */ +struct dbri_mem { + volatile __u32 word1; + volatile __u32 ba; /* Transmit/Receive Buffer Address */ + volatile __u32 nda; /* Next Descriptor Address */ + volatile __u32 word4; +}; + +/* This structure is in a DMA region where it can accessed by both + * the CPU and the DBRI + */ +struct dbri_dma { + volatile s32 cmd[DBRI_NO_CMDS]; /* Place for commands */ + volatile s32 intr[DBRI_NO_INTS * DBRI_INT_BLK]; /* Interrupt field */ + struct dbri_mem desc[DBRI_NO_DESCS]; /* Xmit/receive descriptors */ +}; + +#define dbri_dma_off(member, elem) \ + ((u32)(unsigned long) \ + (&(((struct dbri_dma *)0)->member[elem]))) + +enum in_or_out { PIPEinput, PIPEoutput }; + +struct dbri_pipe { + u32 sdp; /* SDP command word */ + enum in_or_out direction; + int nextpipe; /* Next pipe in linked list */ + int prevpipe; + int cycle; /* Offset of timeslot (bits) */ + int length; /* Length of timeslot (bits) */ + int first_desc; /* Index of first descriptor */ + int desc; /* Index of active descriptor */ + volatile __u32 *recv_fixed_ptr; /* Ptr to receive fixed data */ +}; + +struct dbri_desc { + int inuse; /* Boolean flag */ + int next; /* Index of next desc, or -1 */ + unsigned int len; +}; + +/* Per stream (playback or record) information */ +typedef struct dbri_streaminfo { + snd_pcm_substream_t *substream; + u32 dvma_buffer; /* Device view of Alsa DMA buffer */ + int left; /* # of bytes left in DMA buffer */ + int size; /* Size of DMA buffer */ + size_t offset; /* offset in user buffer */ + int pipe; /* Data pipe used */ + int left_gain; /* mixer elements */ + int right_gain; + int balance; +} dbri_streaminfo_t; + +/* This structure holds the information for both chips (DBRI & CS4215) */ +typedef struct snd_dbri { + snd_card_t *card; /* ALSA card */ + snd_pcm_t *pcm; + + int regs_size, irq; /* Needed for unload */ + struct sbus_dev *sdev; /* SBUS device info */ + spinlock_t lock; + + volatile struct dbri_dma *dma; /* Pointer to our DMA block */ + u32 dma_dvma; /* DBRI visible DMA address */ + + void __iomem *regs; /* dbri HW regs */ + int dbri_version; /* 'e' and up is OK */ + int dbri_irqp; /* intr queue pointer */ + int wait_seen; + + struct dbri_pipe pipes[DBRI_NO_PIPES]; /* DBRI's 32 data pipes */ + struct dbri_desc descs[DBRI_NO_DESCS]; + + int chi_in_pipe; + int chi_out_pipe; + int chi_bpf; + + struct cs4215 mm; /* mmcodec special info */ + /* per stream (playback/record) info */ + struct dbri_streaminfo stream_info[DBRI_NO_STREAMS]; + + struct snd_dbri *next; +} snd_dbri_t; + +/* Needed for the ALSA macros to work */ +#define chip_t snd_dbri_t + +#define DBRI_MAX_VOLUME 63 /* Output volume */ +#define DBRI_MAX_GAIN 15 /* Input gain */ +#define DBRI_RIGHT_BALANCE 255 +#define DBRI_MID_BALANCE (DBRI_RIGHT_BALANCE >> 1) + +/* DBRI Reg0 - Status Control Register - defines. (Page 17) */ +#define D_P (1<<15) /* Program command & queue pointer valid */ +#define D_G (1<<14) /* Allow 4-Word SBus Burst */ +#define D_S (1<<13) /* Allow 16-Word SBus Burst */ +#define D_E (1<<12) /* Allow 8-Word SBus Burst */ +#define D_X (1<<7) /* Sanity Timer Disable */ +#define D_T (1<<6) /* Permit activation of the TE interface */ +#define D_N (1<<5) /* Permit activation of the NT interface */ +#define D_C (1<<4) /* Permit activation of the CHI interface */ +#define D_F (1<<3) /* Force Sanity Timer Time-Out */ +#define D_D (1<<2) /* Disable Master Mode */ +#define D_H (1<<1) /* Halt for Analysis */ +#define D_R (1<<0) /* Soft Reset */ + +/* DBRI Reg1 - Mode and Interrupt Register - defines. (Page 18) */ +#define D_LITTLE_END (1<<8) /* Byte Order */ +#define D_BIG_END (0<<8) /* Byte Order */ +#define D_MRR (1<<4) /* Multiple Error Ack on SBus (readonly) */ +#define D_MLE (1<<3) /* Multiple Late Error on SBus (readonly) */ +#define D_LBG (1<<2) /* Lost Bus Grant on SBus (readonly) */ +#define D_MBE (1<<1) /* Burst Error on SBus (readonly) */ +#define D_IR (1<<0) /* Interrupt Indicator (readonly) */ + +/* DBRI Reg2 - Parallel IO Register - defines. (Page 18) */ +#define D_ENPIO3 (1<<7) /* Enable Pin 3 */ +#define D_ENPIO2 (1<<6) /* Enable Pin 2 */ +#define D_ENPIO1 (1<<5) /* Enable Pin 1 */ +#define D_ENPIO0 (1<<4) /* Enable Pin 0 */ +#define D_ENPIO (0xf0) /* Enable all the pins */ +#define D_PIO3 (1<<3) /* Pin 3: 1: Data mode, 0: Ctrl mode */ +#define D_PIO2 (1<<2) /* Pin 2: 1: Onboard PDN */ +#define D_PIO1 (1<<1) /* Pin 1: 0: Reset */ +#define D_PIO0 (1<<0) /* Pin 0: 1: Speakerbox PDN */ + +/* DBRI Commands (Page 20) */ +#define D_WAIT 0x0 /* Stop execution */ +#define D_PAUSE 0x1 /* Flush long pipes */ +#define D_JUMP 0x2 /* New command queue */ +#define D_IIQ 0x3 /* Initialize Interrupt Queue */ +#define D_REX 0x4 /* Report command execution via interrupt */ +#define D_SDP 0x5 /* Setup Data Pipe */ +#define D_CDP 0x6 /* Continue Data Pipe (reread NULL Pointer) */ +#define D_DTS 0x7 /* Define Time Slot */ +#define D_SSP 0x8 /* Set short Data Pipe */ +#define D_CHI 0x9 /* Set CHI Global Mode */ +#define D_NT 0xa /* NT Command */ +#define D_TE 0xb /* TE Command */ +#define D_CDEC 0xc /* Codec setup */ +#define D_TEST 0xd /* No comment */ +#define D_CDM 0xe /* CHI Data mode command */ + +/* Special bits for some commands */ +#define D_PIPE(v) ((v)<<0) /* Pipe Nr: 0-15 long, 16-21 short */ + +/* Setup Data Pipe */ +/* IRM */ +#define D_SDP_2SAME (1<<18) /* Report 2nd time in a row value rcvd */ +#define D_SDP_CHANGE (2<<18) /* Report any changes */ +#define D_SDP_EVERY (3<<18) /* Report any changes */ +#define D_SDP_EOL (1<<17) /* EOL interrupt enable */ +#define D_SDP_IDLE (1<<16) /* HDLC idle interrupt enable */ + +/* Pipe data MODE */ +#define D_SDP_MEM (0<<13) /* To/from memory */ +#define D_SDP_HDLC (2<<13) +#define D_SDP_HDLC_D (3<<13) /* D Channel (prio control) */ +#define D_SDP_SER (4<<13) /* Serial to serial */ +#define D_SDP_FIXED (6<<13) /* Short only */ +#define D_SDP_MODE(v) ((v)&(7<<13)) + +#define D_SDP_TO_SER (1<<12) /* Direction */ +#define D_SDP_FROM_SER (0<<12) /* Direction */ +#define D_SDP_MSB (1<<11) /* Bit order within Byte */ +#define D_SDP_LSB (0<<11) /* Bit order within Byte */ +#define D_SDP_P (1<<10) /* Pointer Valid */ +#define D_SDP_A (1<<8) /* Abort */ +#define D_SDP_C (1<<7) /* Clear */ + +/* Define Time Slot */ +#define D_DTS_VI (1<<17) /* Valid Input Time-Slot Descriptor */ +#define D_DTS_VO (1<<16) /* Valid Output Time-Slot Descriptor */ +#define D_DTS_INS (1<<15) /* Insert Time Slot */ +#define D_DTS_DEL (0<<15) /* Delete Time Slot */ +#define D_DTS_PRVIN(v) ((v)<<10) /* Previous In Pipe */ +#define D_DTS_PRVOUT(v) ((v)<<5) /* Previous Out Pipe */ + +/* Time Slot defines */ +#define D_TS_LEN(v) ((v)<<24) /* Number of bits in this time slot */ +#define D_TS_CYCLE(v) ((v)<<14) /* Bit Count at start of TS */ +#define D_TS_DI (1<<13) /* Data Invert */ +#define D_TS_1CHANNEL (0<<10) /* Single Channel / Normal mode */ +#define D_TS_MONITOR (2<<10) /* Monitor pipe */ +#define D_TS_NONCONTIG (3<<10) /* Non contiguous mode */ +#define D_TS_ANCHOR (7<<10) /* Starting short pipes */ +#define D_TS_MON(v) ((v)<<5) /* Monitor Pipe */ +#define D_TS_NEXT(v) ((v)<<0) /* Pipe Nr: 0-15 long, 16-21 short */ + +/* Concentration Highway Interface Modes */ +#define D_CHI_CHICM(v) ((v)<<16) /* Clock mode */ +#define D_CHI_IR (1<<15) /* Immediate Interrupt Report */ +#define D_CHI_EN (1<<14) /* CHIL Interrupt enabled */ +#define D_CHI_OD (1<<13) /* Open Drain Enable */ +#define D_CHI_FE (1<<12) /* Sample CHIFS on Rising Frame Edge */ +#define D_CHI_FD (1<<11) /* Frame Drive */ +#define D_CHI_BPF(v) ((v)<<0) /* Bits per Frame */ + +/* NT: These are here for completeness */ +#define D_NT_FBIT (1<<17) /* Frame Bit */ +#define D_NT_NBF (1<<16) /* Number of bad frames to loose framing */ +#define D_NT_IRM_IMM (1<<15) /* Interrupt Report & Mask: Immediate */ +#define D_NT_IRM_EN (1<<14) /* Interrupt Report & Mask: Enable */ +#define D_NT_ISNT (1<<13) /* Configfure interface as NT */ +#define D_NT_FT (1<<12) /* Fixed Timing */ +#define D_NT_EZ (1<<11) /* Echo Channel is Zeros */ +#define D_NT_IFA (1<<10) /* Inhibit Final Activation */ +#define D_NT_ACT (1<<9) /* Activate Interface */ +#define D_NT_MFE (1<<8) /* Multiframe Enable */ +#define D_NT_RLB(v) ((v)<<5) /* Remote Loopback */ +#define D_NT_LLB(v) ((v)<<2) /* Local Loopback */ +#define D_NT_FACT (1<<1) /* Force Activation */ +#define D_NT_ABV (1<<0) /* Activate Bipolar Violation */ + +/* Codec Setup */ +#define D_CDEC_CK(v) ((v)<<24) /* Clock Select */ +#define D_CDEC_FED(v) ((v)<<12) /* FSCOD Falling Edge Delay */ +#define D_CDEC_RED(v) ((v)<<0) /* FSCOD Rising Edge Delay */ + +/* Test */ +#define D_TEST_RAM(v) ((v)<<16) /* RAM Pointer */ +#define D_TEST_SIZE(v) ((v)<<11) /* */ +#define D_TEST_ROMONOFF 0x5 /* Toggle ROM opcode monitor on/off */ +#define D_TEST_PROC 0x6 /* MicroProcessor test */ +#define D_TEST_SER 0x7 /* Serial-Controller test */ +#define D_TEST_RAMREAD 0x8 /* Copy from Ram to system memory */ +#define D_TEST_RAMWRITE 0x9 /* Copy into Ram from system memory */ +#define D_TEST_RAMBIST 0xa /* RAM Built-In Self Test */ +#define D_TEST_MCBIST 0xb /* Microcontroller Built-In Self Test */ +#define D_TEST_DUMP 0xe /* ROM Dump */ + +/* CHI Data Mode */ +#define D_CDM_THI (1<<8) /* Transmit Data on CHIDR Pin */ +#define D_CDM_RHI (1<<7) /* Receive Data on CHIDX Pin */ +#define D_CDM_RCE (1<<6) /* Receive on Rising Edge of CHICK */ +#define D_CDM_XCE (1<<2) /* Transmit Data on Rising Edge of CHICK */ +#define D_CDM_XEN (1<<1) /* Transmit Highway Enable */ +#define D_CDM_REN (1<<0) /* Receive Highway Enable */ + +/* The Interrupts */ +#define D_INTR_BRDY 1 /* Buffer Ready for processing */ +#define D_INTR_MINT 2 /* Marked Interrupt in RD/TD */ +#define D_INTR_IBEG 3 /* Flag to idle transition detected (HDLC) */ +#define D_INTR_IEND 4 /* Idle to flag transition detected (HDLC) */ +#define D_INTR_EOL 5 /* End of List */ +#define D_INTR_CMDI 6 /* Command has bean read */ +#define D_INTR_XCMP 8 /* Transmission of frame complete */ +#define D_INTR_SBRI 9 /* BRI status change info */ +#define D_INTR_FXDT 10 /* Fixed data change */ +#define D_INTR_CHIL 11 /* CHI lost frame sync (channel 36 only) */ +#define D_INTR_COLL 11 /* Unrecoverable D-Channel collision */ +#define D_INTR_DBYT 12 /* Dropped by frame slip */ +#define D_INTR_RBYT 13 /* Repeated by frame slip */ +#define D_INTR_LINT 14 /* Lost Interrupt */ +#define D_INTR_UNDR 15 /* DMA underrun */ + +#define D_INTR_TE 32 +#define D_INTR_NT 34 +#define D_INTR_CHI 36 +#define D_INTR_CMD 38 + +#define D_INTR_GETCHAN(v) (((v)>>24) & 0x3f) +#define D_INTR_GETCODE(v) (((v)>>20) & 0xf) +#define D_INTR_GETCMD(v) (((v)>>16) & 0xf) +#define D_INTR_GETVAL(v) ((v) & 0xffff) +#define D_INTR_GETRVAL(v) ((v) & 0xfffff) + +#define D_P_0 0 /* TE receive anchor */ +#define D_P_1 1 /* TE transmit anchor */ +#define D_P_2 2 /* NT transmit anchor */ +#define D_P_3 3 /* NT receive anchor */ +#define D_P_4 4 /* CHI send data */ +#define D_P_5 5 /* CHI receive data */ +#define D_P_6 6 /* */ +#define D_P_7 7 /* */ +#define D_P_8 8 /* */ +#define D_P_9 9 /* */ +#define D_P_10 10 /* */ +#define D_P_11 11 /* */ +#define D_P_12 12 /* */ +#define D_P_13 13 /* */ +#define D_P_14 14 /* */ +#define D_P_15 15 /* */ +#define D_P_16 16 /* CHI anchor pipe */ +#define D_P_17 17 /* CHI send */ +#define D_P_18 18 /* CHI receive */ +#define D_P_19 19 /* CHI receive */ +#define D_P_20 20 /* CHI receive */ +#define D_P_21 21 /* */ +#define D_P_22 22 /* */ +#define D_P_23 23 /* */ +#define D_P_24 24 /* */ +#define D_P_25 25 /* */ +#define D_P_26 26 /* */ +#define D_P_27 27 /* */ +#define D_P_28 28 /* */ +#define D_P_29 29 /* */ +#define D_P_30 30 /* */ +#define D_P_31 31 /* */ + +/* Transmit descriptor defines */ +#define DBRI_TD_F (1<<31) /* End of Frame */ +#define DBRI_TD_D (1<<30) /* Do not append CRC */ +#define DBRI_TD_CNT(v) ((v)<<16) /* Number of valid bytes in the buffer */ +#define DBRI_TD_B (1<<15) /* Final interrupt */ +#define DBRI_TD_M (1<<14) /* Marker interrupt */ +#define DBRI_TD_I (1<<13) /* Transmit Idle Characters */ +#define DBRI_TD_FCNT(v) (v) /* Flag Count */ +#define DBRI_TD_UNR (1<<3) /* Underrun: transmitter is out of data */ +#define DBRI_TD_ABT (1<<2) /* Abort: frame aborted */ +#define DBRI_TD_TBC (1<<0) /* Transmit buffer Complete */ +#define DBRI_TD_STATUS(v) ((v)&0xff) /* Transmit status */ + /* Maximum buffer size per TD: almost 8Kb */ +#define DBRI_TD_MAXCNT ((1 << 13) - 1) + +/* Receive descriptor defines */ +#define DBRI_RD_F (1<<31) /* End of Frame */ +#define DBRI_RD_C (1<<30) /* Completed buffer */ +#define DBRI_RD_B (1<<15) /* Final interrupt */ +#define DBRI_RD_M (1<<14) /* Marker interrupt */ +#define DBRI_RD_BCNT(v) (v) /* Buffer size */ +#define DBRI_RD_CRC (1<<7) /* 0: CRC is correct */ +#define DBRI_RD_BBC (1<<6) /* 1: Bad Byte received */ +#define DBRI_RD_ABT (1<<5) /* Abort: frame aborted */ +#define DBRI_RD_OVRN (1<<3) /* Overrun: data lost */ +#define DBRI_RD_STATUS(v) ((v)&0xff) /* Receive status */ +#define DBRI_RD_CNT(v) (((v)>>16)&0x1fff) /* Valid bytes in the buffer */ + +/* stream_info[] access */ +/* Translate the ALSA direction into the array index */ +#define DBRI_STREAMNO(substream) \ + (substream->stream == \ + SNDRV_PCM_STREAM_PLAYBACK? DBRI_PLAY: DBRI_REC) + +/* Return a pointer to dbri_streaminfo */ +#define DBRI_STREAM(dbri, substream) &dbri->stream_info[DBRI_STREAMNO(substream)] + +static snd_dbri_t *dbri_list = NULL; /* All DBRI devices */ + +/* + * Short data pipes transmit LSB first. The CS4215 receives MSB first. Grrr. + * So we have to reverse the bits. Note: not all bit lengths are supported + */ +static __u32 reverse_bytes(__u32 b, int len) +{ + switch (len) { + case 32: + b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16); + case 16: + b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8); + case 8: + b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4); + case 4: + b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2); + case 2: + b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1); + case 1: + case 0: + break; + default: + printk(KERN_ERR "DBRI reverse_bytes: unsupported length\n"); + }; + + return b; +} + +/* +**************************************************************************** +************** DBRI initialization and command synchronization ************* +**************************************************************************** + +Commands are sent to the DBRI by building a list of them in memory, +then writing the address of the first list item to DBRI register 8. +The list is terminated with a WAIT command, which can generate a +CPU interrupt if required. + +Since the DBRI can run in parallel with the CPU, several means of +synchronization present themselves. The original scheme (Rudolf's) +was to set a flag when we "cmdlock"ed the DBRI, clear the flag when +an interrupt signaled completion, and wait on a wait_queue if a routine +attempted to cmdlock while the flag was set. The problems arose when +we tried to cmdlock from inside an interrupt handler, which might +cause scheduling in an interrupt (if we waited), etc, etc + +A more sophisticated scheme might involve a circular command buffer +or an array of command buffers. A routine could fill one with +commands and link it onto a list. When a interrupt signaled +completion of the current command buffer, look on the list for +the next one. + +I've decided to implement something much simpler - after each command, +the CPU waits for the DBRI to finish the command by polling the P bit +in DBRI register 0. I've tried to implement this in such a way +that might make implementing a more sophisticated scheme easier. + +Every time a routine wants to write commands to the DBRI, it must +first call dbri_cmdlock() and get an initial pointer into dbri->dma->cmd +in return. After the commands have been writen, dbri_cmdsend() is +called with the final pointer value. + +*/ + +enum dbri_lock_t { NoGetLock, GetLock }; + +static volatile s32 *dbri_cmdlock(snd_dbri_t * dbri, enum dbri_lock_t get) +{ +#ifndef SMP + if ((get == GetLock) && spin_is_locked(&dbri->lock)) { + printk(KERN_ERR "DBRI: cmdlock called while in spinlock."); + } +#endif + + /*if (get == GetLock) spin_lock(&dbri->lock); */ + return &dbri->dma->cmd[0]; +} + +static void dbri_process_interrupt_buffer(snd_dbri_t *); + +static void dbri_cmdsend(snd_dbri_t * dbri, volatile s32 * cmd) +{ + int MAXLOOPS = 1000000; + int maxloops = MAXLOOPS; + volatile s32 *ptr; + + for (ptr = &dbri->dma->cmd[0]; ptr < cmd; ptr++) { + dprintk(D_CMD, "cmd: %lx:%08x\n", (unsigned long)ptr, *ptr); + } + + if ((cmd - &dbri->dma->cmd[0]) >= DBRI_NO_CMDS - 1) { + printk("DBRI: Command buffer overflow! (bug in driver)\n"); + /* Ignore the last part. */ + cmd = &dbri->dma->cmd[DBRI_NO_CMDS - 3]; + } + + *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0); + *(cmd++) = DBRI_CMD(D_WAIT, 1, 0); + dbri->wait_seen = 0; + sbus_writel(dbri->dma_dvma, dbri->regs + REG8); + while ((--maxloops) > 0 && (sbus_readl(dbri->regs + REG0) & D_P)) + barrier(); + if (maxloops == 0) { + printk(KERN_ERR "DBRI: Chip never completed command buffer\n"); + dprintk(D_CMD, "DBRI: Chip never completed command buffer\n"); + } else { + while ((--maxloops) > 0 && (!dbri->wait_seen)) + dbri_process_interrupt_buffer(dbri); + if (maxloops == 0) { + printk(KERN_ERR "DBRI: Chip never acked WAIT\n"); + dprintk(D_CMD, "DBRI: Chip never acked WAIT\n"); + } else { + dprintk(D_CMD, "Chip completed command " + "buffer (%d)\n", MAXLOOPS - maxloops); + } + } + + /*spin_unlock(&dbri->lock); */ +} + +/* Lock must be held when calling this */ +static void dbri_reset(snd_dbri_t * dbri) +{ + int i; + + dprintk(D_GEN, "reset 0:%x 2:%x 8:%x 9:%x\n", + sbus_readl(dbri->regs + REG0), + sbus_readl(dbri->regs + REG2), + sbus_readl(dbri->regs + REG8), sbus_readl(dbri->regs + REG9)); + + sbus_writel(D_R, dbri->regs + REG0); /* Soft Reset */ + for (i = 0; (sbus_readl(dbri->regs + REG0) & D_R) && i < 64; i++) + udelay(10); +} + +/* Lock must not be held before calling this */ +static void dbri_initialize(snd_dbri_t * dbri) +{ + volatile s32 *cmd; + u32 dma_addr, tmp; + unsigned long flags; + int n; + + spin_lock_irqsave(&dbri->lock, flags); + + dbri_reset(dbri); + + cmd = dbri_cmdlock(dbri, NoGetLock); + dprintk(D_GEN, "init: cmd: %p, int: %p\n", + &dbri->dma->cmd[0], &dbri->dma->intr[0]); + + /* + * Initialize the interrupt ringbuffer. + */ + for (n = 0; n < DBRI_NO_INTS - 1; n++) { + dma_addr = dbri->dma_dvma; + dma_addr += dbri_dma_off(intr, ((n + 1) & DBRI_INT_BLK)); + dbri->dma->intr[n * DBRI_INT_BLK] = dma_addr; + } + dma_addr = dbri->dma_dvma + dbri_dma_off(intr, 0); + dbri->dma->intr[n * DBRI_INT_BLK] = dma_addr; + dbri->dbri_irqp = 1; + + /* Initialize pipes */ + for (n = 0; n < DBRI_NO_PIPES; n++) + dbri->pipes[n].desc = dbri->pipes[n].first_desc = -1; + + /* We should query the openprom to see what burst sizes this + * SBus supports. For now, just disable all SBus bursts */ + tmp = sbus_readl(dbri->regs + REG0); + tmp &= ~(D_G | D_S | D_E); + sbus_writel(tmp, dbri->regs + REG0); + + /* + * Set up the interrupt queue + */ + dma_addr = dbri->dma_dvma + dbri_dma_off(intr, 0); + *(cmd++) = DBRI_CMD(D_IIQ, 0, 0); + *(cmd++) = dma_addr; + + dbri_cmdsend(dbri, cmd); + spin_unlock_irqrestore(&dbri->lock, flags); +} + +/* +**************************************************************************** +************************** DBRI data pipe management *********************** +**************************************************************************** + +While DBRI control functions use the command and interrupt buffers, the +main data path takes the form of data pipes, which can be short (command +and interrupt driven), or long (attached to DMA buffers). These functions +provide a rudimentary means of setting up and managing the DBRI's pipes, +but the calling functions have to make sure they respect the pipes' linked +list ordering, among other things. The transmit and receive functions +here interface closely with the transmit and receive interrupt code. + +*/ +static int pipe_active(snd_dbri_t * dbri, int pipe) +{ + return ((pipe >= 0) && (dbri->pipes[pipe].desc != -1)); +} + +/* reset_pipe(dbri, pipe) + * + * Called on an in-use pipe to clear anything being transmitted or received + * Lock must be held before calling this. + */ +static void reset_pipe(snd_dbri_t * dbri, int pipe) +{ + int sdp; + int desc; + volatile int *cmd; + + if (pipe < 0 || pipe > 31) { + printk("DBRI: reset_pipe called with illegal pipe number\n"); + return; + } + + sdp = dbri->pipes[pipe].sdp; + if (sdp == 0) { + printk("DBRI: reset_pipe called on uninitialized pipe\n"); + return; + } + + cmd = dbri_cmdlock(dbri, NoGetLock); + *(cmd++) = DBRI_CMD(D_SDP, 0, sdp | D_SDP_C | D_SDP_P); + *(cmd++) = 0; + dbri_cmdsend(dbri, cmd); + + desc = dbri->pipes[pipe].first_desc; + while (desc != -1) { + dbri->descs[desc].inuse = 0; + desc = dbri->descs[desc].next; + } + + dbri->pipes[pipe].desc = -1; + dbri->pipes[pipe].first_desc = -1; +} + +/* FIXME: direction as an argument? */ +static void setup_pipe(snd_dbri_t * dbri, int pipe, int sdp) +{ + if (pipe < 0 || pipe > 31) { + printk("DBRI: setup_pipe called with illegal pipe number\n"); + return; + } + + if ((sdp & 0xf800) != sdp) { + printk("DBRI: setup_pipe called with strange SDP value\n"); + /* sdp &= 0xf800; */ + } + + /* If this is a fixed receive pipe, arrange for an interrupt + * every time its data changes + */ + if (D_SDP_MODE(sdp) == D_SDP_FIXED && !(sdp & D_SDP_TO_SER)) + sdp |= D_SDP_CHANGE; + + sdp |= D_PIPE(pipe); + dbri->pipes[pipe].sdp = sdp; + dbri->pipes[pipe].desc = -1; + dbri->pipes[pipe].first_desc = -1; + if (sdp & D_SDP_TO_SER) + dbri->pipes[pipe].direction = PIPEoutput; + else + dbri->pipes[pipe].direction = PIPEinput; + + reset_pipe(dbri, pipe); +} + +/* FIXME: direction not needed */ +static void link_time_slot(snd_dbri_t * dbri, int pipe, + enum in_or_out direction, int basepipe, + int length, int cycle) +{ + volatile s32 *cmd; + int val; + int prevpipe; + int nextpipe; + + if (pipe < 0 || pipe > 31 || basepipe < 0 || basepipe > 31) { + printk + ("DBRI: link_time_slot called with illegal pipe number\n"); + return; + } + + if (dbri->pipes[pipe].sdp == 0 || dbri->pipes[basepipe].sdp == 0) { + printk("DBRI: link_time_slot called on uninitialized pipe\n"); + return; + } + + /* Deal with CHI special case: + * "If transmission on edges 0 or 1 is desired, then cycle n + * (where n = # of bit times per frame...) must be used." + * - DBRI data sheet, page 11 + */ + if (basepipe == 16 && direction == PIPEoutput && cycle == 0) + cycle = dbri->chi_bpf; + + if (basepipe == pipe) { + prevpipe = pipe; + nextpipe = pipe; + } else { + /* We're not initializing a new linked list (basepipe != pipe), + * so run through the linked list and find where this pipe + * should be sloted in, based on its cycle. CHI confuses + * things a bit, since it has a single anchor for both its + * transmit and receive lists. + */ + if (basepipe == 16) { + if (direction == PIPEinput) { + prevpipe = dbri->chi_in_pipe; + } else { + prevpipe = dbri->chi_out_pipe; + } + } else { + prevpipe = basepipe; + } + + nextpipe = dbri->pipes[prevpipe].nextpipe; + + while (dbri->pipes[nextpipe].cycle < cycle + && dbri->pipes[nextpipe].nextpipe != basepipe) { + prevpipe = nextpipe; + nextpipe = dbri->pipes[nextpipe].nextpipe; + } + } + + if (prevpipe == 16) { + if (direction == PIPEinput) { + dbri->chi_in_pipe = pipe; + } else { + dbri->chi_out_pipe = pipe; + } + } else { + dbri->pipes[prevpipe].nextpipe = pipe; + } + + dbri->pipes[pipe].nextpipe = nextpipe; + dbri->pipes[pipe].cycle = cycle; + dbri->pipes[pipe].length = length; + + cmd = dbri_cmdlock(dbri, NoGetLock); + + if (direction == PIPEinput) { + val = D_DTS_VI | D_DTS_INS | D_DTS_PRVIN(prevpipe) | pipe; + *(cmd++) = DBRI_CMD(D_DTS, 0, val); + *(cmd++) = + D_TS_LEN(length) | D_TS_CYCLE(cycle) | D_TS_NEXT(nextpipe); + *(cmd++) = 0; + } else { + val = D_DTS_VO | D_DTS_INS | D_DTS_PRVOUT(prevpipe) | pipe; + *(cmd++) = DBRI_CMD(D_DTS, 0, val); + *(cmd++) = 0; + *(cmd++) = + D_TS_LEN(length) | D_TS_CYCLE(cycle) | D_TS_NEXT(nextpipe); + } + + dbri_cmdsend(dbri, cmd); +} + +static void unlink_time_slot(snd_dbri_t * dbri, int pipe, + enum in_or_out direction, int prevpipe, + int nextpipe) +{ + volatile s32 *cmd; + int val; + + if (pipe < 0 || pipe > 31 || prevpipe < 0 || prevpipe > 31) { + printk + ("DBRI: unlink_time_slot called with illegal pipe number\n"); + return; + } + + cmd = dbri_cmdlock(dbri, NoGetLock); + + if (direction == PIPEinput) { + val = D_DTS_VI | D_DTS_DEL | D_DTS_PRVIN(prevpipe) | pipe; + *(cmd++) = DBRI_CMD(D_DTS, 0, val); + *(cmd++) = D_TS_NEXT(nextpipe); + *(cmd++) = 0; + } else { + val = D_DTS_VO | D_DTS_DEL | D_DTS_PRVOUT(prevpipe) | pipe; + *(cmd++) = DBRI_CMD(D_DTS, 0, val); + *(cmd++) = 0; + *(cmd++) = D_TS_NEXT(nextpipe); + } + + dbri_cmdsend(dbri, cmd); +} + +/* xmit_fixed() / recv_fixed() + * + * Transmit/receive data on a "fixed" pipe - i.e, one whose contents are not + * expected to change much, and which we don't need to buffer. + * The DBRI only interrupts us when the data changes (receive pipes), + * or only changes the data when this function is called (transmit pipes). + * Only short pipes (numbers 16-31) can be used in fixed data mode. + * + * These function operate on a 32-bit field, no matter how large + * the actual time slot is. The interrupt handler takes care of bit + * ordering and alignment. An 8-bit time slot will always end up + * in the low-order 8 bits, filled either MSB-first or LSB-first, + * depending on the settings passed to setup_pipe() + */ +static void xmit_fixed(snd_dbri_t * dbri, int pipe, unsigned int data) +{ + volatile s32 *cmd; + + if (pipe < 16 || pipe > 31) { + printk("DBRI: xmit_fixed: Illegal pipe number\n"); + return; + } + + if (D_SDP_MODE(dbri->pipes[pipe].sdp) == 0) { + printk("DBRI: xmit_fixed: Uninitialized pipe %d\n", pipe); + return; + } + + if (D_SDP_MODE(dbri->pipes[pipe].sdp) != D_SDP_FIXED) { + printk("DBRI: xmit_fixed: Non-fixed pipe %d\n", pipe); + return; + } + + if (!(dbri->pipes[pipe].sdp & D_SDP_TO_SER)) { + printk("DBRI: xmit_fixed: Called on receive pipe %d\n", pipe); + return; + } + + /* DBRI short pipes always transmit LSB first */ + + if (dbri->pipes[pipe].sdp & D_SDP_MSB) + data = reverse_bytes(data, dbri->pipes[pipe].length); + + cmd = dbri_cmdlock(dbri, GetLock); + + *(cmd++) = DBRI_CMD(D_SSP, 0, pipe); + *(cmd++) = data; + + dbri_cmdsend(dbri, cmd); +} + +static void recv_fixed(snd_dbri_t * dbri, int pipe, volatile __u32 * ptr) +{ + if (pipe < 16 || pipe > 31) { + printk("DBRI: recv_fixed called with illegal pipe number\n"); + return; + } + + if (D_SDP_MODE(dbri->pipes[pipe].sdp) != D_SDP_FIXED) { + printk("DBRI: recv_fixed called on non-fixed pipe %d\n", pipe); + return; + } + + if (dbri->pipes[pipe].sdp & D_SDP_TO_SER) { + printk("DBRI: recv_fixed called on transmit pipe %d\n", pipe); + return; + } + + dbri->pipes[pipe].recv_fixed_ptr = ptr; +} + +/* setup_descs() + * + * Setup transmit/receive data on a "long" pipe - i.e, one associated + * with a DMA buffer. + * + * Only pipe numbers 0-15 can be used in this mode. + * + * This function takes a stream number pointing to a data buffer, + * and work by building chains of descriptors which identify the + * data buffers. Buffers too large for a single descriptor will + * be spread across multiple descriptors. + */ +static int setup_descs(snd_dbri_t * dbri, int streamno, unsigned int period) +{ + dbri_streaminfo_t *info = &dbri->stream_info[streamno]; + __u32 dvma_buffer; + int desc = 0; + int len; + int first_desc = -1; + int last_desc = -1; + + if (info->pipe < 0 || info->pipe > 15) { + printk("DBRI: setup_descs: Illegal pipe number\n"); + return -2; + } + + if (dbri->pipes[info->pipe].sdp == 0) { + printk("DBRI: setup_descs: Uninitialized pipe %d\n", + info->pipe); + return -2; + } + + dvma_buffer = info->dvma_buffer; + len = info->size; + + if (streamno == DBRI_PLAY) { + if (!(dbri->pipes[info->pipe].sdp & D_SDP_TO_SER)) { + printk("DBRI: setup_descs: Called on receive pipe %d\n", + info->pipe); + return -2; + } + } else { + if (dbri->pipes[info->pipe].sdp & D_SDP_TO_SER) { + printk + ("DBRI: setup_descs: Called on transmit pipe %d\n", + info->pipe); + return -2; + } + /* Should be able to queue multiple buffers to receive on a pipe */ + if (pipe_active(dbri, info->pipe)) { + printk("DBRI: recv_on_pipe: Called on active pipe %d\n", + info->pipe); + return -2; + } + + /* Make sure buffer size is multiple of four */ + len &= ~3; + } + + while (len > 0) { + int mylen; + + for (; desc < DBRI_NO_DESCS; desc++) { + if (!dbri->descs[desc].inuse) + break; + } + if (desc == DBRI_NO_DESCS) { + printk("DBRI: setup_descs: No descriptors\n"); + return -1; + } + + if (len > DBRI_TD_MAXCNT) { + mylen = DBRI_TD_MAXCNT; /* 8KB - 1 */ + } else { + mylen = len; + } + if (mylen > period) { + mylen = period; + } + + dbri->descs[desc].inuse = 1; + dbri->descs[desc].next = -1; + dbri->dma->desc[desc].ba = dvma_buffer; + dbri->dma->desc[desc].nda = 0; + + if (streamno == DBRI_PLAY) { + dbri->descs[desc].len = mylen; + dbri->dma->desc[desc].word1 = DBRI_TD_CNT(mylen); + dbri->dma->desc[desc].word4 = 0; + if (first_desc != -1) + dbri->dma->desc[desc].word1 |= DBRI_TD_M; + } else { + dbri->descs[desc].len = 0; + dbri->dma->desc[desc].word1 = 0; + dbri->dma->desc[desc].word4 = + DBRI_RD_B | DBRI_RD_BCNT(mylen); + } + + if (first_desc == -1) { + first_desc = desc; + } else { + dbri->descs[last_desc].next = desc; + dbri->dma->desc[last_desc].nda = + dbri->dma_dvma + dbri_dma_off(desc, desc); + } + + last_desc = desc; + dvma_buffer += mylen; + len -= mylen; + } + + if (first_desc == -1 || last_desc == -1) { + printk("DBRI: setup_descs: Not enough descriptors available\n"); + return -1; + } + + dbri->dma->desc[last_desc].word1 &= ~DBRI_TD_M; + if (streamno == DBRI_PLAY) { + dbri->dma->desc[last_desc].word1 |= + DBRI_TD_I | DBRI_TD_F | DBRI_TD_B; + } + dbri->pipes[info->pipe].first_desc = first_desc; + dbri->pipes[info->pipe].desc = first_desc; + + for (desc = first_desc; desc != -1; desc = dbri->descs[desc].next) { + dprintk(D_DESC, "DESC %d: %08x %08x %08x %08x\n", + desc, + dbri->dma->desc[desc].word1, + dbri->dma->desc[desc].ba, + dbri->dma->desc[desc].nda, dbri->dma->desc[desc].word4); + } + return 0; +} + +/* +**************************************************************************** +************************** DBRI - CHI interface **************************** +**************************************************************************** + +The CHI is a four-wire (clock, frame sync, data in, data out) time-division +multiplexed serial interface which the DBRI can operate in either master +(give clock/frame sync) or slave (take clock/frame sync) mode. + +*/ + +enum master_or_slave { CHImaster, CHIslave }; + +static void reset_chi(snd_dbri_t * dbri, enum master_or_slave master_or_slave, + int bits_per_frame) +{ + volatile s32 *cmd; + int val; + static int chi_initialized = 0; /* FIXME: mutex? */ + + if (!chi_initialized) { + + cmd = dbri_cmdlock(dbri, GetLock); + + /* Set CHI Anchor: Pipe 16 */ + + val = D_DTS_VI | D_DTS_INS | D_DTS_PRVIN(16) | D_PIPE(16); + *(cmd++) = DBRI_CMD(D_DTS, 0, val); + *(cmd++) = D_TS_ANCHOR | D_TS_NEXT(16); + *(cmd++) = 0; + + val = D_DTS_VO | D_DTS_INS | D_DTS_PRVOUT(16) | D_PIPE(16); + *(cmd++) = DBRI_CMD(D_DTS, 0, val); + *(cmd++) = 0; + *(cmd++) = D_TS_ANCHOR | D_TS_NEXT(16); + + dbri->pipes[16].sdp = 1; + dbri->pipes[16].nextpipe = 16; + dbri->chi_in_pipe = 16; + dbri->chi_out_pipe = 16; + +#if 0 + chi_initialized++; +#endif + } else { + int pipe; + + for (pipe = dbri->chi_in_pipe; + pipe != 16; pipe = dbri->pipes[pipe].nextpipe) { + unlink_time_slot(dbri, pipe, PIPEinput, + 16, dbri->pipes[pipe].nextpipe); + } + for (pipe = dbri->chi_out_pipe; + pipe != 16; pipe = dbri->pipes[pipe].nextpipe) { + unlink_time_slot(dbri, pipe, PIPEoutput, + 16, dbri->pipes[pipe].nextpipe); + } + + dbri->chi_in_pipe = 16; + dbri->chi_out_pipe = 16; + + cmd = dbri_cmdlock(dbri, GetLock); + } + + if (master_or_slave == CHIslave) { + /* Setup DBRI for CHI Slave - receive clock, frame sync (FS) + * + * CHICM = 0 (slave mode, 8 kHz frame rate) + * IR = give immediate CHI status interrupt + * EN = give CHI status interrupt upon change + */ + *(cmd++) = DBRI_CMD(D_CHI, 0, D_CHI_CHICM(0)); + } else { + /* Setup DBRI for CHI Master - generate clock, FS + * + * BPF = bits per 8 kHz frame + * 12.288 MHz / CHICM_divisor = clock rate + * FD = 1 - drive CHIFS on rising edge of CHICK + */ + int clockrate = bits_per_frame * 8; + int divisor = 12288 / clockrate; + + if (divisor > 255 || divisor * clockrate != 12288) + printk("DBRI: illegal bits_per_frame in setup_chi\n"); + + *(cmd++) = DBRI_CMD(D_CHI, 0, D_CHI_CHICM(divisor) | D_CHI_FD + | D_CHI_BPF(bits_per_frame)); + } + + dbri->chi_bpf = bits_per_frame; + + /* CHI Data Mode + * + * RCE = 0 - receive on falling edge of CHICK + * XCE = 1 - transmit on rising edge of CHICK + * XEN = 1 - enable transmitter + * REN = 1 - enable receiver + */ + + *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0); + *(cmd++) = DBRI_CMD(D_CDM, 0, D_CDM_XCE | D_CDM_XEN | D_CDM_REN); + + dbri_cmdsend(dbri, cmd); +} + +/* +**************************************************************************** +*********************** CS4215 audio codec management ********************** +**************************************************************************** + +In the standard SPARC audio configuration, the CS4215 codec is attached +to the DBRI via the CHI interface and few of the DBRI's PIO pins. + +*/ +static void cs4215_setup_pipes(snd_dbri_t * dbri) +{ + /* + * Data mode: + * Pipe 4: Send timeslots 1-4 (audio data) + * Pipe 20: Send timeslots 5-8 (part of ctrl data) + * Pipe 6: Receive timeslots 1-4 (audio data) + * Pipe 21: Receive timeslots 6-7. We can only receive 20 bits via + * interrupt, and the rest of the data (slot 5 and 8) is + * not relevant for us (only for doublechecking). + * + * Control mode: + * Pipe 17: Send timeslots 1-4 (slots 5-8 are readonly) + * Pipe 18: Receive timeslot 1 (clb). + * Pipe 19: Receive timeslot 7 (version). + */ + + setup_pipe(dbri, 4, D_SDP_MEM | D_SDP_TO_SER | D_SDP_MSB); + setup_pipe(dbri, 20, D_SDP_FIXED | D_SDP_TO_SER | D_SDP_MSB); + setup_pipe(dbri, 6, D_SDP_MEM | D_SDP_FROM_SER | D_SDP_MSB); + setup_pipe(dbri, 21, D_SDP_FIXED | D_SDP_FROM_SER | D_SDP_MSB); + + setup_pipe(dbri, 17, D_SDP_FIXED | D_SDP_TO_SER | D_SDP_MSB); + setup_pipe(dbri, 18, D_SDP_FIXED | D_SDP_FROM_SER | D_SDP_MSB); + setup_pipe(dbri, 19, D_SDP_FIXED | D_SDP_FROM_SER | D_SDP_MSB); +} + +static int cs4215_init_data(struct cs4215 *mm) +{ + /* + * No action, memory resetting only. + * + * Data Time Slot 5-8 + * Speaker,Line and Headphone enable. Gain set to the half. + * Input is mike. + */ + mm->data[0] = CS4215_LO(0x20) | CS4215_HE | CS4215_LE; + mm->data[1] = CS4215_RO(0x20) | CS4215_SE; + mm->data[2] = CS4215_LG(0x8) | CS4215_IS | CS4215_PIO0 | CS4215_PIO1; + mm->data[3] = CS4215_RG(0x8) | CS4215_MA(0xf); + + /* + * Control Time Slot 1-4 + * 0: Default I/O voltage scale + * 1: 8 bit ulaw, 8kHz, mono, high pass filter disabled + * 2: Serial enable, CHI master, 128 bits per frame, clock 1 + * 3: Tests disabled + */ + mm->ctrl[0] = CS4215_RSRVD_1 | CS4215_MLB; + mm->ctrl[1] = CS4215_DFR_ULAW | CS4215_FREQ[0].csval; + mm->ctrl[2] = CS4215_XCLK | CS4215_BSEL_128 | CS4215_FREQ[0].xtal; + mm->ctrl[3] = 0; + + mm->status = 0; + mm->version = 0xff; + mm->precision = 8; /* For ULAW */ + mm->channels = 2; + + return 0; +} + +static void cs4215_setdata(snd_dbri_t * dbri, int muted) +{ + if (muted) { + dbri->mm.data[0] |= 63; + dbri->mm.data[1] |= 63; + dbri->mm.data[2] &= ~15; + dbri->mm.data[3] &= ~15; + } else { + /* Start by setting the playback attenuation. */ + dbri_streaminfo_t *info = &dbri->stream_info[DBRI_PLAY]; + int left_gain = info->left_gain % 64; + int right_gain = info->right_gain % 64; + + if (info->balance < DBRI_MID_BALANCE) { + right_gain *= info->balance; + right_gain /= DBRI_MID_BALANCE; + } else { + left_gain *= DBRI_RIGHT_BALANCE - info->balance; + left_gain /= DBRI_MID_BALANCE; + } + + dbri->mm.data[0] &= ~0x3f; /* Reset the volume bits */ + dbri->mm.data[1] &= ~0x3f; + dbri->mm.data[0] |= (DBRI_MAX_VOLUME - left_gain); + dbri->mm.data[1] |= (DBRI_MAX_VOLUME - right_gain); + + /* Now set the recording gain. */ + info = &dbri->stream_info[DBRI_REC]; + left_gain = info->left_gain % 16; + right_gain = info->right_gain % 16; + dbri->mm.data[2] |= CS4215_LG(left_gain); + dbri->mm.data[3] |= CS4215_RG(right_gain); + } + + xmit_fixed(dbri, 20, *(int *)dbri->mm.data); +} + +/* + * Set the CS4215 to data mode. + */ +static void cs4215_open(snd_dbri_t * dbri) +{ + int data_width; + u32 tmp; + + dprintk(D_MM, "cs4215_open: %d channels, %d bits\n", + dbri->mm.channels, dbri->mm.precision); + + /* Temporarily mute outputs, and wait 1/8000 sec (125 us) + * to make sure this takes. This avoids clicking noises. + */ + + cs4215_setdata(dbri, 1); + udelay(125); + + /* + * Data mode: + * Pipe 4: Send timeslots 1-4 (audio data) + * Pipe 20: Send timeslots 5-8 (part of ctrl data) + * Pipe 6: Receive timeslots 1-4 (audio data) + * Pipe 21: Receive timeslots 6-7. We can only receive 20 bits via + * interrupt, and the rest of the data (slot 5 and 8) is + * not relevant for us (only for doublechecking). + * + * Just like in control mode, the time slots are all offset by eight + * bits. The CS4215, it seems, observes TSIN (the delayed signal) + * even if it's the CHI master. Don't ask me... + */ + tmp = sbus_readl(dbri->regs + REG0); + tmp &= ~(D_C); /* Disable CHI */ + sbus_writel(tmp, dbri->regs + REG0); + + /* Switch CS4215 to data mode - set PIO3 to 1 */ + sbus_writel(D_ENPIO | D_PIO1 | D_PIO3 | + (dbri->mm.onboard ? D_PIO0 : D_PIO2), dbri->regs + REG2); + + reset_chi(dbri, CHIslave, 128); + + /* Note: this next doesn't work for 8-bit stereo, because the two + * channels would be on timeslots 1 and 3, with 2 and 4 idle. + * (See CS4215 datasheet Fig 15) + * + * DBRI non-contiguous mode would be required to make this work. + */ + data_width = dbri->mm.channels * dbri->mm.precision; + + link_time_slot(dbri, 20, PIPEoutput, 16, 32, dbri->mm.offset + 32); + link_time_slot(dbri, 4, PIPEoutput, 16, data_width, dbri->mm.offset); + link_time_slot(dbri, 6, PIPEinput, 16, data_width, dbri->mm.offset); + link_time_slot(dbri, 21, PIPEinput, 16, 16, dbri->mm.offset + 40); + + /* FIXME: enable CHI after _setdata? */ + tmp = sbus_readl(dbri->regs + REG0); + tmp |= D_C; /* Enable CHI */ + sbus_writel(tmp, dbri->regs + REG0); + + cs4215_setdata(dbri, 0); +} + +/* + * Send the control information (i.e. audio format) + */ +static int cs4215_setctrl(snd_dbri_t * dbri) +{ + int i, val; + u32 tmp; + + /* FIXME - let the CPU do something useful during these delays */ + + /* Temporarily mute outputs, and wait 1/8000 sec (125 us) + * to make sure this takes. This avoids clicking noises. + */ + + cs4215_setdata(dbri, 1); + udelay(125); + + /* + * Enable Control mode: Set DBRI's PIO3 (4215's D/~C) to 0, then wait + * 12 cycles <= 12/(5512.5*64) sec = 34.01 usec + */ + val = D_ENPIO | D_PIO1 | (dbri->mm.onboard ? D_PIO0 : D_PIO2); + sbus_writel(val, dbri->regs + REG2); + dprintk(D_MM, "cs4215_setctrl: reg2=0x%x\n", val); + udelay(34); + + /* In Control mode, the CS4215 is a slave device, so the DBRI must + * operate as CHI master, supplying clocking and frame synchronization. + * + * In Data mode, however, the CS4215 must be CHI master to insure + * that its data stream is synchronous with its codec. + * + * The upshot of all this? We start by putting the DBRI into master + * mode, program the CS4215 in Control mode, then switch the CS4215 + * into Data mode and put the DBRI into slave mode. Various timing + * requirements must be observed along the way. + * + * Oh, and one more thing, on a SPARCStation 20 (and maybe + * others?), the addressing of the CS4215's time slots is + * offset by eight bits, so we add eight to all the "cycle" + * values in the Define Time Slot (DTS) commands. This is + * done in hardware by a TI 248 that delays the DBRI->4215 + * frame sync signal by eight clock cycles. Anybody know why? + */ + tmp = sbus_readl(dbri->regs + REG0); + tmp &= ~D_C; /* Disable CHI */ + sbus_writel(tmp, dbri->regs + REG0); + + reset_chi(dbri, CHImaster, 128); + + /* + * Control mode: + * Pipe 17: Send timeslots 1-4 (slots 5-8 are readonly) + * Pipe 18: Receive timeslot 1 (clb). + * Pipe 19: Receive timeslot 7 (version). + */ + + link_time_slot(dbri, 17, PIPEoutput, 16, 32, dbri->mm.offset); + link_time_slot(dbri, 18, PIPEinput, 16, 8, dbri->mm.offset); + link_time_slot(dbri, 19, PIPEinput, 16, 8, dbri->mm.offset + 48); + + /* Wait for the chip to echo back CLB (Control Latch Bit) as zero */ + dbri->mm.ctrl[0] &= ~CS4215_CLB; + xmit_fixed(dbri, 17, *(int *)dbri->mm.ctrl); + + tmp = sbus_readl(dbri->regs + REG0); + tmp |= D_C; /* Enable CHI */ + sbus_writel(tmp, dbri->regs + REG0); + + for (i = 64; ((dbri->mm.status & 0xe4) != 0x20); --i) { + udelay(125); + } + if (i == 0) { + dprintk(D_MM, "CS4215 didn't respond to CLB (0x%02x)\n", + dbri->mm.status); + return -1; + } + + /* Disable changes to our copy of the version number, as we are about + * to leave control mode. + */ + recv_fixed(dbri, 19, NULL); + + /* Terminate CS4215 control mode - data sheet says + * "Set CLB=1 and send two more frames of valid control info" + */ + dbri->mm.ctrl[0] |= CS4215_CLB; + xmit_fixed(dbri, 17, *(int *)dbri->mm.ctrl); + + /* Two frames of control info @ 8kHz frame rate = 250 us delay */ + udelay(250); + + cs4215_setdata(dbri, 0); + + return 0; +} + +/* + * Setup the codec with the sampling rate, audio format and number of + * channels. + * As part of the process we resend the settings for the data + * timeslots as well. + */ +static int cs4215_prepare(snd_dbri_t * dbri, unsigned int rate, + snd_pcm_format_t format, unsigned int channels) +{ + int freq_idx; + int ret = 0; + + /* Lookup index for this rate */ + for (freq_idx = 0; CS4215_FREQ[freq_idx].freq != 0; freq_idx++) { + if (CS4215_FREQ[freq_idx].freq == rate) + break; + } + if (CS4215_FREQ[freq_idx].freq != rate) { + printk(KERN_WARNING "DBRI: Unsupported rate %d Hz\n", rate); + return -1; + } + + switch (format) { + case SNDRV_PCM_FORMAT_MU_LAW: + dbri->mm.ctrl[1] = CS4215_DFR_ULAW; + dbri->mm.precision = 8; + break; + case SNDRV_PCM_FORMAT_A_LAW: + dbri->mm.ctrl[1] = CS4215_DFR_ALAW; + dbri->mm.precision = 8; + break; + case SNDRV_PCM_FORMAT_U8: + dbri->mm.ctrl[1] = CS4215_DFR_LINEAR8; + dbri->mm.precision = 8; + break; + case SNDRV_PCM_FORMAT_S16_BE: + dbri->mm.ctrl[1] = CS4215_DFR_LINEAR16; + dbri->mm.precision = 16; + break; + default: + printk(KERN_WARNING "DBRI: Unsupported format %d\n", format); + return -1; + } + + /* Add rate parameters */ + dbri->mm.ctrl[1] |= CS4215_FREQ[freq_idx].csval; + dbri->mm.ctrl[2] = CS4215_XCLK | + CS4215_BSEL_128 | CS4215_FREQ[freq_idx].xtal; + + dbri->mm.channels = channels; + /* Stereo bit: 8 bit stereo not working yet. */ + if ((channels > 1) && (dbri->mm.precision == 16)) + dbri->mm.ctrl[1] |= CS4215_DFR_STEREO; + + ret = cs4215_setctrl(dbri); + if (ret == 0) + cs4215_open(dbri); /* set codec to data mode */ + + return ret; +} + +/* + * + */ +static int cs4215_init(snd_dbri_t * dbri) +{ + u32 reg2 = sbus_readl(dbri->regs + REG2); + dprintk(D_MM, "cs4215_init: reg2=0x%x\n", reg2); + + /* Look for the cs4215 chips */ + if (reg2 & D_PIO2) { + dprintk(D_MM, "Onboard CS4215 detected\n"); + dbri->mm.onboard = 1; + } + if (reg2 & D_PIO0) { + dprintk(D_MM, "Speakerbox detected\n"); + dbri->mm.onboard = 0; + + if (reg2 & D_PIO2) { + printk(KERN_INFO "DBRI: Using speakerbox / " + "ignoring onboard mmcodec.\n"); + sbus_writel(D_ENPIO2, dbri->regs + REG2); + } + } + + if (!(reg2 & (D_PIO0 | D_PIO2))) { + printk(KERN_ERR "DBRI: no mmcodec found.\n"); + return -EIO; + } + + cs4215_setup_pipes(dbri); + + cs4215_init_data(&dbri->mm); + + /* Enable capture of the status & version timeslots. */ + recv_fixed(dbri, 18, &dbri->mm.status); + recv_fixed(dbri, 19, &dbri->mm.version); + + dbri->mm.offset = dbri->mm.onboard ? 0 : 8; + if (cs4215_setctrl(dbri) == -1 || dbri->mm.version == 0xff) { + dprintk(D_MM, "CS4215 failed probe at offset %d\n", + dbri->mm.offset); + return -EIO; + } + dprintk(D_MM, "Found CS4215 at offset %d\n", dbri->mm.offset); + + return 0; +} + +/* +**************************************************************************** +*************************** DBRI interrupt handler ************************* +**************************************************************************** + +The DBRI communicates with the CPU mainly via a circular interrupt +buffer. When an interrupt is signaled, the CPU walks through the +buffer and calls dbri_process_one_interrupt() for each interrupt word. +Complicated interrupts are handled by dedicated functions (which +appear first in this file). Any pending interrupts can be serviced by +calling dbri_process_interrupt_buffer(), which works even if the CPU's +interrupts are disabled. This function is used by dbri_cmdsend() +to make sure we're synced up with the chip after each command sequence, +even if we're running cli'ed. + +*/ + +/* xmit_descs() + * + * Transmit the current TD's for recording/playing, if needed. + * For playback, ALSA has filled the DMA memory with new data (we hope). + */ +static void xmit_descs(unsigned long data) +{ + snd_dbri_t *dbri = (snd_dbri_t *) data; + dbri_streaminfo_t *info; + volatile s32 *cmd; + unsigned long flags; + int first_td; + + if (dbri == NULL) + return; /* Disabled */ + + /* First check the recording stream for buffer overflow */ + info = &dbri->stream_info[DBRI_REC]; + spin_lock_irqsave(&dbri->lock, flags); + + if ((info->left >= info->size) && (info->pipe >= 0)) { + first_td = dbri->pipes[info->pipe].first_desc; + + dprintk(D_DESC, "xmit_descs rec @ TD %d\n", first_td); + + /* Stream could be closed by the time we run. */ + if (first_td < 0) { + goto play; + } + + cmd = dbri_cmdlock(dbri, NoGetLock); + *(cmd++) = DBRI_CMD(D_SDP, 0, + dbri->pipes[info->pipe].sdp + | D_SDP_P | D_SDP_EVERY | D_SDP_C); + *(cmd++) = dbri->dma_dvma + dbri_dma_off(desc, first_td); + dbri_cmdsend(dbri, cmd); + + /* Reset our admin of the pipe & bytes read. */ + dbri->pipes[info->pipe].desc = first_td; + info->left = 0; + } + +play: + spin_unlock_irqrestore(&dbri->lock, flags); + + /* Now check the playback stream for buffer underflow */ + info = &dbri->stream_info[DBRI_PLAY]; + spin_lock_irqsave(&dbri->lock, flags); + + if ((info->left <= 0) && (info->pipe >= 0)) { + first_td = dbri->pipes[info->pipe].first_desc; + + dprintk(D_DESC, "xmit_descs play @ TD %d\n", first_td); + + /* Stream could be closed by the time we run. */ + if (first_td < 0) { + spin_unlock_irqrestore(&dbri->lock, flags); + return; + } + + cmd = dbri_cmdlock(dbri, NoGetLock); + *(cmd++) = DBRI_CMD(D_SDP, 0, + dbri->pipes[info->pipe].sdp + | D_SDP_P | D_SDP_EVERY | D_SDP_C); + *(cmd++) = dbri->dma_dvma + dbri_dma_off(desc, first_td); + dbri_cmdsend(dbri, cmd); + + /* Reset our admin of the pipe & bytes written. */ + dbri->pipes[info->pipe].desc = first_td; + info->left = info->size; + } + spin_unlock_irqrestore(&dbri->lock, flags); +} + +DECLARE_TASKLET(xmit_descs_task, xmit_descs, 0); + +/* transmission_complete_intr() + * + * Called by main interrupt handler when DBRI signals transmission complete + * on a pipe (interrupt triggered by the B bit in a transmit descriptor). + * + * Walks through the pipe's list of transmit buffer descriptors, releasing + * each one's DMA buffer (if present), flagging the descriptor available, + * and signaling its callback routine (if present), before proceeding + * to the next one. Stops when the first descriptor is found without + * TBC (Transmit Buffer Complete) set, or we've run through them all. + */ + +static void transmission_complete_intr(snd_dbri_t * dbri, int pipe) +{ + dbri_streaminfo_t *info; + int td; + int status; + + info = &dbri->stream_info[DBRI_PLAY]; + + td = dbri->pipes[pipe].desc; + while (td >= 0) { + if (td >= DBRI_NO_DESCS) { + printk(KERN_ERR "DBRI: invalid td on pipe %d\n", pipe); + return; + } + + status = DBRI_TD_STATUS(dbri->dma->desc[td].word4); + if (!(status & DBRI_TD_TBC)) { + break; + } + + dprintk(D_INT, "TD %d, status 0x%02x\n", td, status); + + dbri->dma->desc[td].word4 = 0; /* Reset it for next time. */ + info->offset += dbri->descs[td].len; + info->left -= dbri->descs[td].len; + + /* On the last TD, transmit them all again. */ + if (dbri->descs[td].next == -1) { + if (info->left > 0) { + printk(KERN_WARNING + "%d bytes left after last transfer.\n", + info->left); + info->left = 0; + } + tasklet_schedule(&xmit_descs_task); + } + + td = dbri->descs[td].next; + dbri->pipes[pipe].desc = td; + } + + /* Notify ALSA */ + if (spin_is_locked(&dbri->lock)) { + spin_unlock(&dbri->lock); + snd_pcm_period_elapsed(info->substream); + spin_lock(&dbri->lock); + } else + snd_pcm_period_elapsed(info->substream); +} + +static void reception_complete_intr(snd_dbri_t * dbri, int pipe) +{ + dbri_streaminfo_t *info; + int rd = dbri->pipes[pipe].desc; + s32 status; + + if (rd < 0 || rd >= DBRI_NO_DESCS) { + printk(KERN_ERR "DBRI: invalid rd on pipe %d\n", pipe); + return; + } + + dbri->descs[rd].inuse = 0; + dbri->pipes[pipe].desc = dbri->descs[rd].next; + status = dbri->dma->desc[rd].word1; + dbri->dma->desc[rd].word1 = 0; /* Reset it for next time. */ + + info = &dbri->stream_info[DBRI_REC]; + info->offset += DBRI_RD_CNT(status); + info->left += DBRI_RD_CNT(status); + + /* FIXME: Check status */ + + dprintk(D_INT, "Recv RD %d, status 0x%02x, len %d\n", + rd, DBRI_RD_STATUS(status), DBRI_RD_CNT(status)); + + /* On the last TD, transmit them all again. */ + if (dbri->descs[rd].next == -1) { + if (info->left > info->size) { + printk(KERN_WARNING + "%d bytes recorded in %d size buffer.\n", + info->left, info->size); + } + tasklet_schedule(&xmit_descs_task); + } + + /* Notify ALSA */ + if (spin_is_locked(&dbri->lock)) { + spin_unlock(&dbri->lock); + snd_pcm_period_elapsed(info->substream); + spin_lock(&dbri->lock); + } else + snd_pcm_period_elapsed(info->substream); +} + +static void dbri_process_one_interrupt(snd_dbri_t * dbri, int x) +{ + int val = D_INTR_GETVAL(x); + int channel = D_INTR_GETCHAN(x); + int command = D_INTR_GETCMD(x); + int code = D_INTR_GETCODE(x); +#ifdef DBRI_DEBUG + int rval = D_INTR_GETRVAL(x); +#endif + + if (channel == D_INTR_CMD) { + dprintk(D_CMD, "INTR: Command: %-5s Value:%d\n", + cmds[command], val); + } else { + dprintk(D_INT, "INTR: Chan:%d Code:%d Val:%#x\n", + channel, code, rval); + } + + if (channel == D_INTR_CMD && command == D_WAIT) { + dbri->wait_seen++; + return; + } + + switch (code) { + case D_INTR_BRDY: + reception_complete_intr(dbri, channel); + break; + case D_INTR_XCMP: + case D_INTR_MINT: + transmission_complete_intr(dbri, channel); + break; + case D_INTR_UNDR: + /* UNDR - Transmission underrun + * resend SDP command with clear pipe bit (C) set + */ + { + volatile s32 *cmd; + + int pipe = channel; + int td = dbri->pipes[pipe].desc; + + dbri->dma->desc[td].word4 = 0; + cmd = dbri_cmdlock(dbri, NoGetLock); + *(cmd++) = DBRI_CMD(D_SDP, 0, + dbri->pipes[pipe].sdp + | D_SDP_P | D_SDP_C | D_SDP_2SAME); + *(cmd++) = dbri->dma_dvma + dbri_dma_off(desc, td); + dbri_cmdsend(dbri, cmd); + } + break; + case D_INTR_FXDT: + /* FXDT - Fixed data change */ + if (dbri->pipes[channel].sdp & D_SDP_MSB) + val = reverse_bytes(val, dbri->pipes[channel].length); + + if (dbri->pipes[channel].recv_fixed_ptr) + *(dbri->pipes[channel].recv_fixed_ptr) = val; + break; + default: + if (channel != D_INTR_CMD) + printk(KERN_WARNING + "DBRI: Ignored Interrupt: %d (0x%x)\n", code, x); + } +} + +/* dbri_process_interrupt_buffer advances through the DBRI's interrupt + * buffer until it finds a zero word (indicating nothing more to do + * right now). Non-zero words require processing and are handed off + * to dbri_process_one_interrupt AFTER advancing the pointer. This + * order is important since we might recurse back into this function + * and need to make sure the pointer has been advanced first. + */ +static void dbri_process_interrupt_buffer(snd_dbri_t * dbri) +{ + s32 x; + + while ((x = dbri->dma->intr[dbri->dbri_irqp]) != 0) { + dbri->dma->intr[dbri->dbri_irqp] = 0; + dbri->dbri_irqp++; + if (dbri->dbri_irqp == (DBRI_NO_INTS * DBRI_INT_BLK)) + dbri->dbri_irqp = 1; + else if ((dbri->dbri_irqp & (DBRI_INT_BLK - 1)) == 0) + dbri->dbri_irqp++; + + dbri_process_one_interrupt(dbri, x); + } +} + +static irqreturn_t snd_dbri_interrupt(int irq, void *dev_id, + struct pt_regs *regs) +{ + snd_dbri_t *dbri = dev_id; + static int errcnt = 0; + int x; + + if (dbri == NULL) + return IRQ_NONE; + spin_lock(&dbri->lock); + + /* + * Read it, so the interrupt goes away. + */ + x = sbus_readl(dbri->regs + REG1); + + if (x & (D_MRR | D_MLE | D_LBG | D_MBE)) { + u32 tmp; + + if (x & D_MRR) + printk(KERN_ERR + "DBRI: Multiple Error Ack on SBus reg1=0x%x\n", + x); + if (x & D_MLE) + printk(KERN_ERR + "DBRI: Multiple Late Error on SBus reg1=0x%x\n", + x); + if (x & D_LBG) + printk(KERN_ERR + "DBRI: Lost Bus Grant on SBus reg1=0x%x\n", x); + if (x & D_MBE) + printk(KERN_ERR + "DBRI: Burst Error on SBus reg1=0x%x\n", x); + + /* Some of these SBus errors cause the chip's SBus circuitry + * to be disabled, so just re-enable and try to keep going. + * + * The only one I've seen is MRR, which will be triggered + * if you let a transmit pipe underrun, then try to CDP it. + * + * If these things persist, we should probably reset + * and re-init the chip. + */ + if ((++errcnt) % 10 == 0) { + dprintk(D_INT, "Interrupt errors exceeded.\n"); + dbri_reset(dbri); + } else { + tmp = sbus_readl(dbri->regs + REG0); + tmp &= ~(D_D); + sbus_writel(tmp, dbri->regs + REG0); + } + } + + dbri_process_interrupt_buffer(dbri); + + /* FIXME: Write 0 into regs to ACK interrupt */ + + spin_unlock(&dbri->lock); + + return IRQ_HANDLED; +} + +/**************************************************************************** + PCM Interface +****************************************************************************/ +static snd_pcm_hardware_t snd_dbri_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_MU_LAW | + SNDRV_PCM_FMTBIT_A_LAW | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (64 * 1024), + .period_bytes_min = 1, + .period_bytes_max = DBRI_TD_MAXCNT, + .periods_min = 1, + .periods_max = 1024, +}; + +static int snd_dbri_open(snd_pcm_substream_t * substream) +{ + snd_dbri_t *dbri = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime = substream->runtime; + dbri_streaminfo_t *info = DBRI_STREAM(dbri, substream); + unsigned long flags; + + dprintk(D_USR, "open audio output.\n"); + runtime->hw = snd_dbri_pcm_hw; + + spin_lock_irqsave(&dbri->lock, flags); + info->substream = substream; + info->left = 0; + info->offset = 0; + info->dvma_buffer = 0; + info->pipe = -1; + spin_unlock_irqrestore(&dbri->lock, flags); + + cs4215_open(dbri); + + return 0; +} + +static int snd_dbri_close(snd_pcm_substream_t * substream) +{ + snd_dbri_t *dbri = snd_pcm_substream_chip(substream); + dbri_streaminfo_t *info = DBRI_STREAM(dbri, substream); + + dprintk(D_USR, "close audio output.\n"); + info->substream = NULL; + info->left = 0; + info->offset = 0; + + return 0; +} + +static int snd_dbri_hw_params(snd_pcm_substream_t * substream, + snd_pcm_hw_params_t * hw_params) +{ + snd_pcm_runtime_t *runtime = substream->runtime; + snd_dbri_t *dbri = snd_pcm_substream_chip(substream); + dbri_streaminfo_t *info = DBRI_STREAM(dbri, substream); + int direction; + int ret; + + /* set sampling rate, audio format and number of channels */ + ret = cs4215_prepare(dbri, params_rate(hw_params), + params_format(hw_params), + params_channels(hw_params)); + if (ret != 0) + return ret; + + if ((ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params))) < 0) { + snd_printk(KERN_ERR "malloc_pages failed with %d\n", ret); + return ret; + } + + /* hw_params can get called multiple times. Only map the DMA once. + */ + if (info->dvma_buffer == 0) { + if (DBRI_STREAMNO(substream) == DBRI_PLAY) + direction = SBUS_DMA_TODEVICE; + else + direction = SBUS_DMA_FROMDEVICE; + + info->dvma_buffer = sbus_map_single(dbri->sdev, + runtime->dma_area, + params_buffer_bytes(hw_params), + direction); + } + + direction = params_buffer_bytes(hw_params); + dprintk(D_USR, "hw_params: %d bytes, dvma=%x\n", + direction, info->dvma_buffer); + return 0; +} + +static int snd_dbri_hw_free(snd_pcm_substream_t * substream) +{ + snd_dbri_t *dbri = snd_pcm_substream_chip(substream); + dbri_streaminfo_t *info = DBRI_STREAM(dbri, substream); + int direction; + dprintk(D_USR, "hw_free.\n"); + + /* hw_free can get called multiple times. Only unmap the DMA once. + */ + if (info->dvma_buffer) { + if (DBRI_STREAMNO(substream) == DBRI_PLAY) + direction = SBUS_DMA_TODEVICE; + else + direction = SBUS_DMA_FROMDEVICE; + + sbus_unmap_single(dbri->sdev, info->dvma_buffer, + substream->runtime->buffer_size, direction); + info->dvma_buffer = 0; + } + info->pipe = -1; + + return snd_pcm_lib_free_pages(substream); +} + +static int snd_dbri_prepare(snd_pcm_substream_t * substream) +{ + snd_dbri_t *dbri = snd_pcm_substream_chip(substream); + dbri_streaminfo_t *info = DBRI_STREAM(dbri, substream); + snd_pcm_runtime_t *runtime = substream->runtime; + int ret; + + info->size = snd_pcm_lib_buffer_bytes(substream); + if (DBRI_STREAMNO(substream) == DBRI_PLAY) + info->pipe = 4; /* Send pipe */ + else { + info->pipe = 6; /* Receive pipe */ + info->left = info->size; /* To trigger submittal */ + } + + spin_lock_irq(&dbri->lock); + + /* Setup the all the transmit/receive desciptors to cover the + * whole DMA buffer. + */ + ret = setup_descs(dbri, DBRI_STREAMNO(substream), + snd_pcm_lib_period_bytes(substream)); + + runtime->stop_threshold = DBRI_TD_MAXCNT / runtime->channels; + + spin_unlock_irq(&dbri->lock); + + dprintk(D_USR, "prepare audio output. %d bytes\n", info->size); + return ret; +} + +static int snd_dbri_trigger(snd_pcm_substream_t * substream, int cmd) +{ + snd_dbri_t *dbri = snd_pcm_substream_chip(substream); + dbri_streaminfo_t *info = DBRI_STREAM(dbri, substream); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dprintk(D_USR, "start audio, period is %d bytes\n", + (int)snd_pcm_lib_period_bytes(substream)); + /* Enable & schedule the tasklet that re-submits the TDs. */ + xmit_descs_task.data = (unsigned long)dbri; + tasklet_schedule(&xmit_descs_task); + break; + case SNDRV_PCM_TRIGGER_STOP: + dprintk(D_USR, "stop audio.\n"); + /* Make the tasklet bail out immediately. */ + xmit_descs_task.data = 0; + reset_pipe(dbri, info->pipe); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t snd_dbri_pointer(snd_pcm_substream_t * substream) +{ + snd_dbri_t *dbri = snd_pcm_substream_chip(substream); + dbri_streaminfo_t *info = DBRI_STREAM(dbri, substream); + snd_pcm_uframes_t ret; + + ret = bytes_to_frames(substream->runtime, info->offset) + % substream->runtime->buffer_size; + dprintk(D_USR, "I/O pointer: %ld frames, %d bytes left.\n", + ret, info->left); + return ret; +} + +static snd_pcm_ops_t snd_dbri_ops = { + .open = snd_dbri_open, + .close = snd_dbri_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_dbri_hw_params, + .hw_free = snd_dbri_hw_free, + .prepare = snd_dbri_prepare, + .trigger = snd_dbri_trigger, + .pointer = snd_dbri_pointer, +}; + +static int __devinit snd_dbri_pcm(snd_dbri_t * dbri) +{ + snd_pcm_t *pcm; + int err; + + if ((err = snd_pcm_new(dbri->card, + /* ID */ "sun_dbri", + /* device */ 0, + /* playback count */ 1, + /* capture count */ 1, &pcm)) < 0) + return err; + snd_assert(pcm != NULL, return -EINVAL); + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_dbri_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_dbri_ops); + + pcm->private_data = dbri; + pcm->info_flags = 0; + strcpy(pcm->name, dbri->card->shortname); + dbri->pcm = pcm; + + if ((err = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 64 * 1024, 64 * 1024)) < 0) { + return err; + } + + return 0; +} + +/***************************************************************************** + Mixer interface +*****************************************************************************/ + +static int snd_cs4215_info_volume(snd_kcontrol_t * kcontrol, + snd_ctl_elem_info_t * uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + if (kcontrol->private_value == DBRI_PLAY) { + uinfo->value.integer.max = DBRI_MAX_VOLUME; + } else { + uinfo->value.integer.max = DBRI_MAX_GAIN; + } + return 0; +} + +static int snd_cs4215_get_volume(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + snd_dbri_t *dbri = snd_kcontrol_chip(kcontrol); + dbri_streaminfo_t *info; + snd_assert(dbri != NULL, return -EINVAL); + info = &dbri->stream_info[kcontrol->private_value]; + snd_assert(info != NULL, return -EINVAL); + + ucontrol->value.integer.value[0] = info->left_gain; + ucontrol->value.integer.value[1] = info->right_gain; + return 0; +} + +static int snd_cs4215_put_volume(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + snd_dbri_t *dbri = snd_kcontrol_chip(kcontrol); + dbri_streaminfo_t *info = &dbri->stream_info[kcontrol->private_value]; + unsigned long flags; + int changed = 0; + + if (info->left_gain != ucontrol->value.integer.value[0]) { + info->left_gain = ucontrol->value.integer.value[0]; + changed = 1; + } + if (info->right_gain != ucontrol->value.integer.value[1]) { + info->right_gain = ucontrol->value.integer.value[1]; + changed = 1; + } + if (changed == 1) { + /* First mute outputs, and wait 1/8000 sec (125 us) + * to make sure this takes. This avoids clicking noises. + */ + spin_lock_irqsave(&dbri->lock, flags); + + cs4215_setdata(dbri, 1); + udelay(125); + cs4215_setdata(dbri, 0); + + spin_unlock_irqrestore(&dbri->lock, flags); + } + return changed; +} + +static int snd_cs4215_info_single(snd_kcontrol_t * kcontrol, + snd_ctl_elem_info_t * uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = (mask == 1) ? + SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} + +static int snd_cs4215_get_single(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + snd_dbri_t *dbri = snd_kcontrol_chip(kcontrol); + int elem = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 1; + snd_assert(dbri != NULL, return -EINVAL); + + if (elem < 4) { + ucontrol->value.integer.value[0] = + (dbri->mm.data[elem] >> shift) & mask; + } else { + ucontrol->value.integer.value[0] = + (dbri->mm.ctrl[elem - 4] >> shift) & mask; + } + + if (invert == 1) { + ucontrol->value.integer.value[0] = + mask - ucontrol->value.integer.value[0]; + } + return 0; +} + +static int snd_cs4215_put_single(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + snd_dbri_t *dbri = snd_kcontrol_chip(kcontrol); + unsigned long flags; + int elem = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 1; + int changed = 0; + unsigned short val; + snd_assert(dbri != NULL, return -EINVAL); + + val = (ucontrol->value.integer.value[0] & mask); + if (invert == 1) + val = mask - val; + val <<= shift; + + if (elem < 4) { + dbri->mm.data[elem] = (dbri->mm.data[elem] & + ~(mask << shift)) | val; + changed = (val != dbri->mm.data[elem]); + } else { + dbri->mm.ctrl[elem - 4] = (dbri->mm.ctrl[elem - 4] & + ~(mask << shift)) | val; + changed = (val != dbri->mm.ctrl[elem - 4]); + } + + dprintk(D_GEN, "put_single: mask=0x%x, changed=%d, " + "mixer-value=%ld, mm-value=0x%x\n", + mask, changed, ucontrol->value.integer.value[0], + dbri->mm.data[elem & 3]); + + if (changed) { + /* First mute outputs, and wait 1/8000 sec (125 us) + * to make sure this takes. This avoids clicking noises. + */ + spin_lock_irqsave(&dbri->lock, flags); + + cs4215_setdata(dbri, 1); + udelay(125); + cs4215_setdata(dbri, 0); + + spin_unlock_irqrestore(&dbri->lock, flags); + } + return changed; +} + +/* Entries 0-3 map to the 4 data timeslots, entries 4-7 map to the 4 control + timeslots. Shift is the bit offset in the timeslot, mask defines the + number of bits. invert is a boolean for use with attenuation. + */ +#define CS4215_SINGLE(xname, entry, shift, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_cs4215_info_single, \ + .get = snd_cs4215_get_single, .put = snd_cs4215_put_single, \ + .private_value = entry | (shift << 8) | (mask << 16) | (invert << 24) }, + +static snd_kcontrol_new_t dbri_controls[] __devinitdata = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Playback Volume", + .info = snd_cs4215_info_volume, + .get = snd_cs4215_get_volume, + .put = snd_cs4215_put_volume, + .private_value = DBRI_PLAY, + }, + CS4215_SINGLE("Headphone switch", 0, 7, 1, 0) + CS4215_SINGLE("Line out switch", 0, 6, 1, 0) + CS4215_SINGLE("Speaker switch", 1, 6, 1, 0) + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Volume", + .info = snd_cs4215_info_volume, + .get = snd_cs4215_get_volume, + .put = snd_cs4215_put_volume, + .private_value = DBRI_REC, + }, + /* FIXME: mic/line switch */ + CS4215_SINGLE("Line in switch", 2, 4, 1, 0) + CS4215_SINGLE("High Pass Filter switch", 5, 7, 1, 0) + CS4215_SINGLE("Monitor Volume", 3, 4, 0xf, 1) + CS4215_SINGLE("Mic boost", 4, 4, 1, 1) +}; + +#define NUM_CS4215_CONTROLS (sizeof(dbri_controls)/sizeof(snd_kcontrol_new_t)) + +static int __init snd_dbri_mixer(snd_dbri_t * dbri) +{ + snd_card_t *card; + int idx, err; + + snd_assert(dbri != NULL && dbri->card != NULL, return -EINVAL); + + card = dbri->card; + strcpy(card->mixername, card->shortname); + + for (idx = 0; idx < NUM_CS4215_CONTROLS; idx++) { + if ((err = snd_ctl_add(card, + snd_ctl_new1(&dbri_controls[idx], + dbri))) < 0) + return err; + } + + for (idx = DBRI_REC; idx < DBRI_NO_STREAMS; idx++) { + dbri->stream_info[idx].left_gain = 0; + dbri->stream_info[idx].right_gain = 0; + dbri->stream_info[idx].balance = DBRI_MID_BALANCE; + } + + return 0; +} + +/**************************************************************************** + /proc interface +****************************************************************************/ +static void dbri_regs_read(snd_info_entry_t * entry, snd_info_buffer_t * buffer) +{ + snd_dbri_t *dbri = entry->private_data; + + snd_iprintf(buffer, "REG0: 0x%x\n", sbus_readl(dbri->regs + REG0)); + snd_iprintf(buffer, "REG2: 0x%x\n", sbus_readl(dbri->regs + REG2)); + snd_iprintf(buffer, "REG8: 0x%x\n", sbus_readl(dbri->regs + REG8)); + snd_iprintf(buffer, "REG9: 0x%x\n", sbus_readl(dbri->regs + REG9)); +} + +#ifdef DBRI_DEBUG +static void dbri_debug_read(snd_info_entry_t * entry, + snd_info_buffer_t * buffer) +{ + snd_dbri_t *dbri = entry->private_data; + int pipe; + snd_iprintf(buffer, "debug=%d\n", dbri_debug); + + snd_iprintf(buffer, "CHI pipe in=%d, out=%d\n", + dbri->chi_in_pipe, dbri->chi_out_pipe); + for (pipe = 0; pipe < 32; pipe++) { + if (pipe_active(dbri, pipe)) { + struct dbri_pipe *pptr = &dbri->pipes[pipe]; + snd_iprintf(buffer, + "Pipe %d: %s SDP=0x%x desc=%d, " + "len=%d @ %d prev: %d next %d\n", + pipe, + (pptr->direction == + PIPEinput ? "input" : "output"), pptr->sdp, + pptr->desc, pptr->length, pptr->cycle, + pptr->prevpipe, pptr->nextpipe); + } + } +} + +static void dbri_debug_write(snd_info_entry_t * entry, + snd_info_buffer_t * buffer) +{ + char line[80]; + int i; + + if (snd_info_get_line(buffer, line, 80) == 0) { + sscanf(line, "%d\n", &i); + dbri_debug = i & 0x3f; + } +} +#endif + +void snd_dbri_proc(snd_dbri_t * dbri) +{ + snd_info_entry_t *entry; + int err; + + err = snd_card_proc_new(dbri->card, "regs", &entry); + snd_info_set_text_ops(entry, dbri, 1024, dbri_regs_read); + +#ifdef DBRI_DEBUG + err = snd_card_proc_new(dbri->card, "debug", &entry); + snd_info_set_text_ops(entry, dbri, 4096, dbri_debug_read); + entry->mode = S_IFREG | S_IRUGO | S_IWUSR; /* Writable for root */ + entry->c.text.write_size = 256; + entry->c.text.write = dbri_debug_write; +#endif +} + +/* +**************************************************************************** +**************************** Initialization ******************************** +**************************************************************************** +*/ +static void snd_dbri_free(snd_dbri_t * dbri); + +static int __init snd_dbri_create(snd_card_t * card, + struct sbus_dev *sdev, + struct linux_prom_irqs *irq, int dev) +{ + snd_dbri_t *dbri = card->private_data; + int err; + + spin_lock_init(&dbri->lock); + dbri->card = card; + dbri->sdev = sdev; + dbri->irq = irq->pri; + dbri->dbri_version = sdev->prom_name[9]; + + dbri->dma = sbus_alloc_consistent(sdev, sizeof(struct dbri_dma), + &dbri->dma_dvma); + memset((void *)dbri->dma, 0, sizeof(struct dbri_dma)); + + dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n", + dbri->dma, dbri->dma_dvma); + + /* Map the registers into memory. */ + dbri->regs_size = sdev->reg_addrs[0].reg_size; + dbri->regs = sbus_ioremap(&sdev->resource[0], 0, + dbri->regs_size, "DBRI Registers"); + if (!dbri->regs) { + printk(KERN_ERR "DBRI: could not allocate registers\n"); + sbus_free_consistent(sdev, sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); + return -EIO; + } + + err = request_irq(dbri->irq, snd_dbri_interrupt, SA_SHIRQ, + "DBRI audio", dbri); + if (err) { + printk(KERN_ERR "DBRI: Can't get irq %d\n", dbri->irq); + sbus_iounmap(dbri->regs, dbri->regs_size); + sbus_free_consistent(sdev, sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); + return err; + } + + /* Do low level initialization of the DBRI and CS4215 chips */ + dbri_initialize(dbri); + err = cs4215_init(dbri); + if (err) { + snd_dbri_free(dbri); + return err; + } + + dbri->next = dbri_list; + dbri_list = dbri; + + return 0; +} + +static void snd_dbri_free(snd_dbri_t * dbri) +{ + dprintk(D_GEN, "snd_dbri_free\n"); + dbri_reset(dbri); + + if (dbri->irq) + free_irq(dbri->irq, dbri); + + if (dbri->regs) + sbus_iounmap(dbri->regs, dbri->regs_size); + + if (dbri->dma) + sbus_free_consistent(dbri->sdev, sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); +} + +static int __init dbri_attach(int prom_node, struct sbus_dev *sdev) +{ + snd_dbri_t *dbri; + struct linux_prom_irqs irq; + struct resource *rp; + snd_card_t *card; + static int dev = 0; + int err; + + if (sdev->prom_name[9] < 'e') { + printk(KERN_ERR "DBRI: unsupported chip version %c found.\n", + sdev->prom_name[9]); + return -EIO; + } + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + prom_getproperty(prom_node, "intr", (char *)&irq, sizeof(irq)); + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, + sizeof(snd_dbri_t)); + if (card == NULL) + return -ENOMEM; + + strcpy(card->driver, "DBRI"); + strcpy(card->shortname, "Sun DBRI"); + rp = &sdev->resource[0]; + sprintf(card->longname, "%s at 0x%02lx:0x%08lx, irq %s", + card->shortname, + rp->flags & 0xffL, rp->start, __irq_itoa(irq.pri)); + + if ((err = snd_dbri_create(card, sdev, &irq, dev)) < 0) { + snd_card_free(card); + return err; + } + + dbri = (snd_dbri_t *) card->private_data; + if ((err = snd_dbri_pcm(dbri)) < 0) { + snd_dbri_free(dbri); + snd_card_free(card); + return err; + } + + if ((err = snd_dbri_mixer(dbri)) < 0) { + snd_dbri_free(dbri); + snd_card_free(card); + return err; + } + + /* /proc file handling */ + snd_dbri_proc(dbri); + + if ((err = snd_card_register(card)) < 0) { + snd_dbri_free(dbri); + snd_card_free(card); + return err; + } + + printk(KERN_INFO "audio%d at %p (irq %d) is DBRI(%c)+CS4215(%d)\n", + dev, dbri->regs, + dbri->irq, dbri->dbri_version, dbri->mm.version); + dev++; + + return 0; +} + +/* Probe for the dbri chip and then attach the driver. */ +static int __init dbri_init(void) +{ + struct sbus_bus *sbus; + struct sbus_dev *sdev; + int found = 0; + + /* Probe each SBUS for the DBRI chip(s). */ + for_all_sbusdev(sdev, sbus) { + /* + * The version is coded in the last character + */ + if (!strncmp(sdev->prom_name, "SUNW,DBRI", 9)) { + dprintk(D_GEN, "DBRI: Found %s in SBUS slot %d\n", + sdev->prom_name, sdev->slot); + + if (dbri_attach(sdev->prom_node, sdev) == 0) + found++; + } + } + + return (found > 0) ? 0 : -EIO; +} + +static void __exit dbri_exit(void) +{ + snd_dbri_t *this = dbri_list; + + while (this != NULL) { + snd_dbri_t *next = this->next; + snd_card_t *card = this->card; + + snd_dbri_free(this); + snd_card_free(card); + this = next; + } + dbri_list = NULL; +} + +module_init(dbri_init); +module_exit(dbri_exit); -- cgit v1.2.3 From 2c484df0d249323d223f7f58e0f3b992b7414be8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2005 18:54:04 +0200 Subject: [ALSA] Add ARM PXA2xx AC97 driver Documentation,ARM,/arm/Makefile,ARM PXA2XX driver Added ARM PXA2xx AC97 driver by Nicolas Pitre (moved from alsa-driver tree). Signed-off-by: Takashi Iwai --- sound/arm/Kconfig | 14 +- sound/arm/Makefile | 8 +- sound/arm/pxa2xx-ac97.c | 410 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/arm/pxa2xx-pcm.c | 367 +++++++++++++++++++++++++++++++++++++++++++ sound/arm/pxa2xx-pcm.h | 29 ++++ 5 files changed, 824 insertions(+), 4 deletions(-) create mode 100644 sound/arm/pxa2xx-ac97.c create mode 100644 sound/arm/pxa2xx-pcm.c create mode 100644 sound/arm/pxa2xx-pcm.h (limited to 'sound') diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 34c1740aa6e9..2e4a5e0d16db 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -20,5 +20,17 @@ config SND_ARMAACI select SND_PCM select SND_AC97_CODEC -endmenu +config SND_PXA2XX_PCM + tristate + select SND_PCM + +config SND_PXA2XX_AC97 + tristate "AC97 driver for the Intel PXA2xx chip" + depends on ARCH_PXA && SND + select SND_PXA2XX_PCM + select SND_AC97_CODEC + help + Say Y or M if you want to support any AC97 codec attached to + the PXA2xx AC97 interface. +endmenu diff --git a/sound/arm/Makefile b/sound/arm/Makefile index f74ec28e1068..103f136926d9 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -3,9 +3,11 @@ # snd-sa11xx-uda1341-objs := sa11xx-uda1341.o +snd-aaci-objs := aaci.o devdma.o +snd-pxa2xx-pcm-objs := pxa2xx-pcm.o +snd-pxa2xx-ac97-objs := pxa2xx-ac97.o -# Toplevel Module Dependency obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o - obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o -snd-aaci-objs := aaci.o devdma.o +obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o +obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c new file mode 100644 index 000000000000..46052304e230 --- /dev/null +++ b/sound/arm/pxa2xx-ac97.c @@ -0,0 +1,410 @@ +/* + * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip. + * + * Author: Nicolas Pitre + * Created: Dec 02, 2004 + * Copyright: MontaVista Software Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "pxa2xx-pcm.h" + + +static DECLARE_MUTEX(car_mutex); +static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); +static volatile long gsr_bits; + +static unsigned short pxa2xx_ac97_read(ac97_t *ac97, unsigned short reg) +{ + unsigned short val = -1; + volatile u32 *reg_addr; + + down(&car_mutex); + if (CAR & CAR_CAIP) { + printk(KERN_CRIT"%s: CAR_CAIP already set\n", __FUNCTION__); + goto out; + } + + /* set up primary or secondary codec space */ + reg_addr = (ac97->num & 1) ? &SAC_REG_BASE : &PAC_REG_BASE; + reg_addr += (reg >> 1); + + /* start read access across the ac97 link */ + gsr_bits = 0; + val = *reg_addr; + if (reg == AC97_GPIO_STATUS) + goto out; + wait_event_timeout(gsr_wq, gsr_bits & GSR_SDONE, 1); + if (!gsr_bits & GSR_SDONE) { + printk(KERN_ERR "%s: read error (ac97_reg=%d GSR=%#lx)\n", + __FUNCTION__, reg, gsr_bits); + val = -1; + goto out; + } + + /* valid data now */ + gsr_bits = 0; + val = *reg_addr; + /* but we've just started another cycle... */ + wait_event_timeout(gsr_wq, gsr_bits & GSR_SDONE, 1); + +out: up(&car_mutex); + return val; +} + +static void pxa2xx_ac97_write(ac97_t *ac97, unsigned short reg, unsigned short val) +{ + volatile u32 *reg_addr; + + down(&car_mutex); + + if (CAR & CAR_CAIP) { + printk(KERN_CRIT "%s: CAR_CAIP already set\n", __FUNCTION__); + goto out; + } + + /* set up primary or secondary codec space */ + reg_addr = (ac97->num & 1) ? &SAC_REG_BASE : &PAC_REG_BASE; + reg_addr += (reg >> 1); + gsr_bits = 0; + *reg_addr = val; + wait_event_timeout(gsr_wq, gsr_bits & GSR_CDONE, 1); + if (!gsr_bits & GSR_SDONE) + printk(KERN_ERR "%s: write error (ac97_reg=%d GSR=%#lx)\n", + __FUNCTION__, reg, gsr_bits); + +out: up(&car_mutex); +} + +static void pxa2xx_ac97_reset(ac97_t *ac97) +{ + /* First, try cold reset */ + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ + GCR &= ~GCR_COLD_RST; /* then assert nCRST */ + + gsr_bits = 0; +#ifdef CONFIG_PXA27x + /* PXA27x Developers Manual section 13.5.2.2.1 */ + pxa_set_cken(1 << 31, 1); + udelay(5); + pxa_set_cken(1 << 31, 0); + GCR = GCR_COLD_RST; + udelay(50); +#else + GCR = GCR_COLD_RST; + GCR |= GCR_CDONE_IE|GCR_SDONE_IE; + wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); +#endif + + if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { + printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", + __FUNCTION__, gsr_bits); + + /* let's try warm reset */ + gsr_bits = 0; +#ifdef CONFIG_PXA27x + /* warm reset broken on Bulverde, + so manually keep AC97 reset high */ + pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + udelay(10); + GCR |= GCR_WARM_RST; + pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + udelay(50); +#else + GCR |= GCR_WARM_RST|GCR_PRIRDY_IEN|GCR_SECRDY_IEN;; + wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); +#endif + + if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) + printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", + __FUNCTION__, gsr_bits); + } + + GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); + GCR |= GCR_SDONE_IE|GCR_CDONE_IE; +} + +static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id, struct pt_regs *regs) +{ + long status; + + status = GSR; + if (status) { + GSR = status; + gsr_bits |= status; + wake_up(&gsr_wq); + +#ifdef CONFIG_PXA27x + /* Although we don't use those we still need to clear them + since they tend to spuriously trigger when MMC is used + (hardware bug? go figure)... */ + MISR = MISR_EOC; + PISR = PISR_EOC; + MCSR = MCSR_EOC; +#endif + + return IRQ_HANDLED; + } + + return IRQ_NONE; +} + +static ac97_bus_ops_t pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read, + .write = pxa2xx_ac97_write, + .reset = pxa2xx_ac97_reset, +}; + +static pxa2xx_pcm_dma_params_t pxa2xx_ac97_pcm_out = { + .name = "AC97 PCM out", + .dev_addr = __PREG(PCDR), + .drcmr = &DRCMRTXPCDR, + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST32 | DCMD_WIDTH4, +}; + +static pxa2xx_pcm_dma_params_t pxa2xx_ac97_pcm_in = { + .name = "AC97 PCM in", + .dev_addr = __PREG(PCDR), + .drcmr = &DRCMRRXPCDR, + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST32 | DCMD_WIDTH4, +}; + +static snd_pcm_t *pxa2xx_ac97_pcm; +static ac97_t *pxa2xx_ac97_ac97; + +static int pxa2xx_ac97_pcm_startup(snd_pcm_substream_t *substream) +{ + snd_pcm_runtime_t *runtime = substream->runtime; + pxa2xx_audio_ops_t *platform_ops; + int r; + + runtime->hw.channels_min = 2; + runtime->hw.channels_max = 2; + + r = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + AC97_RATES_FRONT_DAC : AC97_RATES_ADC; + runtime->hw.rates = pxa2xx_ac97_ac97->rates[r]; + snd_pcm_limit_hw_rates(runtime); + + platform_ops = substream->pcm->card->dev->platform_data; + if (platform_ops && platform_ops->startup) + return platform_ops->startup(substream, platform_ops->priv); + else + return 0; +} + +static void pxa2xx_ac97_pcm_shutdown(snd_pcm_substream_t *substream) +{ + pxa2xx_audio_ops_t *platform_ops; + + platform_ops = substream->pcm->card->dev->platform_data; + if (platform_ops && platform_ops->shutdown) + platform_ops->shutdown(substream, platform_ops->priv); +} + +static int pxa2xx_ac97_pcm_prepare(snd_pcm_substream_t *substream) +{ + snd_pcm_runtime_t *runtime = substream->runtime; + int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; + return snd_ac97_set_rate(pxa2xx_ac97_ac97, reg, runtime->rate); +} + +static pxa2xx_pcm_client_t pxa2xx_ac97_pcm_client = { + .playback_params = &pxa2xx_ac97_pcm_out, + .capture_params = &pxa2xx_ac97_pcm_in, + .startup = pxa2xx_ac97_pcm_startup, + .shutdown = pxa2xx_ac97_pcm_shutdown, + .prepare = pxa2xx_ac97_pcm_prepare, +}; + +#ifdef CONFIG_PM + +static int pxa2xx_ac97_do_suspend(snd_card_t *card, unsigned int state) +{ + if (card->power_state != SNDRV_CTL_POWER_D3cold) { + pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; + snd_pcm_suspend_all(pxa2xx_ac97_pcm); + snd_ac97_suspend(pxa2xx_ac97_ac97); + snd_power_change_state(card, SNDRV_CTL_POWER_D3cold); + if (platform_ops && platform_ops->suspend) + platform_ops->suspend(platform_ops->priv); + GCR |= GCR_ACLINK_OFF; + pxa_set_cken(CKEN2_AC97, 0); + } + + return 0; +} + +static int pxa2xx_ac97_do_resume(snd_card_t *card, unsigned int state) +{ + if (card->power_state != SNDRV_CTL_POWER_D0) { + pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; + pxa_set_cken(CKEN2_AC97, 1); + if (platform_ops && platform_ops->resume) + platform_ops->resume(platform_ops->priv); + snd_ac97_resume(pxa2xx_ac97_ac97); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + } + + return 0; +} + +static int pxa2xx_ac97_suspend(struct device *_dev, u32 state, u32 level) +{ + snd_card_t *card = dev_get_drvdata(_dev); + int ret = 0; + + if (card && level == SUSPEND_DISABLE) + ret = pxa2xx_ac97_do_suspend(card, SNDRV_CTL_POWER_D3cold); + + return ret; +} + +static int pxa2xx_ac97_resume(struct device *_dev, u32 level) +{ + snd_card_t *card = dev_get_drvdata(_dev); + int ret = 0; + + if (card && level == RESUME_ENABLE) + ret = pxa2xx_ac97_do_resume(card, SNDRV_CTL_POWER_D0); + + return ret; +} + +#else +#define pxa2xx_ac97_suspend NULL +#define pxa2xx_ac97_resume NULL +#endif + +static int pxa2xx_ac97_probe(struct device *dev) +{ + snd_card_t *card; + ac97_bus_t *ac97_bus; + ac97_template_t ac97_template; + int ret; + + ret = -ENOMEM; + card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, 0); + if (!card) + goto err; + + card->dev = dev; + strncpy(card->driver, dev->driver->name, sizeof(card->driver)); + + ret = pxa2xx_pcm_new(card, &pxa2xx_ac97_pcm_client, &pxa2xx_ac97_pcm); + if (ret) + goto err; + + ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL); + if (ret < 0) + goto err; + + pxa_gpio_mode(GPIO31_SYNC_AC97_MD); + pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); + pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); + pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); +#ifdef CONFIG_PXA27x + /* Use GPIO 113 as AC97 Reset on Bulverde */ + pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); +#endif + pxa_set_cken(CKEN2_AC97, 1); + + ret = snd_ac97_bus(card, 0, &pxa2xx_ac97_ops, NULL, &ac97_bus); + if (ret) + goto err; + memset(&ac97_template, 0, sizeof(ac97_template)); + ret = snd_ac97_mixer(ac97_bus, &ac97_template, &pxa2xx_ac97_ac97); + if (ret) + goto err; + + snprintf(card->shortname, sizeof(card->shortname), + "%s", snd_ac97_get_short_name(pxa2xx_ac97_ac97)); + snprintf(card->longname, sizeof(card->longname), + "%s (%s)", dev->driver->name, card->mixername); + + snd_card_set_pm_callback(card, pxa2xx_ac97_do_suspend, + pxa2xx_ac97_do_resume, NULL); + ret = snd_card_register(card); + if (ret == 0) { + dev_set_drvdata(dev, card); + return 0; + } + + err: + if (card) + snd_card_free(card); + if (CKEN & CKEN2_AC97) { + GCR |= GCR_ACLINK_OFF; + free_irq(IRQ_AC97, NULL); + pxa_set_cken(CKEN2_AC97, 0); + } + return ret; +} + +static int pxa2xx_ac97_remove(struct device *dev) +{ + snd_card_t *card = dev_get_drvdata(dev); + + if (card) { + snd_card_free(card); + dev_set_drvdata(dev, NULL); + GCR |= GCR_ACLINK_OFF; + free_irq(IRQ_AC97, NULL); + pxa_set_cken(CKEN2_AC97, 0); + } + + return 0; +} + +static struct device_driver pxa2xx_ac97_driver = { + .name = "pxa2xx-ac97", + .bus = &platform_bus_type, + .probe = pxa2xx_ac97_probe, + .remove = pxa2xx_ac97_remove, + .suspend = pxa2xx_ac97_suspend, + .resume = pxa2xx_ac97_resume, +}; + +static int __init pxa2xx_ac97_init(void) +{ + return driver_register(&pxa2xx_ac97_driver); +} + +static void __exit pxa2xx_ac97_exit(void) +{ + driver_unregister(&pxa2xx_ac97_driver); +} + +module_init(pxa2xx_ac97_init); +module_exit(pxa2xx_ac97_exit); + +MODULE_AUTHOR("Nicolas Pitre"); +MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); +MODULE_LICENSE("GPL"); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c new file mode 100644 index 000000000000..b1eb53b02eae --- /dev/null +++ b/sound/arm/pxa2xx-pcm.c @@ -0,0 +1,367 @@ +/* + * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include +#include +#include + +#include "pxa2xx-pcm.h" + + +static const snd_pcm_hardware_t pxa2xx_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192 - 32, + .periods_min = 1, + .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc), + .buffer_bytes_max = 128 * 1024, + .fifo_size = 32, +}; + +struct pxa2xx_runtime_data { + int dma_ch; + pxa2xx_pcm_dma_params_t *params; + pxa_dma_desc *dma_desc_array; + dma_addr_t dma_desc_array_phys; +}; + +static int pxa2xx_pcm_hw_params(snd_pcm_substream_t *substream, + snd_pcm_hw_params_t *params) +{ + snd_pcm_runtime_t *runtime = substream->runtime; + struct pxa2xx_runtime_data *rtd = runtime->private_data; + size_t totsize = params_buffer_bytes(params); + size_t period = params_period_bytes(params); + pxa_dma_desc *dma_desc; + dma_addr_t dma_buff_phys, next_desc_phys; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = totsize; + + dma_desc = rtd->dma_desc_array; + next_desc_phys = rtd->dma_desc_array_phys; + dma_buff_phys = runtime->dma_addr; + do { + next_desc_phys += sizeof(pxa_dma_desc); + dma_desc->ddadr = next_desc_phys; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_desc->dsadr = dma_buff_phys; + dma_desc->dtadr = rtd->params->dev_addr; + } else { + dma_desc->dsadr = rtd->params->dev_addr; + dma_desc->dtadr = dma_buff_phys; + } + if (period > totsize) + period = totsize; + dma_desc->dcmd = rtd->params->dcmd | period | DCMD_ENDIRQEN; + dma_desc++; + dma_buff_phys += period; + } while (totsize -= period); + dma_desc[-1].ddadr = rtd->dma_desc_array_phys; + + return 0; +} + +static int pxa2xx_pcm_hw_free(snd_pcm_substream_t *substream) +{ + struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; + + *rtd->params->drcmr = 0; + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int pxa2xx_pcm_prepare(snd_pcm_substream_t *substream) +{ + pxa2xx_pcm_client_t *client = substream->private_data; + snd_pcm_runtime_t *runtime = substream->runtime; + struct pxa2xx_runtime_data *rtd = runtime->private_data; + + DCSR(rtd->dma_ch) &= ~DCSR_RUN; + DCSR(rtd->dma_ch) = 0; + DCMD(rtd->dma_ch) = 0; + *rtd->params->drcmr = rtd->dma_ch | DRCMR_MAPVLD; + + return client->prepare(substream); +} + +static int pxa2xx_pcm_trigger(snd_pcm_substream_t *substream, int cmd) +{ + struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + DDADR(rtd->dma_ch) = rtd->dma_desc_array_phys; + DCSR(rtd->dma_ch) = DCSR_RUN; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + DCSR(rtd->dma_ch) &= ~DCSR_RUN; + break; + + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + DCSR(rtd->dma_ch) |= DCSR_RUN; + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id, struct pt_regs *regs) +{ + snd_pcm_substream_t *substream = dev_id; + struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; + int dcsr; + + dcsr = DCSR(dma_ch); + DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN; + + if (dcsr & DCSR_ENDINTR) { + snd_pcm_period_elapsed(substream); + } else { + printk( KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", + rtd->params->name, dma_ch, dcsr ); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + } +} + +static snd_pcm_uframes_t pxa2xx_pcm_pointer(snd_pcm_substream_t *substream) +{ + snd_pcm_runtime_t *runtime = substream->runtime; + struct pxa2xx_runtime_data *rtd = runtime->private_data; + dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + DSADR(rtd->dma_ch) : DTADR(rtd->dma_ch); + snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr); + if (x == runtime->buffer_size) + x = 0; + return x; +} + +static int +pxa2xx_pcm_hw_rule_mult32(snd_pcm_hw_params_t *params, snd_pcm_hw_rule_t *rule) +{ + snd_interval_t *i = hw_param_interval(params, rule->var); + int changed = 0; + + if (i->min & 31) { + i->min = (i->min & ~31) + 32; + i->openmin = 0; + changed = 1; + } + + if (i->max & 31) { + i->max &= ~31; + i->openmax = 0; + changed = 1; + } + + return changed; +} + +static int pxa2xx_pcm_open(snd_pcm_substream_t *substream) +{ + pxa2xx_pcm_client_t *client = substream->private_data; + snd_pcm_runtime_t *runtime = substream->runtime; + struct pxa2xx_runtime_data *rtd; + int ret; + + runtime->hw = pxa2xx_pcm_hardware; + + /* + * For mysterious reasons (and despite what the manual says) + * playback samples are lost if the DMA count is not a multiple + * of the DMA burst size. Let's add a rule to enforce that. + */ + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + pxa2xx_pcm_hw_rule_mult32, NULL, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, -1); + if (ret) + goto out; + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + pxa2xx_pcm_hw_rule_mult32, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1); + if (ret) + goto out; + + ret = -ENOMEM; + rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + if (!rtd) + goto out; + rtd->dma_desc_array = + dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE, + &rtd->dma_desc_array_phys, GFP_KERNEL); + if (!rtd->dma_desc_array) + goto err1; + + rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + client->playback_params : client->capture_params; + ret = pxa_request_dma(rtd->params->name, DMA_PRIO_LOW, + pxa2xx_pcm_dma_irq, substream); + if (ret < 0) + goto err2; + rtd->dma_ch = ret; + + runtime->private_data = rtd; + ret = client->startup(substream); + if (!ret) + goto out; + + pxa_free_dma(rtd->dma_ch); + err2: + dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, + rtd->dma_desc_array, rtd->dma_desc_array_phys); + err1: + kfree(rtd); + out: + return ret; +} + +static int pxa2xx_pcm_close(snd_pcm_substream_t *substream) +{ + pxa2xx_pcm_client_t *client = substream->private_data; + struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; + + pxa_free_dma(rtd->dma_ch); + client->shutdown(substream); + dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, + rtd->dma_desc_array, rtd->dma_desc_array_phys); + kfree(rtd); + return 0; +} + +static int +pxa2xx_pcm_mmap(snd_pcm_substream_t *substream, struct vm_area_struct *vma) +{ + snd_pcm_runtime_t *runtime = substream->runtime; + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static snd_pcm_ops_t pxa2xx_pcm_ops = { + .open = pxa2xx_pcm_open, + .close = pxa2xx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pxa2xx_pcm_hw_params, + .hw_free = pxa2xx_pcm_hw_free, + .prepare = pxa2xx_pcm_prepare, + .trigger = pxa2xx_pcm_trigger, + .pointer = pxa2xx_pcm_pointer, + .mmap = pxa2xx_pcm_mmap, +}; + +static int pxa2xx_pcm_preallocate_dma_buffer(snd_pcm_t *pcm, int stream) +{ + snd_pcm_substream_t *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = pxa2xx_pcm_hardware.buffer_bytes_max; + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static void pxa2xx_pcm_free_dma_buffers(snd_pcm_t *pcm) +{ + snd_pcm_substream_t *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 pxa2xx_pcm_dmamask = 0xffffffff; + +int pxa2xx_pcm_new(snd_card_t *card, pxa2xx_pcm_client_t *client, snd_pcm_t **rpcm) +{ + snd_pcm_t *pcm; + int play = client->playback_params ? 1 : 0; + int capt = client->capture_params ? 1 : 0; + int ret; + + ret = snd_pcm_new(card, "PXA2xx-PCM", 0, play, capt, &pcm); + if (ret) + goto out; + + pcm->private_data = client; + pcm->private_free = pxa2xx_pcm_free_dma_buffers; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &pxa2xx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (play) { + int stream = SNDRV_PCM_STREAM_PLAYBACK; + snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + } + if (capt) { + int stream = SNDRV_PCM_STREAM_CAPTURE; + snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops); + ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream); + if (ret) + goto out; + } + + if (rpcm) + *rpcm = pcm; + ret = 0; + + out: + return ret; +} + +EXPORT_SYMBOL(pxa2xx_pcm_new); + +MODULE_AUTHOR("Nicolas Pitre"); +MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h new file mode 100644 index 000000000000..43517597cab9 --- /dev/null +++ b/sound/arm/pxa2xx-pcm.h @@ -0,0 +1,29 @@ +/* + * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +typedef struct { + char *name; /* stream identifier */ + u32 dcmd; /* DMA descriptor dcmd field */ + volatile u32 *drcmr; /* the DMA request channel to use */ + u32 dev_addr; /* device physical address for DMA */ +} pxa2xx_pcm_dma_params_t; + +typedef struct { + pxa2xx_pcm_dma_params_t *playback_params; + pxa2xx_pcm_dma_params_t *capture_params; + int (*startup)(snd_pcm_substream_t *); + void (*shutdown)(snd_pcm_substream_t *); + int (*prepare)(snd_pcm_substream_t *); +} pxa2xx_pcm_client_t; + +extern int pxa2xx_pcm_new(snd_card_t *, pxa2xx_pcm_client_t *, snd_pcm_t **); + -- cgit v1.2.3 From 0ac2ac0ad7d1e8c84241d1c40b0d196b9b7c8104 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 1 Jul 2005 16:19:39 +0200 Subject: [ALSA] usb-audio: add support for an unknown Yamaha USB MIDI device USB generic driver Add a quirk for the Yamaha USB MIDI device with USB ID 0x103d. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f5135641b3e2..cc2e3c9933ee 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -116,6 +116,7 @@ YAMAHA_DEVICE(0x1039, NULL), YAMAHA_DEVICE(0x103a, NULL), YAMAHA_DEVICE(0x103b, NULL), YAMAHA_DEVICE(0x103c, NULL), +YAMAHA_DEVICE(0x103d, NULL), YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- cgit v1.2.3 From 7bc71ecd6477db90221efc08fb742b3df4f49b46 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sat, 2 Jul 2005 15:20:57 +0200 Subject: [ALSA] via82xx - added MSI K7T266 Pro2 - 4005:4710 to white list (DXS enable) VIA82xx driver Reporter: Marko Kohtala Signed-off-by: Jaroslav Kysela --- sound/pci/via82xx.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 42c48f0ce8e8..1a286e1a60d4 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2187,6 +2187,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x1695, .subdevice = 0x3005, .action = VIA_DXS_ENABLE }, /* EPoX EP-8K9A */ { .subvendor = 0x1849, .subdevice = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */ { .subvendor = 0x1919, .subdevice = 0x200a, .action = VIA_DXS_NO_VRA }, /* Soltek SL-K8Tpro-939 */ + { .subvendor = 0x4005, .subdevice = 0x4710, .action = VIA_DXS_ENABLE }, /* MSI K7T266 Pro2 (MS-6380 V2.0) BIOS 3.7 */ { } /* terminator */ }; struct dxs_whitelist *w; -- cgit v1.2.3 From e0474e53985c5fac97a5bb85d66ec0d017b5faf3 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 2 Jul 2005 16:33:34 +0200 Subject: [ALSA] snd-emu10k1: Card capabilities tidy up. EMU10K1/EMU10K2 driver Signed-off-by: James Courtier-Dutton --- sound/pci/emu10k1/emu10k1.c | 4 ++-- sound/pci/emu10k1/emu10k1_main.c | 8 +++++--- 2 files changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 2085a998eaeb..2f96b2fc74fb 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -140,7 +140,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, return err; } /* This stores the periods table. */ - if (emu->audigy && emu->revision == 4) { /* P16V */ + if (emu->card_capabilities->ca0151_chip) { /* P16V */ if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &emu->p16v_buffer) < 0) { snd_p16v_free(emu); return -ENOMEM; @@ -161,7 +161,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, snd_card_free(card); return err; } - if (emu->audigy && emu->revision == 4) { /* P16V */ + if (emu->card_capabilities->ca0151_chip) { /* P16V */ if ((err = snd_p16v_pcm(emu, 4, NULL)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index a341e758acde..9478a95e410f 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -191,7 +191,7 @@ static int __devinit snd_emu10k1_init(emu10k1_t * emu, int enable_ir) /* Set playback routing. */ snd_emu10k1_ptr20_write(emu, CAPTURE_P16V_SOURCE, 0, 0x78e4); } - if (emu->audigy && (emu->serial == 0x10011102) ) { /* audigy2 Value */ + if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */ /* Hacks for Alice3 to work independent of haP16V driver */ u32 tmp; @@ -253,6 +253,8 @@ static int __devinit snd_emu10k1_init(emu10k1_t * emu, int enable_ir) HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG); else outl(HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG); + /* FIXME: Remove all these emu->model and replace it with a card recognition parameter, + * e.g. card_capabilities->joystick */ } else if (emu->model == 0x20 || emu->model == 0xc400 || (emu->model == 0x21 && emu->revision < 6)) @@ -299,12 +301,12 @@ static int __devinit snd_emu10k1_init(emu10k1_t * emu, int enable_ir) if (emu->audigy) { outl(inl(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG); - if (emu->revision == 4) { /* audigy2 */ + if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ /* Unmute Analog now. Set GPO6 to 1 for Apollo. * This has to be done after init ALice3 I2SOut beyond 48KHz. * So, sequence is important. */ outl(inl(emu->port + A_IOCFG) | 0x0040, emu->port + A_IOCFG); - } else if (emu->serial == 0x10011102) { /* audigy2 value */ + } else if (emu->card_capabilities->ca0108_chip) { /* audigy2 value */ /* Unmute Analog now. */ outl(inl(emu->port + A_IOCFG) | 0x0060, emu->port + A_IOCFG); } else { -- cgit v1.2.3 From 3818152e64866b54020b5656ff5fdd0f5e085183 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sat, 2 Jul 2005 18:03:37 +0200 Subject: [ALSA] snd-emu10k1: Tidy mixer controls. EMU10K1/EMU10K2 driver Signed-off-by: James Courtier-Dutton --- sound/pci/emu10k1/p16v.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 98f980189892..a1691330d3b6 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -822,7 +822,7 @@ static int snd_p16v_volume_put_analog_unknown(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_p16v_volume_control_analog_front = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD Analog Front Volume", + .name = "HD Analog Front Playback Volume", .info = snd_p16v_volume_info, .get = snd_p16v_volume_get_analog_front, .put = snd_p16v_volume_put_analog_front @@ -831,7 +831,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_analog_front = static snd_kcontrol_new_t snd_p16v_volume_control_analog_center_lfe = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD Analog Center/LFE Volume", + .name = "HD Analog Center/LFE Playback Volume", .info = snd_p16v_volume_info, .get = snd_p16v_volume_get_analog_center_lfe, .put = snd_p16v_volume_put_analog_center_lfe @@ -840,7 +840,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_analog_center_lfe = static snd_kcontrol_new_t snd_p16v_volume_control_analog_unknown = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD Analog Unknown Volume", + .name = "HD Analog Unknown Playback Volume", .info = snd_p16v_volume_info, .get = snd_p16v_volume_get_analog_unknown, .put = snd_p16v_volume_put_analog_unknown @@ -849,7 +849,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_analog_unknown = static snd_kcontrol_new_t snd_p16v_volume_control_analog_rear = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD Analog Rear Volume", + .name = "HD Analog Rear Playback Volume", .info = snd_p16v_volume_info, .get = snd_p16v_volume_get_analog_rear, .put = snd_p16v_volume_put_analog_rear @@ -858,7 +858,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_analog_rear = static snd_kcontrol_new_t snd_p16v_volume_control_spdif_front = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD SPDIF Front Volume", + .name = "HD SPDIF Front Playback Volume", .info = snd_p16v_volume_info, .get = snd_p16v_volume_get_spdif_front, .put = snd_p16v_volume_put_spdif_front @@ -867,7 +867,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_spdif_front = static snd_kcontrol_new_t snd_p16v_volume_control_spdif_center_lfe = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD SPDIF Center/LFE Volume", + .name = "HD SPDIF Center/LFE Playback Volume", .info = snd_p16v_volume_info, .get = snd_p16v_volume_get_spdif_center_lfe, .put = snd_p16v_volume_put_spdif_center_lfe @@ -876,7 +876,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_spdif_center_lfe = static snd_kcontrol_new_t snd_p16v_volume_control_spdif_unknown = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD SPDIF Unknown Volume", + .name = "HD SPDIF Unknown Playback Volume", .info = snd_p16v_volume_info, .get = snd_p16v_volume_get_spdif_unknown, .put = snd_p16v_volume_put_spdif_unknown @@ -885,7 +885,7 @@ static snd_kcontrol_new_t snd_p16v_volume_control_spdif_unknown = static snd_kcontrol_new_t snd_p16v_volume_control_spdif_rear = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD SPDIF Rear Volume", + .name = "HD SPDIF Rear Playback Volume", .info = snd_p16v_volume_info, .get = snd_p16v_volume_get_spdif_rear, .put = snd_p16v_volume_put_spdif_rear @@ -936,7 +936,7 @@ static int snd_p16v_capture_source_put(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_p16v_capture_source __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD Capture source", + .name = "HD source Capture", .info = snd_p16v_capture_source_info, .get = snd_p16v_capture_source_get, .put = snd_p16v_capture_source_put @@ -985,7 +985,7 @@ static int snd_p16v_capture_channel_put(snd_kcontrol_t * kcontrol, static snd_kcontrol_new_t snd_p16v_capture_channel __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "HD Capture channel", + .name = "HD channel Capture", .info = snd_p16v_capture_channel_info, .get = snd_p16v_capture_channel_get, .put = snd_p16v_capture_channel_put -- cgit v1.2.3 From a6f6192bb38a76c4ad44c894144b1fbf3d14606b Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sun, 3 Jul 2005 12:32:40 +0200 Subject: [ALSA] emu10k1: Sort by card id. EMU10K1/EMU10K2 driver Signed-off-by: James Courtier-Dutton --- sound/pci/emu10k1/emu10k1_main.c | 80 +++++++++++++++++++--------------------- 1 file changed, 37 insertions(+), 43 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 9478a95e410f..6dca38120b90 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -714,54 +714,49 @@ static emu_chip_details_t emu_chip_details[] = { .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102, - .driver = "EMU10K1", .name = "E-mu APS [4001]", - .id = "APS", - .emu10k1_chip = 1, - .ecard = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102, - .driver = "EMU10K1", .name = "SBLive! Player 5.1 [SB0060]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102, + .driver = "EMU10K1", .name = "SBLive! [SB0105]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102, - .driver = "EMU10K1", .name = "SB Live 5.1", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102, + .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102, - .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", - .id = "Live", - .emu10k1_chip = 1, - .ac97_chip = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102, - .driver = "EMU10K1", .name = "SBLive! [CT4620]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102, + .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102, + .driver = "EMU10K1", .name = "SB Live 5.1", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102, + .driver = "EMU10K1", .name = "SBLive! Player 5.1 [SB0060]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102, - .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102, + .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, @@ -772,50 +767,49 @@ static emu_chip_details_t emu_chip_details[] = { .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102, + .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102, - .driver = "EMU10K1", .name = "SBLive! Value [SB0060]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102, - .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", - .id = "Live", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102, + .driver = "EMU10K1", .name = "E-mu APS [4001]", + .id = "APS", .emu10k1_chip = 1, - .ac97_chip = 1, - .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102, - .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", + .ecard = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102, + .driver = "EMU10K1", .name = "SBLive! [CT4620]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102, - .driver = "EMU10K1", .name = "SBLive! [SB0105]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102, + .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, -- cgit v1.2.3 From 88dc0e5dadf9b0cb529c89b12cd10f75d5b1bce4 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sun, 3 Jul 2005 12:54:29 +0200 Subject: [ALSA] emu10k1: Added tested status comments. EMU10K1/EMU10K2 driver Signed-off-by: James Courtier-Dutton --- sound/pci/emu10k1/emu10k1_main.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 6dca38120b90..8bc9bc18c74b 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -616,6 +616,7 @@ static int snd_emu10k1_dev_free(snd_device_t *device) static emu_chip_details_t emu_chip_details[] = { /* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/ + /* Tested by James@superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102, .driver = "Audigy2", .name = "Audigy 2 Value [SB0400]", .id = "Audigy2", @@ -629,6 +630,7 @@ static emu_chip_details_t emu_chip_details[] = { .emu10k2_chip = 1, .ca0108_chip = 1, .ac97_chip = 1} , + /* Tested by James@superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102, .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", .id = "Audigy2", @@ -773,6 +775,7 @@ static emu_chip_details_t emu_chip_details[] = { .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , + /* Tested by James@superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102, .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", .id = "Live", -- cgit v1.2.3 From b1c6ef52e2623c81c2124801c783a903f6e5437a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Jul 2005 09:18:46 +0200 Subject: [ALSA] usb-audio - enable high speed transfers with Audiy 2 NX USB generic driver This patch enables the boot commands to activate high speed mode (and associated sample formats like 8 channels with 24 bits at 96 kHz) on the SB Audigy 2 NX. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c57b44511b54..aee3c0f28eb6 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2938,8 +2938,6 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) { -#if 0 - /* TODO: enable this when high speed synchronization actually works */ u8 buf = 1; snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a, @@ -2951,7 +2949,6 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) 1, 2000, NULL, 0, 1000); return -ENODEV; } -#endif return 0; } -- cgit v1.2.3 From 6155aff84b98b2aa35eaa4384b539dfbab86afcc Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Jul 2005 09:20:42 +0200 Subject: [ALSA] usb-audio - rename QUIRK_MIDI_MOTU to QUIRK_MIDI_RAW USB generic driver Rename the protocol used by the MOTU FastLane to 'raw' because it might be useful with other devices, and there are other MOTU interfaces that do not use this protocol. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 2 +- sound/usb/usbaudio.h | 4 ++-- sound/usb/usbmidi.c | 18 +++++++++--------- sound/usb/usbquirks.h | 2 +- 4 files changed, 13 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index aee3c0f28eb6..3eaa08e3e6a6 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2970,7 +2970,7 @@ static int snd_usb_create_quirk(snd_usb_audio_t *chip, case QUIRK_MIDI_YAMAHA: case QUIRK_MIDI_MIDIMAN: case QUIRK_MIDI_NOVATION: - case QUIRK_MIDI_MOTU: + case QUIRK_MIDI_RAW: case QUIRK_MIDI_EMAGIC: return snd_usb_create_midi_interface(chip, iface, quirk); case QUIRK_COMPOSITE: diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index aedb42aaa749..a87912205477 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -165,7 +165,7 @@ struct snd_usb_audio { #define QUIRK_AUDIO_EDIROL_UA1000 8 #define QUIRK_IGNORE_INTERFACE 9 #define QUIRK_MIDI_NOVATION 10 -#define QUIRK_MIDI_MOTU 11 +#define QUIRK_MIDI_RAW 11 #define QUIRK_MIDI_EMAGIC 12 typedef struct snd_usb_audio_quirk snd_usb_audio_quirk_t; @@ -205,7 +205,7 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_IGNORE_INTERFACE, data is NULL */ -/* for QUIRK_MIDI_NOVATION and _MOTU, data is NULL */ +/* for QUIRK_MIDI_NOVATION and _RAW, data is NULL */ /* for QUIRK_MIDI_EMAGIC, data points to a snd_usb_midi_endpoint_info * structure (out_cables and in_cables only) */ diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index bee70068dce0..5c754879c2b6 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -524,16 +524,16 @@ static struct usb_protocol_ops snd_usbmidi_novation_ops = { }; /* - * Mark of the Unicorn USB MIDI protocol: raw MIDI. + * "raw" protocol: used by the MOTU FastLane. */ -static void snd_usbmidi_motu_input(snd_usb_midi_in_endpoint_t* ep, - uint8_t* buffer, int buffer_length) +static void snd_usbmidi_raw_input(snd_usb_midi_in_endpoint_t* ep, + uint8_t* buffer, int buffer_length) { snd_usbmidi_input_data(ep, 0, buffer, buffer_length); } -static void snd_usbmidi_motu_output(snd_usb_midi_out_endpoint_t* ep) +static void snd_usbmidi_raw_output(snd_usb_midi_out_endpoint_t* ep) { int count; @@ -549,9 +549,9 @@ static void snd_usbmidi_motu_output(snd_usb_midi_out_endpoint_t* ep) ep->urb->transfer_buffer_length = count; } -static struct usb_protocol_ops snd_usbmidi_motu_ops = { - .input = snd_usbmidi_motu_input, - .output = snd_usbmidi_motu_output, +static struct usb_protocol_ops snd_usbmidi_raw_ops = { + .input = snd_usbmidi_raw_input, + .output = snd_usbmidi_raw_output, }; /* @@ -1505,8 +1505,8 @@ int snd_usb_create_midi_interface(snd_usb_audio_t* chip, umidi->usb_protocol_ops = &snd_usbmidi_novation_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; - case QUIRK_MIDI_MOTU: - umidi->usb_protocol_ops = &snd_usbmidi_motu_ops; + case QUIRK_MIDI_RAW: + umidi->usb_protocol_ops = &snd_usbmidi_raw_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; case QUIRK_MIDI_EMAGIC: diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index cc2e3c9933ee..ed1eba1ef5c6 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1269,7 +1269,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = & (const snd_usb_audio_quirk_t[]) { { .ifnum = 0, - .type = QUIRK_MIDI_MOTU + .type = QUIRK_MIDI_RAW }, { .ifnum = 1, -- cgit v1.2.3 From bbd4615cdb68de943b2814e956ec14899606dc45 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Jul 2005 09:21:45 +0200 Subject: [ALSA] usb-audio - use bDeviceSubClass to detect MOTU FastLane USB generic driver MOTU builds other USB MIDI interfaces with the same product ID as the FastLane, so we have to check the bDeviceSubClass field to differentiate between them. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index ed1eba1ef5c6..67796430e588 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1260,7 +1260,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Mark of the Unicorn devices */ { /* thanks to Robert A. Lerche */ - USB_DEVICE(0x07fd, 0x0001), + .match_flags = USB_DEVICE_ID_MATCH_VENDOR | + USB_DEVICE_ID_MATCH_PRODUCT | + USB_DEVICE_ID_MATCH_DEV_SUBCLASS, + .idVendor = 0x07fd, + .idProduct = 0x0001, + .bDeviceSubClass = 2, .driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) { .vendor_name = "MOTU", .product_name = "Fastlane", -- cgit v1.2.3 From 41e2fce431070cb2d91391808077378582d3e6b1 Mon Sep 17 00:00:00 2001 From: Matt Date: Mon, 4 Jul 2005 17:49:55 +0200 Subject: [ALSA] hda: enable unsolicited responses HDA Intel driver Patch enables unsolicited responses on the HDA controller. Without the UREN bit set, the controller will not place unsolicited responses in a RIRB. Signed-off-by: Matt Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5e0cca36ed57..288ab0764830 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -178,6 +178,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ #define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ +/* GCTL unsolicited response enable bit */ +#define ICH6_GCTL_UREN (1<<8) + /* GCTL reset bit */ #define ICH6_GCTL_RESET (1<<0) @@ -562,6 +565,9 @@ static int azx_reset(azx_t *chip) return -EBUSY; } + /* Accept unsolicited responses */ + azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UREN); + /* detect codecs */ if (! chip->codec_mask) { chip->codec_mask = azx_readw(chip, STATESTS); -- cgit v1.2.3 From 4e55096e27d745908e44c6abd2cc0c5b615854a4 Mon Sep 17 00:00:00 2001 From: Matt Date: Mon, 4 Jul 2005 17:51:39 +0200 Subject: [ALSA] hda: add sigmatel hp detect support HDA Codec driver Adds support for detecting hp insertion/removal and enable/disable of lineouts based on unsolicited events. Signed-off-by: Matt Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 3 ++ sound/pci/hda/patch_sigmatel.c | 62 ++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 63 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 59991560d492..dd0d99d2ad27 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -262,6 +262,9 @@ enum { #define AC_PINCTL_OUT_EN (1<<6) #define AC_PINCTL_HP_EN (1<<7) +/* Unsolicited response - 8bit */ +#define AC_USRSP_EN (1<<7) + /* configuration default - 32bit */ #define AC_DEFCFG_SEQUENCE (0xf<<0) #define AC_DEFCFG_DEF_ASSOC (0xf<<4) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 07d06f7c4390..9d503da7320d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -36,6 +36,10 @@ #undef STAC_TEST +#define NUM_CONTROL_ALLOC 32 +#define STAC_HP_EVENT 0x37 +#define STAC_UNSOL_ENABLE (AC_USRSP_EN | STAC_HP_EVENT) + struct sigmatel_spec { snd_kcontrol_new_t *mixers[4]; unsigned int num_mixers; @@ -507,8 +511,6 @@ static int stac92xx_build_pcms(struct hda_codec *codec) return 0; } -#define NUM_CONTROL_ALLOC 32 - enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, @@ -617,10 +619,18 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin hda_nid_t pin = cfg->hp_pin; hda_nid_t nid; int i, err; + unsigned int wid_caps; if (! pin) return 0; + wid_caps = snd_hda_param_read(codec, pin, AC_PAR_AUDIO_WIDGET_CAP); + if (wid_caps & AC_WCAP_UNSOL_CAP) + /* Enable unsolicited responses on the HP widget */ + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + STAC_UNSOL_ENABLE); + nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; for (i = 0; i < cfg->line_outs; i++) { if (! spec->multiout.dac_nids[i]) @@ -828,6 +838,53 @@ static void stac92xx_free(struct hda_codec *codec) kfree(spec); } +static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, + unsigned int flag) +{ + unsigned int pin_ctl = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl | flag); +} + +static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, + unsigned int flag) +{ + unsigned int pin_ctl = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl & ~flag); +} + +static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, presence; + + if ((res >> 26) != STAC_HP_EVENT) + return; + + presence = snd_hda_codec_read(codec, cfg->hp_pin, 0, + AC_VERB_GET_PIN_SENSE, 0x00) >> 31; + + if (presence) { + /* disable lineouts, enable hp */ + for (i = 0; i < cfg->line_outs; i++) + stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], + AC_PINCTL_OUT_EN); + stac92xx_set_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN); + } else { + /* enable lineouts, disable hp */ + for (i = 0; i < cfg->line_outs; i++) + stac92xx_set_pinctl(codec, cfg->line_out_pins[i], + AC_PINCTL_OUT_EN); + stac92xx_reset_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN); + } +} + #ifdef CONFIG_PM static int stac92xx_resume(struct hda_codec *codec) { @@ -851,6 +908,7 @@ static struct hda_codec_ops stac92xx_patch_ops = { .build_pcms = stac92xx_build_pcms, .init = stac92xx_init, .free = stac92xx_free, + .unsol_event = stac92xx_unsol_event, #ifdef CONFIG_PM .resume = stac92xx_resume, #endif -- cgit v1.2.3 From e3ea4d896109edd64dc549ecaeeff8d89025fb57 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Jul 2005 18:12:39 +0200 Subject: [ALSA] hdsp - Add 'Sample Clock Source Locking' control RME HDSP driver Added 'Sample Clock Source Locking' control. If this switch is on, the clock source can't be changed via PCM hw_params API (as sample rate). This will fix the problem of OSS-emulation, for example. Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 62 +++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 53 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index a673cc438b91..0db558a92871 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -445,6 +445,7 @@ struct _hdsp { u32 control2_register; /* cached value */ u32 creg_spdif; u32 creg_spdif_stream; + int clock_source_locked; char *card_name; /* digiface/multiface */ HDSP_IO_Type io_type; /* ditto, but for code use */ unsigned short firmware_rev; @@ -2095,6 +2096,34 @@ static int snd_hdsp_put_clock_source(snd_kcontrol_t * kcontrol, snd_ctl_elem_val return change; } +static int snd_hdsp_info_clock_source_lock(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_hdsp_get_clock_source_lock(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol) +{ + hdsp_t *hdsp = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = hdsp->clock_source_locked; + return 0; +} + +static int snd_hdsp_put_clock_source_lock(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol) +{ + hdsp_t *hdsp = snd_kcontrol_chip(kcontrol); + int change; + + change = (int)ucontrol->value.integer.value[0] != hdsp->clock_source_locked; + if (change) + hdsp->clock_source_locked = ucontrol->value.integer.value[0]; + return change; +} + #define HDSP_DA_GAIN(xname, xindex) \ { .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ .name = xname, \ @@ -3117,6 +3146,15 @@ HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0), HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0), /* 'Sample Clock Source' complies with the alsa control naming scheme */ HDSP_CLOCK_SOURCE("Sample Clock Source", 0), +{ + /* FIXME: should be PCM or MIXER? */ + /* .iface = SNDRV_CTL_ELEM_IFACE_PCM, */ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sample Clock Source Locking", + .info = snd_hdsp_info_clock_source_lock, + .get = snd_hdsp_get_clock_source_lock, + .put = snd_hdsp_put_clock_source_lock, +}, HDSP_SYSTEM_CLOCK_MODE("System Clock Mode", 0), HDSP_PREF_SYNC_REF("Preferred Sync Reference", 0), HDSP_AUTOSYNC_REF("AutoSync Reference", 0), @@ -3349,6 +3387,7 @@ snd_hdsp_proc_read(snd_info_entry_t *entry, snd_info_buffer_t *buffer) snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode); snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate); + snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No"); snd_iprintf(buffer, "\n"); @@ -3853,13 +3892,14 @@ static int snd_hdsp_hw_params(snd_pcm_substream_t *substream, */ spin_lock_irq(&hdsp->lock); - if ((err = hdsp_set_rate(hdsp, params_rate(params), 0)) < 0) { - spin_unlock_irq(&hdsp->lock); - _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE); - return err; - } else { - spin_unlock_irq(&hdsp->lock); + if (! hdsp->clock_source_locked) { + if ((err = hdsp_set_rate(hdsp, params_rate(params), 0)) < 0) { + spin_unlock_irq(&hdsp->lock); + _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE); + return err; + } } + spin_unlock_irq(&hdsp->lock); if ((err = hdsp_set_interrupt_interval(hdsp, params_period_size(params))) < 0) { _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); @@ -4284,13 +4324,17 @@ static int snd_hdsp_playback_open(snd_pcm_substream_t *substream) snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hdsp_hw_constraints_period_sizes); - if (hdsp->io_type == H9632) { - runtime->hw.channels_min = hdsp->qs_out_channels; - runtime->hw.channels_max = hdsp->ss_out_channels; + if (hdsp->clock_source_locked) { + runtime->hw.rate_min = runtime->hw.rate_max = hdsp->system_sample_rate; + } else if (hdsp->io_type == H9632) { runtime->hw.rate_max = 192000; runtime->hw.rates = SNDRV_PCM_RATE_KNOT; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates); } + if (hdsp->io_type == H9632) { + runtime->hw.channels_min = hdsp->qs_out_channels; + runtime->hw.channels_max = hdsp->ss_out_channels; + } snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_out_channels, hdsp, -- cgit v1.2.3 From 2201987c562f7c810440d399ef7a85fe79be01e7 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 5 Jul 2005 10:27:09 +0200 Subject: [ALSA] via82xx - changed MSI K7T266 Pro2 - 4005:4710 in white list (SRC enable) VIA82xx driver Signed-off-by: Jaroslav Kysela --- sound/pci/via82xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 1a286e1a60d4..064972b14d00 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2187,7 +2187,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x1695, .subdevice = 0x3005, .action = VIA_DXS_ENABLE }, /* EPoX EP-8K9A */ { .subvendor = 0x1849, .subdevice = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */ { .subvendor = 0x1919, .subdevice = 0x200a, .action = VIA_DXS_NO_VRA }, /* Soltek SL-K8Tpro-939 */ - { .subvendor = 0x4005, .subdevice = 0x4710, .action = VIA_DXS_ENABLE }, /* MSI K7T266 Pro2 (MS-6380 V2.0) BIOS 3.7 */ + { .subvendor = 0x4005, .subdevice = 0x4710, .action = VIA_DXS_SRC }, /* MSI K7T266 Pro2 (MS-6380 V2.0) BIOS 3.7 */ { } /* terminator */ }; struct dxs_whitelist *w; -- cgit v1.2.3 From e66bc8b2a7d85166935a2da651b94efb9e7a2f11 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Wed, 6 Jul 2005 22:21:51 +0200 Subject: [ALSA] emu10k1: Add module option uint subsystem. EMU10K1/EMU10K2 driver It allows the user to force the snd-emu10k1 module to think the user has a particular sound card. Useful if their particular sound card is not yet recognised. Signed-off-by: James Courtier-Dutton --- sound/pci/emu10k1/emu10k1.c | 6 ++++-- sound/pci/emu10k1/emu10k1_main.c | 25 ++++++++++++++++++------- 2 files changed, 22 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 2f96b2fc74fb..b17142cabead 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -52,6 +52,7 @@ static int seq_ports[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4}; static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64}; static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128}; static int enable_ir[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static uint subsystem[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; /* Force card subsystem model */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard."); @@ -71,7 +72,8 @@ module_param_array(max_buffer_size, int, NULL, 0444); MODULE_PARM_DESC(max_buffer_size, "Maximum sample buffer size in MB."); module_param_array(enable_ir, bool, NULL, 0444); MODULE_PARM_DESC(enable_ir, "Enable IR."); - +module_param_array(subsystem, uint, NULL, 0444); +MODULE_PARM_DESC(subsystem, "Force card subsystem model."); /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ @@ -122,7 +124,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, max_buffer_size[dev] = 1024; if ((err = snd_emu10k1_create(card, pci, extin[dev], extout[dev], (long)max_buffer_size[dev] * 1024 * 1024, - enable_ir[dev], + enable_ir[dev], subsystem[dev], &emu)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 8bc9bc18c74b..4ced4b092539 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -832,6 +832,7 @@ int __devinit snd_emu10k1_create(snd_card_t * card, unsigned short extout_mask, long max_cache_bytes, int enable_ir, + uint subsystem, emu10k1_t ** remu) { emu10k1_t *emu; @@ -877,10 +878,16 @@ int __devinit snd_emu10k1_create(snd_card_t * card, for (c = emu_chip_details; c->vendor; c++) { if (c->vendor == pci->vendor && c->device == pci->device) { - if (c->subsystem && c->subsystem != emu->serial) - continue; - if (c->revision && c->revision != emu->revision) - continue; + if (subsystem) { + if (c->subsystem && (c->subsystem == subsystem) ) { + break; + } else continue; + } else { + if (c->subsystem && (c->subsystem != emu->serial) ) + continue; + if (c->revision && c->revision != emu->revision) + continue; + } break; } } @@ -891,10 +898,14 @@ int __devinit snd_emu10k1_create(snd_card_t * card, return -ENOENT; } emu->card_capabilities = c; - if (c->subsystem != 0) + if (c->subsystem && !subsystem) snd_printdd("Sound card name=%s\n", c->name); - else - snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x\n", c->name, pci->vendor, pci->device, emu->serial); + else if (subsystem) + snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x. Forced to subsytem=0x%x\n", + c->name, pci->vendor, pci->device, emu->serial, c->subsystem); + else + snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x.\n", + c->name, pci->vendor, pci->device, emu->serial); if (!*card->id && c->id) { int i, n = 0; -- cgit v1.2.3 From ae3a72d8cb4e5b30606c5e3ac9c59b729117579a Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Wed, 6 Jul 2005 22:36:18 +0200 Subject: [ALSA] snd-emu10k1: Fixes recognition of Audigy ES. EMU10K1/EMU10K2 driver Fixes ALSA bug #1237. Signed-off-by: James Courtier-Dutton --- sound/pci/emu10k1/emu10k1_main.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 4ced4b092539..7c31d9b30240 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -691,18 +691,18 @@ static emu_chip_details_t emu_chip_details[] = { .ca0151_chip = 1, .spdif_bug = 1, .ac97_chip = 1} , - {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10020052, - .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102, + .driver = "Audigy", .name = "Audigy 1 [SB0090]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, - .spdif_bug = 1, .ac97_chip = 1} , - {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102, - .driver = "Audigy", .name = "Audigy 1 [SB0090]", + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00521102, + .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, + .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00511102, .driver = "Audigy", .name = "Audigy 1 [SB0090]", -- cgit v1.2.3 From 5b738babf13d51285710ed57336ee5f072ac9490 Mon Sep 17 00:00:00 2001 From: Henrik Kretzschmar Date: Fri, 8 Jul 2005 09:29:18 +0200 Subject: [ALSA] fix compiler warning GUS Library This patch fixes a compiler warning if sequencer is disabled. Signed-off-by: Henrik Kretzschmar Signed-off-by: Jaroslav Kysela --- sound/isa/gus/gus_main.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 94bbd344be5e..a636d9ce3502 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -417,11 +417,13 @@ static int snd_gus_check_version(snd_gus_card_t * gus) return 0; } +#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) static void snd_gus_seq_dev_free(snd_seq_device_t *seq_dev) { snd_gus_card_t *gus = seq_dev->private_data; gus->seq_dev = NULL; } +#endif int snd_gus_initialize(snd_gus_card_t *gus) { -- cgit v1.2.3 From 856def8a4695066e6cbd2919c5987f1df23dbe8a Mon Sep 17 00:00:00 2001 From: Henrik Kretzschmar Date: Fri, 8 Jul 2005 13:53:42 +0200 Subject: [ALSA] typo-fix and snd_assert()-expression-split ALSA Core This patch corrects a typo in the kerneldocs of snd_info_get_str(). It also splits the expressions of snd_assert() in snd_info_unregister() into one-expression-per-call for better debugging. Signed-off-by: Henrik Kretzschmar Signed-off-by: Takashi Iwai --- sound/core/info.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/info.c b/sound/core/info.c index 5e122bbe7c92..7f8bdf7b0058 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -702,7 +702,7 @@ int snd_info_get_line(snd_info_buffer_t * buffer, char *line, int len) } /** - * snd_info_get_line - parse a string token + * snd_info_get_str - parse a string token * @dest: the buffer to store the string token * @src: the original string * @len: the max. length of token - 1 @@ -939,7 +939,8 @@ int snd_info_unregister(snd_info_entry_t * entry) { struct proc_dir_entry *root; - snd_assert(entry != NULL && entry->p != NULL, return -ENXIO); + snd_assert(entry != NULL, return -ENXIO); + snd_assert(entry->p != NULL, return -ENXIO); root = entry->parent == NULL ? snd_proc_root : entry->parent->p; snd_assert(root, return -ENXIO); down(&info_mutex); -- cgit v1.2.3 From 072c01194df6e4843582d09380b780987f642d6d Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sat, 9 Jul 2005 10:07:55 +0200 Subject: [ALSA] ens1371 - added extra delay for ac97 codec initialization ENS1370/1+ driver Signed-off-by: Jaroslav Kysela --- sound/pci/ens1370.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 4e63498a58b2..d4287338c042 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -685,6 +685,15 @@ static unsigned short snd_es1371_codec_read(ac97_t *ac97, return 0; } +static void snd_es1371_codec_wait(ac97_t *ac97) +{ + msleep(750); + snd_es1371_codec_read(ac97, AC97_RESET); + snd_es1371_codec_read(ac97, AC97_VENDOR_ID1); + snd_es1371_codec_read(ac97, AC97_VENDOR_ID2); + msleep(50); +} + static void snd_es1371_adc_rate(ensoniq_t * ensoniq, unsigned int rate) { unsigned int n, truncm, freq, result; @@ -1585,6 +1594,7 @@ static int snd_ensoniq_1371_mixer(ensoniq_t * ensoniq) static ac97_bus_ops_t ops = { .write = snd_es1371_codec_write, .read = snd_es1371_codec_read, + .wait = snd_es1371_codec_wait, }; if ((err = snd_ac97_bus(card, 0, &ops, NULL, &pbus)) < 0) -- cgit v1.2.3 From ef21ca24faf28df6d06939e77d5032a313490289 Mon Sep 17 00:00:00 2001 From: Nishanth Aravamudan Date: Sat, 9 Jul 2005 10:13:22 +0200 Subject: [ALSA] sound/pci: fix-up sleeping paths ENS1370/1+ driver,ES1968 driver,Intel8x0 driver,VIA82xx driver VIA82xx-modem driver,AC97 Codec,ALI5451 driver,CS46xx driver MIXART driver,RME HDSP driver,Trident driver,YMFPCI driver Description: Fix-up sleeping in sound/pci. These changes fall under the following two categories: 1) Replace schedule_timeout() with msleep() to guarantee the task delays as expected. This also involved replacing/removing custom sleep functions. 2) Do not assume jiffies will only increment by one if you request a 1 jiffy sleep, i.e. use time_after/time_before in while loops. Signed-off-by: Nishanth Aravamudan Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_codec.c | 17 +++++++++-------- sound/pci/ali5451/ali5451.c | 4 ++-- sound/pci/cs46xx/cs46xx_lib.c | 15 +++++---------- sound/pci/ens1370.c | 12 +----------- sound/pci/es1968.c | 14 ++++---------- sound/pci/intel8x0.c | 3 +-- sound/pci/mixart/mixart.c | 4 ++-- sound/pci/rme9652/hdsp.c | 6 ++---- sound/pci/trident/trident_main.c | 3 +-- sound/pci/via82xx.c | 13 ++++++------- sound/pci/via82xx_modem.c | 13 ++++++------- sound/pci/ymfpci/ymfpci_main.c | 6 +++--- 12 files changed, 42 insertions(+), 68 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 0677d41239a9..94cd989cff20 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2227,6 +2227,7 @@ void snd_ac97_restore_iec958(ac97_t *ac97) void snd_ac97_resume(ac97_t *ac97) { int i; + unsigned long end_time; if (ac97->bus->ops->reset) { ac97->bus->ops->reset(ac97); @@ -2244,26 +2245,26 @@ void snd_ac97_resume(ac97_t *ac97) snd_ac97_write(ac97, AC97_POWERDOWN, ac97->regs[AC97_POWERDOWN]); if (ac97_is_audio(ac97)) { ac97->bus->ops->write(ac97, AC97_MASTER, 0x8101); - for (i = HZ/10; i >= 0; i--) { + end_time = jiffies + msecs_to_jiffies(100); + do { if (snd_ac97_read(ac97, AC97_MASTER) == 0x8101) break; set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(1); - } + } while (time_after_eq(end_time, jiffies)); /* FIXME: extra delay */ ac97->bus->ops->write(ac97, AC97_MASTER, 0x8000); - if (snd_ac97_read(ac97, AC97_MASTER) != 0x8000) { - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(HZ/4); - } + if (snd_ac97_read(ac97, AC97_MASTER) != 0x8000) + msleep(250); } else { - for (i = HZ/10; i >= 0; i--) { + end_time = jiffies + msecs_to_jiffies(100); + do { unsigned short val = snd_ac97_read(ac97, AC97_EXTENDED_MID); if (val != 0xffff && (val & 1) != 0) break; set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(1); - } + } while (time_after_eq(end_time, jiffies)); } __reset_ready: diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index eb5c36d31a52..f08ae71f902d 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -399,7 +399,7 @@ static int snd_ali_codec_ready( ali_t *codec, unsigned long end_time; unsigned int res; - end_time = jiffies + 10 * (HZ >> 2); + end_time = jiffies + 10 * msecs_to_jiffies(250); do { res = snd_ali_5451_peek(codec,port); if (! (res & 0x8000)) @@ -422,7 +422,7 @@ static int snd_ali_stimer_ready(ali_t *codec, int sched) dwChk1 = snd_ali_5451_peek(codec, ALI_STIMER); dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); - end_time = jiffies + 10 * (HZ >> 2); + end_time = jiffies + 10 * msecs_to_jiffies(250); do { dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); if (dwChk2 != dwChk1) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index fd4c50c88bc9..ff28af1f658e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2400,8 +2400,7 @@ static void snd_cs46xx_codec_reset (ac97_t * ac97) if ((err = snd_ac97_read(ac97, AC97_REC_GAIN)) == 0x8a05) return; - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(HZ/100); + msleep(10); } while (time_after_eq(end_time, jiffies)); snd_printk("CS46xx secondary codec dont respond!\n"); @@ -2435,8 +2434,7 @@ static int __devinit cs46xx_detect_codec(cs46xx_t *chip, int codec) err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97[codec]); return err; } - set_current_state(TASK_INTERRUPTIBLE); - schedule_timeout(HZ/100); + msleep(10); } snd_printdd("snd_cs46xx: codec %d detection timeout\n", codec); return -ENXIO; @@ -3018,8 +3016,7 @@ static int snd_cs46xx_chip_init(cs46xx_t *chip) /* * Wait until the PLL has stabilized. */ - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(HZ/10); /* 100ms */ + msleep(100); /* * Turn on clocking of the core so that we can setup the serial ports. @@ -3072,8 +3069,7 @@ static int snd_cs46xx_chip_init(cs46xx_t *chip) */ if (snd_cs46xx_peekBA0(chip, BA0_ACSTS) & ACSTS_CRDY) goto ok1; - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout((HZ+99)/100); + msleep(10); } @@ -3122,8 +3118,7 @@ static int snd_cs46xx_chip_init(cs46xx_t *chip) */ if ((snd_cs46xx_peekBA0(chip, BA0_ACISV) & (ACISV_ISV3 | ACISV_ISV4)) == (ACISV_ISV3 | ACISV_ISV4)) goto ok2; - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout((HZ+99)/100); + msleep(10); } #ifndef CONFIG_SND_CS46XX_NEW_DSP diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index d4287338c042..78a81f3912a1 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2018,21 +2018,11 @@ static int __devinit snd_ensoniq_create(snd_card_t * card, if (pci->vendor == es1371_ac97_reset_hack[idx].vid && pci->device == es1371_ac97_reset_hack[idx].did && ensoniq->rev == es1371_ac97_reset_hack[idx].rev) { - unsigned long tmo; - signed long tmo2; - ensoniq->cssr |= ES_1371_ST_AC97_RST; outl(ensoniq->cssr, ES_REG(ensoniq, STATUS)); /* need to delay around 20ms(bleech) to give some CODECs enough time to wakeup */ - tmo = jiffies + (HZ / 50) + 1; - while (1) { - tmo2 = tmo - jiffies; - if (tmo2 <= 0) - break; - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(tmo2); - } + msleep(20); break; } /* AC'97 warm reset to start the bitclk */ diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 327a341e276b..9d7a28783930 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -664,11 +664,6 @@ static inline u16 maestro_read(es1968_t *chip, u16 reg) return result; } -#define big_mdelay(msec) do {\ - set_current_state(TASK_UNINTERRUPTIBLE);\ - schedule_timeout(((msec) * HZ + 999) / 1000);\ -} while (0) - /* Wait for the codec bus to be free */ static int snd_es1968_ac97_wait(es1968_t *chip) { @@ -1809,8 +1804,7 @@ static void __devinit es1968_measure_clock(es1968_t *chip) snd_es1968_trigger_apu(chip, apu, ESM_APU_16BITLINEAR); do_gettimeofday(&start_time); spin_unlock_irq(&chip->reg_lock); - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(HZ / 20); /* 50 msec */ + msleep(50); spin_lock_irq(&chip->reg_lock); offset = __apu_get_register(chip, apu, 5); do_gettimeofday(&stop_time); @@ -2093,7 +2087,7 @@ static void snd_es1968_ac97_reset(es1968_t *chip) outw(0x0000, ioaddr + 0x60); /* write 0 to gpio 0 */ udelay(20); outw(0x0001, ioaddr + 0x60); /* write 1 to gpio 1 */ - big_mdelay(20); + msleep(20); outw(save_68 | 0x1, ioaddr + 0x68); /* now restore .. */ outw((inw(ioaddr + 0x38) & 0xfffc) | 0x1, ioaddr + 0x38); @@ -2109,7 +2103,7 @@ static void snd_es1968_ac97_reset(es1968_t *chip) outw(0x0001, ioaddr + 0x60); /* write 1 to gpio */ udelay(20); outw(0x0009, ioaddr + 0x60); /* write 9 to gpio */ - big_mdelay(500); + msleep(500); //outw(inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38); outw(inw(ioaddr + 0x3a) & 0xfffc, ioaddr + 0x3a); outw(inw(ioaddr + 0x3c) & 0xfffc, ioaddr + 0x3c); @@ -2135,7 +2129,7 @@ static void snd_es1968_ac97_reset(es1968_t *chip) if (w > 10000) { outb(inb(ioaddr + 0x37) | 0x08, ioaddr + 0x37); /* do a software reset */ - big_mdelay(500); /* oh my.. */ + msleep(500); /* oh my.. */ outb(inb(ioaddr + 0x37) & ~0x08, ioaddr + 0x37); udelay(1); diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index c3c3b68b4540..7c806bd9cc90 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2464,8 +2464,7 @@ static void __devinit intel8x0_measure_ac97_clock(intel8x0_t *chip) } do_gettimeofday(&start_time); spin_unlock_irq(&chip->reg_lock); - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(HZ / 20); + msleep(50); spin_lock_irq(&chip->reg_lock); /* check the position */ pos = ichdev->fragsize1; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 082c0d0f73d2..6c868d913634 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -445,9 +445,9 @@ static int snd_mixart_trigger(snd_pcm_substream_t *subs, int cmd) static int mixart_sync_nonblock_events(mixart_mgr_t *mgr) { - int timeout = HZ; + unsigned long timeout = jiffies + HZ; while (atomic_read(&mgr->msg_processed) > 0) { - if (! timeout--) { + if (time_after(jiffies, timeout)) { snd_printk(KERN_ERR "mixart: cannot process nonblock events!\n"); return -EBUSY; } diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 0db558a92871..796621de5009 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -679,8 +679,7 @@ static int snd_hdsp_load_firmware_from_cache(hdsp_t *hdsp) { } if ((1000 / HZ) < 3000) { - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout((3000 * HZ + 999) / 1000); + ssleep(3); } else { mdelay(3000); } @@ -5080,8 +5079,7 @@ static int __devinit snd_hdsp_create(snd_card_t *card, if (!is_9652 && !is_9632) { /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ if ((1000 / HZ) < 2000) { - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout((2000 * HZ + 999) / 1000); + ssleep(2); } else { mdelay(2000); } diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index b01c91bb5f6a..29d89bfba0a4 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3153,8 +3153,7 @@ static int snd_trident_gameport_open(struct gameport *gameport, int mode) switch (mode) { case GAMEPORT_MODE_COOKED: outb(GAMEPORT_MODE_ADC, TRID_REG(chip, GAMEPORT_GCR)); - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(1 + 20 * HZ / 1000); /* 20msec */ + msleep(20); return 0; case GAMEPORT_MODE_RAW: outb(0, TRID_REG(chip, GAMEPORT_GCR)); diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 064972b14d00..890582ce874d 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -547,8 +547,7 @@ static void snd_via82xx_codec_wait(ac97_t *ac97) int err; err = snd_via82xx_codec_ready(chip, ac97->num); /* here we need to wait fairly for long time.. */ - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(HZ/2); + msleep(500); } static void snd_via82xx_codec_write(ac97_t *ac97, @@ -1847,7 +1846,7 @@ static void __devinit snd_via82xx_proc_init(via82xx_t *chip) static int snd_via82xx_chip_init(via82xx_t *chip) { unsigned int val; - int max_count; + unsigned long end_time; unsigned char pval; #if 0 /* broken on K7M? */ @@ -1889,14 +1888,14 @@ static int snd_via82xx_chip_init(via82xx_t *chip) } /* wait until codec ready */ - max_count = ((3 * HZ) / 4) + 1; + end_time = jiffies + msecs_to_jiffies(750); do { pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */ break; set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(1); - } while (--max_count > 0); + } while (time_before(jiffies, end_time)); if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY) snd_printk("AC'97 codec is not ready [0x%x]\n", val); @@ -1905,7 +1904,7 @@ static int snd_via82xx_chip_init(via82xx_t *chip) snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ | VIA_REG_AC97_SECONDARY_VALID | (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT)); - max_count = ((3 * HZ) / 4) + 1; + end_time = jiffies + msecs_to_jiffies(750); snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ | VIA_REG_AC97_SECONDARY_VALID | (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT)); @@ -1916,7 +1915,7 @@ static int snd_via82xx_chip_init(via82xx_t *chip) } set_current_state(TASK_INTERRUPTIBLE); schedule_timeout(1); - } while (--max_count > 0); + } while (time_before(jiffies, end_time)); /* This is ok, the most of motherboards have only one codec */ __ac97_ok2: diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 5896d289f9ac..4a9779cc9733 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -408,8 +408,7 @@ static void snd_via82xx_codec_wait(ac97_t *ac97) int err; err = snd_via82xx_codec_ready(chip, ac97->num); /* here we need to wait fairly for long time.. */ - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(HZ/2); + msleep(500); } static void snd_via82xx_codec_write(ac97_t *ac97, @@ -923,7 +922,7 @@ static void __devinit snd_via82xx_proc_init(via82xx_t *chip) static int snd_via82xx_chip_init(via82xx_t *chip) { unsigned int val; - int max_count; + unsigned long end_time; unsigned char pval; pci_read_config_byte(chip->pci, VIA_MC97_CTRL, &pval); @@ -962,14 +961,14 @@ static int snd_via82xx_chip_init(via82xx_t *chip) } /* wait until codec ready */ - max_count = ((3 * HZ) / 4) + 1; + end_time = jiffies + msecs_to_jiffies(750); do { pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */ break; set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(1); - } while (--max_count > 0); + } while (time_before(jiffies, end_time)); if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY) snd_printk("AC'97 codec is not ready [0x%x]\n", val); @@ -977,7 +976,7 @@ static int snd_via82xx_chip_init(via82xx_t *chip) snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ | VIA_REG_AC97_SECONDARY_VALID | (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT)); - max_count = ((3 * HZ) / 4) + 1; + end_time = jiffies + msecs_to_jiffies(750); snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ | VIA_REG_AC97_SECONDARY_VALID | (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT)); @@ -988,7 +987,7 @@ static int snd_via82xx_chip_init(via82xx_t *chip) } set_current_state(TASK_INTERRUPTIBLE); schedule_timeout(1); - } while (--max_count > 0); + } while (time_before(jiffies, end_time)); /* This is ok, the most of motherboards have only one codec */ __ac97_ok2: diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 2ae79610ecb5..d54f88a1b525 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -84,16 +84,16 @@ static inline void snd_ymfpci_writel(ymfpci_t *chip, u32 offset, u32 val) static int snd_ymfpci_codec_ready(ymfpci_t *chip, int secondary) { - signed long end_time; + unsigned long end_time; u32 reg = secondary ? YDSXGR_SECSTATUSADR : YDSXGR_PRISTATUSADR; - end_time = (jiffies + ((3 * HZ) / 4)) + 1; + end_time = jiffies + msecs_to_jiffies(750); do { if ((snd_ymfpci_readw(chip, reg) & 0x8000) == 0) return 0; set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(1); - } while (end_time - (signed long)jiffies >= 0); + } while (time_before(jiffies, end_time)); snd_printk("codec_ready: codec %i is not ready [0x%x]\n", secondary, snd_ymfpci_readw(chip, reg)); return -EBUSY; } -- cgit v1.2.3 From 989a0b248bbf32c89e60dc6f02219e446b320712 Mon Sep 17 00:00:00 2001 From: Nishanth Aravamudan Date: Sat, 9 Jul 2005 10:53:24 +0200 Subject: [ALSA] Fix-up sleeping in sound/ppc PPC AWACS driver,PPC PMAC driver,PPC Tumbler driver Description: Fix-up sleeping in sound/ppc. Replace big_mdelay() with msleep() to guarantee the task delays as expected. This also involved replacing/removing custom sleep functions. Signed-off-by: Nishanth Aravamudan Signed-off-by: Jaroslav Kysela --- sound/ppc/awacs.c | 6 +++--- sound/ppc/pmac.h | 5 ----- sound/ppc/tumbler.c | 16 ++++++++-------- 3 files changed, 11 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 061e52d3d771..758ca1bcbcf2 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -103,7 +103,7 @@ static void screamer_recalibrate(pmac_t *chip) snd_pmac_awacs_write_noreg(chip, 1, chip->awacs_reg[1]); if (chip->manufacturer == 0x1) /* delay for broken crystal part */ - big_mdelay(750); + msleep(750); snd_pmac_awacs_write_noreg(chip, 1, chip->awacs_reg[1] | MASK_RECALIBRATE | MASK_CMUTE | MASK_AMUTE); snd_pmac_awacs_write_noreg(chip, 1, chip->awacs_reg[1]); @@ -653,10 +653,10 @@ static void snd_pmac_awacs_resume(pmac_t *chip) { if (machine_is_compatible("PowerBook3,1") || machine_is_compatible("PowerBook3,2")) { - big_mdelay(100); + msleep(100); snd_pmac_awacs_write_reg(chip, 1, chip->awacs_reg[1] & ~MASK_PAROUT); - big_mdelay(300); + msleep(300); } awacs_restore_all_regs(chip); diff --git a/sound/ppc/pmac.h b/sound/ppc/pmac.h index 582db5220119..ae3bb6c6edff 100644 --- a/sound/ppc/pmac.h +++ b/sound/ppc/pmac.h @@ -212,9 +212,4 @@ int snd_pmac_boolean_mono_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *ui int snd_pmac_add_automute(pmac_t *chip); -#define big_mdelay(msec) do {\ - set_current_state(TASK_UNINTERRUPTIBLE);\ - schedule_timeout(((msec) * HZ + 999) / 1000);\ -} while (0) - #endif /* __PMAC_H */ diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 36c5d5d45bb1..b94437c024b1 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -945,7 +945,7 @@ static void device_change_handler(void *self) check_mute(chip, &mix->line_mute, 0, mix->auto_mute_notify, chip->lineout_sw_ctl); if (mix->anded_reset) - big_mdelay(10); + msleep(10); check_mute(chip, &mix->amp_mute, 1, mix->auto_mute_notify, chip->speaker_sw_ctl); mix->drc_enable = 0; @@ -954,7 +954,7 @@ static void device_change_handler(void *self) check_mute(chip, &mix->amp_mute, 0, mix->auto_mute_notify, chip->speaker_sw_ctl); if (mix->anded_reset) - big_mdelay(10); + msleep(10); check_mute(chip, &mix->hp_mute, 1, mix->auto_mute_notify, chip->master_sw_ctl); if (mix->line_mute.addr != 0) @@ -1109,22 +1109,22 @@ static void tumbler_reset_audio(pmac_t *chip) DBG("(I) codec anded reset !\n"); write_audio_gpio(&mix->hp_mute, 0); write_audio_gpio(&mix->amp_mute, 0); - big_mdelay(200); + msleep(200); write_audio_gpio(&mix->hp_mute, 1); write_audio_gpio(&mix->amp_mute, 1); - big_mdelay(100); + msleep(100); write_audio_gpio(&mix->hp_mute, 0); write_audio_gpio(&mix->amp_mute, 0); - big_mdelay(100); + msleep(100); } else { DBG("(I) codec normal reset !\n"); write_audio_gpio(&mix->audio_reset, 0); - big_mdelay(200); + msleep(200); write_audio_gpio(&mix->audio_reset, 1); - big_mdelay(100); + msleep(100); write_audio_gpio(&mix->audio_reset, 0); - big_mdelay(100); + msleep(100); } } -- cgit v1.2.3 From b27c187f95cd6c9f13f26a5088bea384ac557b45 Mon Sep 17 00:00:00 2001 From: Nishanth Aravamudan Date: Sat, 9 Jul 2005 10:54:37 +0200 Subject: [ALSA] Fix-up sleeping in sound/usb USB generic driver,USB USX2Y Description: Fix-up sleeping in sound/usb. Replace big_mdelay() with msleep() to guarantee the task delays as expected. This also involved replacing/removing custom sleep functions. Patch is compile-tested. Signed-off-by: Nishanth Aravamudan Signed-off-by: Jaroslav Kysela --- sound/usb/usbaudio.c | 4 ++-- sound/usb/usx2y/usX2Yhwdep.c | 3 +-- sound/usb/usx2y/usx2yhwdeppcm.c | 6 ++---- 3 files changed, 5 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 3eaa08e3e6a6..f2b760d8d77e 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -792,7 +792,7 @@ static int start_urbs(snd_usb_substream_t *subs, snd_pcm_runtime_t *runtime) */ static int wait_clear_urbs(snd_usb_substream_t *subs) { - int timeout = HZ; + unsigned long end_time = jiffies + msecs_to_jiffies(1000); unsigned int i; int alive; @@ -812,7 +812,7 @@ static int wait_clear_urbs(snd_usb_substream_t *subs) break; set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(1); - } while (--timeout > 0); + } while (time_before(jiffies, end_time)); if (alive) snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); return 0; diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index bef9b0c142c4..0281a362857a 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -232,8 +232,7 @@ static int snd_usX2Y_hwdep_dsp_load(snd_hwdep_t *hw, snd_hwdep_dsp_image_t *dsp) if (err) return err; if (dsp->index == 1) { - set_current_state(TASK_UNINTERRUPTIBLE); - schedule_timeout(HZ/4); // give the device some time + msleep(250); // give the device some time err = usX2Y_AsyncSeq04_init(priv); if (err) { snd_printk("usX2Y_AsyncSeq04_init error \n"); diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index bb2c8e9000c6..ef28061287f2 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -50,6 +50,7 @@ Currently rawusb dma pcm buffer transport (this file) is only available to snd-usb-usx2y. */ +#include #include "usbusx2yaudio.c" #if defined(USX2Y_NRPACKS_VARIABLE) || (!defined(USX2Y_NRPACKS_VARIABLE) && USX2Y_NRPACKS == 1) @@ -520,11 +521,8 @@ static int snd_usX2Y_usbpcm_prepare(snd_pcm_substream_t *substream) usX2Y->hwdep_pcm_shm->playback_iso_start = -1; if (atomic_read(&subs->state) < state_PREPARED) { while (usX2Y_iso_frames_per_buffer(runtime, usX2Y) > usX2Y->hwdep_pcm_shm->captured_iso_frames) { - signed long timeout; snd_printd("Wait: iso_frames_per_buffer=%i,captured_iso_frames=%i\n", usX2Y_iso_frames_per_buffer(runtime, usX2Y), usX2Y->hwdep_pcm_shm->captured_iso_frames); - set_current_state(TASK_INTERRUPTIBLE); - timeout = schedule_timeout(HZ/100 + 1); - if (signal_pending(current)) { + if (msleep_interruptible(10)) { err = -ERESTARTSYS; goto up_prepare_mutex; } -- cgit v1.2.3 From 7c1d549aa9b22365fe5405c372f840cdbc6315f5 Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sun, 10 Jul 2005 11:50:36 +0200 Subject: [ALSA] emu10k1: Add EMU 1212m card entry and document it as not supported yet. EMU10K1/EMU10K2 driver Signed-off-by: James Courtier-Dutton --- sound/pci/emu10k1/emu10k1_main.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 7c31d9b30240..746b51ef3966 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -630,6 +630,13 @@ static emu_chip_details_t emu_chip_details[] = { .emu10k2_chip = 1, .ca0108_chip = 1, .ac97_chip = 1} , + /* Tested by James@superbug.co.uk 8th July 2005. No sound available yet. */ + {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, + .driver = "Audigy2", .name = "E-mu 1212m [4001]", + .id = "EMU1212m", + .emu10k2_chip = 1, + .ca0102_chip = 1, + .ecard = 1} , /* Tested by James@superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102, .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", -- cgit v1.2.3 From c9eab129fcbcef364b34fb3a70cb2531847e1edf Mon Sep 17 00:00:00 2001 From: James Courtier-Dutton Date: Sun, 10 Jul 2005 12:04:29 +0200 Subject: [ALSA] ac97: Fix volume control bit size detection for STAC9704. AC97 Codec Signed-off-by: James Courtier-Dutton --- sound/pci/ac97/ac97_codec.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 94cd989cff20..1f09653dc0f3 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1078,6 +1078,11 @@ static void check_volume_resolution(ac97_t *ac97, int reg, unsigned char *lo_max for (i = 0 ; i < ARRAY_SIZE(cbit); i++) { unsigned short val; snd_ac97_write(ac97, reg, 0x8080 | cbit[i] | (cbit[i] << 8)); + /* Do the read twice due to buffers on some ac97 codecs. + * e.g. The STAC9704 returns exactly what you wrote the the register + * if you read it immediately. This causes the detect routine to fail. + */ + val = snd_ac97_read(ac97, reg); val = snd_ac97_read(ac97, reg); if (! *lo_max && (val & 0x7f) == cbit[i]) *lo_max = max[i]; -- cgit v1.2.3 From 7858ffa062886706026cfff3ba80b8400b520501 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 11 Jul 2005 15:37:19 +0200 Subject: [ALSA] ac97 - remove unused variable AC97 Codec remove a variable made obsolete by the last change Signed-off-by: Clemens Ladisch --- sound/pci/ac97/ac97_codec.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 1f09653dc0f3..6983eea226da 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2231,7 +2231,6 @@ void snd_ac97_restore_iec958(ac97_t *ac97) */ void snd_ac97_resume(ac97_t *ac97) { - int i; unsigned long end_time; if (ac97->bus->ops->reset) { -- cgit v1.2.3 From d06e4c4001cf26147a6af0718703368944f0df32 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 21 Jul 2005 08:01:22 +0200 Subject: [ALSA] seq-midi - silently ignore non-MIDI events ALSA sequencer When non-MIDI sequencer events are sent to a RawMIDI port, silently ignore them instead of returning a confusing error code which may upset the sequencer and abort the current write() to /dev/snd/seq. Signed-off-by: Clemens Ladisch --- sound/core/seq/seq_midi.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 57be9155eb62..4374829ea770 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -134,7 +134,7 @@ static int event_process_midi(snd_seq_event_t * ev, int direct, seq_midisynth_t *msynth = (seq_midisynth_t *) private_data; unsigned char msg[10]; /* buffer for constructing midi messages */ snd_rawmidi_substream_t *substream; - int res; + int len; snd_assert(msynth != NULL, return -EINVAL); substream = msynth->output_rfile.output; @@ -146,20 +146,16 @@ static int event_process_midi(snd_seq_event_t * ev, int direct, snd_printd("seq_midi: invalid sysex event flags = 0x%x\n", ev->flags); return 0; } - res = snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)dump_midi, substream); + snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)dump_midi, substream); snd_midi_event_reset_decode(msynth->parser); - if (res < 0) - return res; } else { if (msynth->parser == NULL) return -EIO; - res = snd_midi_event_decode(msynth->parser, msg, sizeof(msg), ev); - if (res < 0) - return res; - if ((res = dump_midi(substream, msg, res)) < 0) { + len = snd_midi_event_decode(msynth->parser, msg, sizeof(msg), ev); + if (len < 0) + return 0; + if (dump_midi(substream, msg, len) < 0) snd_midi_event_reset_decode(msynth->parser); - return res; - } } return 0; } -- cgit v1.2.3 From f38275fe994c333b809796230f4f98090f8d919b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 Jul 2005 16:17:29 +0200 Subject: [ALSA] usb-audio - add support for Miditech USB MIDI keyboards USB generic driver Add support for Miditech Midistart and MidiStudio keyboards (another case of devices using the standard protocol but having no descriptors). Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 1 + sound/usb/usbaudio.h | 3 +++ sound/usb/usbmidi.c | 3 +++ sound/usb/usbquirks.h | 19 +++++++++++++++++++ 4 files changed, 26 insertions(+) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index f2b760d8d77e..9a0b0899d156 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2972,6 +2972,7 @@ static int snd_usb_create_quirk(snd_usb_audio_t *chip, case QUIRK_MIDI_NOVATION: case QUIRK_MIDI_RAW: case QUIRK_MIDI_EMAGIC: + case QUIRK_MIDI_MIDITECH: return snd_usb_create_midi_interface(chip, iface, quirk); case QUIRK_COMPOSITE: return create_composite_quirk(chip, iface, quirk); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a87912205477..c1415f4b6004 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -167,6 +167,7 @@ struct snd_usb_audio { #define QUIRK_MIDI_NOVATION 10 #define QUIRK_MIDI_RAW 11 #define QUIRK_MIDI_EMAGIC 12 +#define QUIRK_MIDI_MIDITECH 13 typedef struct snd_usb_audio_quirk snd_usb_audio_quirk_t; typedef struct snd_usb_midi_endpoint_info snd_usb_midi_endpoint_info_t; @@ -210,6 +211,8 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_EMAGIC, data points to a snd_usb_midi_endpoint_info * structure (out_cables and in_cables only) */ +/* for QUIRK_MIDI_MIDITECH, data is NULL */ + /* */ diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 5c754879c2b6..5778a9b725ec 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1515,6 +1515,9 @@ int snd_usb_create_midi_interface(snd_usb_audio_t* chip, sizeof(snd_usb_midi_endpoint_info_t)); err = snd_usbmidi_detect_endpoints(umidi, &endpoints[0], 1); break; + case QUIRK_MIDI_MIDITECH: + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 67796430e588..f74e652a1e51 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1378,6 +1378,25 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + USB_DEVICE(0x4752, 0x0011), + .driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) { + .vendor_name = "Miditech", + .product_name = "Midistart-2", + .ifnum = 0, + .type = QUIRK_MIDI_MIDITECH + } +}, +{ + USB_DEVICE(0x7104, 0x2202), + .driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) { + .vendor_name = "Miditech", + .product_name = "MidiStudio-2", + .ifnum = 0, + .type = QUIRK_MIDI_MIDITECH + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- cgit v1.2.3 From 854af9578cb84e4ca3cb1551a6be40c4e81bb455 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 Jul 2005 16:19:10 +0200 Subject: [ALSA] usb-audio - change quirk type handling USB generic driver Make the quirk type an enum instead of a #defined integer, and use a table for the quirk constructor functions instead of a big switch statement. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 59 ++++++++++++++++++++++++++++++---------------------- sound/usb/usbaudio.h | 35 +++++++++++++++++-------------- 2 files changed, 53 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 9a0b0899d156..8298c462c291 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2735,7 +2735,8 @@ static int create_standard_interface_quirk(snd_usb_audio_t *chip, * to detect the sample rate is by looking at wMaxPacketSize. */ static int create_ua700_ua25_quirk(snd_usb_audio_t *chip, - struct usb_interface *iface) + struct usb_interface *iface, + const snd_usb_audio_quirk_t *quirk) { static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, @@ -2826,7 +2827,9 @@ static int create_ua700_ua25_quirk(snd_usb_audio_t *chip, /* * Create a stream for an Edirol UA-1000 interface. */ -static int create_ua1000_quirk(snd_usb_audio_t *chip, struct usb_interface *iface) +static int create_ua1000_quirk(snd_usb_audio_t *chip, + struct usb_interface *iface, + const snd_usb_audio_quirk_t *quirk) { static const struct audioformat ua1000_format = { .format = SNDRV_PCM_FORMAT_S32_LE, @@ -2903,6 +2906,13 @@ static int create_composite_quirk(snd_usb_audio_t *chip, return 0; } +static int ignore_interface_quirk(snd_usb_audio_t *chip, + struct usb_interface *iface, + const snd_usb_audio_quirk_t *quirk) +{ + return 0; +} + /* * boot quirks @@ -2965,29 +2975,28 @@ static int snd_usb_create_quirk(snd_usb_audio_t *chip, struct usb_interface *iface, const snd_usb_audio_quirk_t *quirk) { - switch (quirk->type) { - case QUIRK_MIDI_FIXED_ENDPOINT: - case QUIRK_MIDI_YAMAHA: - case QUIRK_MIDI_MIDIMAN: - case QUIRK_MIDI_NOVATION: - case QUIRK_MIDI_RAW: - case QUIRK_MIDI_EMAGIC: - case QUIRK_MIDI_MIDITECH: - return snd_usb_create_midi_interface(chip, iface, quirk); - case QUIRK_COMPOSITE: - return create_composite_quirk(chip, iface, quirk); - case QUIRK_AUDIO_FIXED_ENDPOINT: - return create_fixed_stream_quirk(chip, iface, quirk); - case QUIRK_AUDIO_STANDARD_INTERFACE: - case QUIRK_MIDI_STANDARD_INTERFACE: - return create_standard_interface_quirk(chip, iface, quirk); - case QUIRK_AUDIO_EDIROL_UA700_UA25: - return create_ua700_ua25_quirk(chip, iface); - case QUIRK_AUDIO_EDIROL_UA1000: - return create_ua1000_quirk(chip, iface); - case QUIRK_IGNORE_INTERFACE: - return 0; - default: + typedef int (*quirk_func_t)(snd_usb_audio_t *, struct usb_interface *, + const snd_usb_audio_quirk_t *); + static const quirk_func_t quirk_funcs[] = { + [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk, + [QUIRK_COMPOSITE] = create_composite_quirk, + [QUIRK_MIDI_STANDARD_INTERFACE] = snd_usb_create_midi_interface, + [QUIRK_MIDI_FIXED_ENDPOINT] = snd_usb_create_midi_interface, + [QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface, + [QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface, + [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface, + [QUIRK_MIDI_RAW] = snd_usb_create_midi_interface, + [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface, + [QUIRK_MIDI_MIDITECH] = snd_usb_create_midi_interface, + [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_interface_quirk, + [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, + [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, + [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, + }; + + if (quirk->type < QUIRK_TYPE_COUNT) { + return quirk_funcs[quirk->type](chip, iface, quirk); + } else { snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); return -ENXIO; } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index c1415f4b6004..ad9eab211d8f 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -153,21 +153,24 @@ struct snd_usb_audio { #define QUIRK_NO_INTERFACE -2 #define QUIRK_ANY_INTERFACE -1 -/* quirk type */ -#define QUIRK_MIDI_FIXED_ENDPOINT 0 -#define QUIRK_MIDI_YAMAHA 1 -#define QUIRK_MIDI_MIDIMAN 2 -#define QUIRK_COMPOSITE 3 -#define QUIRK_AUDIO_FIXED_ENDPOINT 4 -#define QUIRK_AUDIO_STANDARD_INTERFACE 5 -#define QUIRK_MIDI_STANDARD_INTERFACE 6 -#define QUIRK_AUDIO_EDIROL_UA700_UA25 7 -#define QUIRK_AUDIO_EDIROL_UA1000 8 -#define QUIRK_IGNORE_INTERFACE 9 -#define QUIRK_MIDI_NOVATION 10 -#define QUIRK_MIDI_RAW 11 -#define QUIRK_MIDI_EMAGIC 12 -#define QUIRK_MIDI_MIDITECH 13 +enum quirk_type { + QUIRK_IGNORE_INTERFACE, + QUIRK_COMPOSITE, + QUIRK_MIDI_STANDARD_INTERFACE, + QUIRK_MIDI_FIXED_ENDPOINT, + QUIRK_MIDI_YAMAHA, + QUIRK_MIDI_MIDIMAN, + QUIRK_MIDI_NOVATION, + QUIRK_MIDI_RAW, + QUIRK_MIDI_EMAGIC, + QUIRK_MIDI_MIDITECH, + QUIRK_AUDIO_STANDARD_INTERFACE, + QUIRK_AUDIO_FIXED_ENDPOINT, + QUIRK_AUDIO_EDIROL_UA700_UA25, + QUIRK_AUDIO_EDIROL_UA1000, + + QUIRK_TYPE_COUNT +}; typedef struct snd_usb_audio_quirk snd_usb_audio_quirk_t; typedef struct snd_usb_midi_endpoint_info snd_usb_midi_endpoint_info_t; @@ -176,7 +179,7 @@ struct snd_usb_audio_quirk { const char *vendor_name; const char *product_name; int16_t ifnum; - int16_t type; + uint16_t type; const void *data; }; -- cgit v1.2.3 From 5a0f217d96656068f0f1e5cda16c35945f979b16 Mon Sep 17 00:00:00 2001 From: Victor Fusco Date: Tue, 26 Jul 2005 13:42:31 +0200 Subject: [ALSA] sound/core Fix the sparse warning 'implicit cast to nocast type' Memalloc module,ALSA Core,Instrument layer Fix the sparse warning 'implicit cast to nocast type' File/Subsystem:sound/core Signed-off-by: Victor Fusco Signed-off-by: Domen Puncer Signed-off-by: Jaroslav Kysela --- sound/core/memalloc.c | 3 ++- sound/core/memory.c | 8 ++++---- sound/core/seq/instr/ainstr_gf1.c | 3 ++- sound/core/seq/instr/ainstr_iw.c | 8 +++++--- sound/core/wrappers.c | 2 +- 5 files changed, 14 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index dbc23e35fa06..02132561c3f8 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -105,7 +105,8 @@ struct snd_mem_list { */ static void *snd_dma_hack_alloc_coherent(struct device *dev, size_t size, - dma_addr_t *dma_handle, int flags) + dma_addr_t *dma_handle, + unsigned int __nocast flags) { void *ret; u64 dma_mask, coherent_dma_mask; diff --git a/sound/core/memory.c b/sound/core/memory.c index c1fb28e84330..f6895577bf86 100644 --- a/sound/core/memory.c +++ b/sound/core/memory.c @@ -89,7 +89,7 @@ void snd_memory_done(void) } } -static void *__snd_kmalloc(size_t size, int flags, void *caller) +static void *__snd_kmalloc(size_t size, unsigned int __nocast flags, void *caller) { unsigned long cpu_flags; struct snd_alloc_track *t; @@ -111,12 +111,12 @@ static void *__snd_kmalloc(size_t size, int flags, void *caller) } #define _snd_kmalloc(size, flags) __snd_kmalloc((size), (flags), __builtin_return_address(0)); -void *snd_hidden_kmalloc(size_t size, int flags) +void *snd_hidden_kmalloc(size_t size, unsigned int __nocast flags) { return _snd_kmalloc(size, flags); } -void *snd_hidden_kcalloc(size_t n, size_t size, int flags) +void *snd_hidden_kcalloc(size_t n, size_t size, unsigned int __nocast flags) { void *ret = NULL; if (n != 0 && size > INT_MAX / n) @@ -184,7 +184,7 @@ void snd_hidden_vfree(void *obj) snd_wrapper_vfree(obj); } -char *snd_hidden_kstrdup(const char *s, int flags) +char *snd_hidden_kstrdup(const char *s, unsigned int __nocast flags) { int len; char *buf; diff --git a/sound/core/seq/instr/ainstr_gf1.c b/sound/core/seq/instr/ainstr_gf1.c index 0779c41ca037..32e91c6b25fe 100644 --- a/sound/core/seq/instr/ainstr_gf1.c +++ b/sound/core/seq/instr/ainstr_gf1.c @@ -50,7 +50,8 @@ static int snd_seq_gf1_copy_wave_from_stream(snd_gf1_ops_t *ops, { gf1_wave_t *wp, *prev; gf1_xwave_t xp; - int err, gfp_mask; + int err; + unsigned int gfp_mask; unsigned int real_size; gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL; diff --git a/sound/core/seq/instr/ainstr_iw.c b/sound/core/seq/instr/ainstr_iw.c index 39ff72b2aab3..2622b8679ca7 100644 --- a/sound/core/seq/instr/ainstr_iw.c +++ b/sound/core/seq/instr/ainstr_iw.c @@ -58,7 +58,7 @@ static int snd_seq_iwffff_copy_env_from_stream(__u32 req_stype, iwffff_xenv_t *ex, char __user **data, long *len, - int gfp_mask) + unsigned int __nocast gfp_mask) { __u32 stype; iwffff_env_record_t *rp, *rp_last; @@ -128,7 +128,8 @@ static int snd_seq_iwffff_copy_wave_from_stream(snd_iwffff_ops_t *ops, { iwffff_wave_t *wp, *prev; iwffff_xwave_t xp; - int err, gfp_mask; + int err; + unsigned int gfp_mask; unsigned int real_size; gfp_mask = atomic ? GFP_ATOMIC : GFP_KERNEL; @@ -234,7 +235,8 @@ static int snd_seq_iwffff_put(void *private_data, snd_seq_kinstr_t *instr, iwffff_xinstrument_t ix; iwffff_layer_t *lp, *prev_lp; iwffff_xlayer_t lx; - int err, gfp_mask; + int err; + unsigned int gfp_mask; if (cmd != SNDRV_SEQ_INSTR_PUT_CMD_CREATE) return -EINVAL; diff --git a/sound/core/wrappers.c b/sound/core/wrappers.c index 9f393023c327..508e6d67ee19 100644 --- a/sound/core/wrappers.c +++ b/sound/core/wrappers.c @@ -27,7 +27,7 @@ #include #ifdef CONFIG_SND_DEBUG_MEMORY -void *snd_wrapper_kmalloc(size_t size, int flags) +void *snd_wrapper_kmalloc(size_t size, unsigned int __nocast flags) { return kmalloc(size, flags); } -- cgit v1.2.3 From fb92e6f05e84f6c217d786208e2ed5acf633b6ce Mon Sep 17 00:00:00 2001 From: Nicolas Graziano Date: Wed, 27 Jul 2005 17:25:08 +0200 Subject: [ALSA] hda driver, correct bug in model 'auto' HDA Codec driver - Correct some index variable inversion in patch_cmedia.c Signed-off-by: Nicolas Graziano Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cmedia.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 2d6e3e3d0a38..86f195f19eef 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -408,7 +408,7 @@ static int cmi9880_fill_multi_dac_nids(struct hda_codec *codec, const struct aut /* search for an empty channel */ for (j = 0; j < cfg->line_outs; j++) { if (! assigned[j]) { - spec->dac_nids[i] = i + 0x03; + spec->dac_nids[i] = j + 0x03; assigned[j] = 1; break; } @@ -444,11 +444,10 @@ static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pi len = snd_hda_get_connections(codec, nid, conn, 4); for (k = 0; k < len; k++) if (conn[k] == spec->dac_nids[i]) { - spec->multi_init[j].param = j; + spec->multi_init[j].param = k; break; } j++; - break; } } return 0; -- cgit v1.2.3 From 7b566054b33474cdd674289a8c7dd282c02e536e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jul 2005 17:26:59 +0200 Subject: [ALSA] vx-driver - Fix the calculation of frequency parameter Digigram VX core Fixed the calculation of frequency parameter of vx boards. Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_uer.c | 46 ++++++++++++++++++---------------------------- 1 file changed, 18 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/drivers/vx/vx_uer.c b/sound/drivers/vx/vx_uer.c index 18114713c3b3..4fc38bde34f4 100644 --- a/sound/drivers/vx/vx_uer.c +++ b/sound/drivers/vx/vx_uer.c @@ -162,34 +162,24 @@ static int vx_read_uer_status(vx_core_t *chip, int *mode) static int vx_calc_clock_from_freq(vx_core_t *chip, int freq) { -#define XX_FECH48000 0x0000004B -#define XX_FECH32000 0x00000171 -#define XX_FECH24000 0x0000024B -#define XX_FECH16000 0x00000371 -#define XX_FECH12000 0x0000044B -#define XX_FECH8000 0x00000571 -#define XX_FECH44100 0x0000007F -#define XX_FECH29400 0x0000016F -#define XX_FECH22050 0x0000027F -#define XX_FECH14000 0x000003EF -#define XX_FECH11025 0x0000047F -#define XX_FECH7350 0x000005BF - - switch (freq) { - case 48000: return XX_FECH48000; - case 44100: return XX_FECH44100; - case 32000: return XX_FECH32000; - case 29400: return XX_FECH29400; - case 24000: return XX_FECH24000; - case 22050: return XX_FECH22050; - case 16000: return XX_FECH16000; - case 14000: return XX_FECH14000; - case 12000: return XX_FECH12000; - case 11025: return XX_FECH11025; - case 8000: return XX_FECH8000; - case 7350: return XX_FECH7350; - default: return freq; /* The value is already correct */ - } + int hexfreq; + + snd_assert(freq > 0, return 0); + + hexfreq = (28224000 * 10) / freq; + hexfreq = (hexfreq + 5) / 10; + + /* max freq = 55125 Hz */ + snd_assert(hexfreq > 0x00000200, return 0); + + if (hexfreq <= 0x03ff) + return hexfreq - 0x00000201; + if (hexfreq <= 0x07ff) + return (hexfreq / 2) - 1; + if (hexfreq <= 0x0fff) + return (hexfreq / 4) + 0x000001ff; + + return 0x5fe; /* min freq = 6893 Hz */ } -- cgit v1.2.3 From eeacb5457cf5f0802fb29f385befa0b1d166cadb Mon Sep 17 00:00:00 2001 From: Sergey Ulanov Date: Wed, 27 Jul 2005 17:28:58 +0200 Subject: [ALSA] Jack Sense support for AD1980 and AD1888 AC97 Codec Attached patch adds 'Jack Sense' controls for AD1980 and AD1888 chips. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index a15eb8522b7c..66edc857d3e6 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1528,6 +1528,9 @@ static const snd_kcontrol_new_t snd_ac97_ad1888_controls[] = { }, AC97_SURROUND_JACK_MODE_CTL, AC97_CHANNEL_MODE_CTL, + + AC97_SINGLE("Headphone Jack Sense", AC97_AD_JACK_SPDIF, 10, 1, 0), + AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 12, 1, 0), }; static int patch_ad1888_specific(ac97_t *ac97) -- cgit v1.2.3 From 69c3e5f8562c7854d9dd8d7820a89286f9440e41 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jul 2005 17:30:14 +0200 Subject: [ALSA] via82xx - Fix dxs_support of twinhead laptop VIA82xx driver Changed the dxs_support value of twinhead laptop to DXS_SRC. Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 890582ce874d..4889600387c8 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2177,7 +2177,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x147b, .subdevice = 0x1413, .action = VIA_DXS_ENABLE }, /* ABIT KV8 Pro */ { .subvendor = 0x147b, .subdevice = 0x1415, .action = VIA_DXS_NO_VRA }, /* Abit AV8 */ { .subvendor = 0x14ff, .subdevice = 0x0403, .action = VIA_DXS_ENABLE }, /* Twinhead mobo */ - { .subvendor = 0x14ff, .subdevice = 0x0408, .action = VIA_DXS_NO_VRA }, /* Twinhead mobo */ + { .subvendor = 0x14ff, .subdevice = 0x0408, .action = VIA_DXS_SRC }, /* Twinhead laptop */ { .subvendor = 0x1584, .subdevice = 0x8120, .action = VIA_DXS_ENABLE }, /* Gericom/Targa/Vobis/Uniwill laptop */ { .subvendor = 0x1584, .subdevice = 0x8123, .action = VIA_DXS_NO_VRA }, /* Uniwill (Targa Visionary XP-210) */ { .subvendor = 0x161f, .subdevice = 0x202b, .action = VIA_DXS_NO_VRA }, /* Amira Note book */ -- cgit v1.2.3 From da8ea98b21236f29a5df723e3cc5abdc6530c07c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Jul 2005 15:22:55 +0200 Subject: [ALSA] wavefront - declare initialization data as static Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/isa/wavefront/wavefront_fx.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c index 0e13623f69f0..32379688eed4 100644 --- a/sound/isa/wavefront/wavefront_fx.c +++ b/sound/isa/wavefront/wavefront_fx.c @@ -34,7 +34,7 @@ /* weird stuff, derived from port I/O tracing with dosemu */ -unsigned char page_zero[] __initdata = { +static unsigned char page_zero[] __initdata = { 0x01, 0x7c, 0x00, 0x1e, 0x00, 0x00, 0x00, 0x00, 0x00, 0xf5, 0x00, 0x11, 0x00, 0x20, 0x00, 0x32, 0x00, 0x40, 0x00, 0x13, 0x00, 0x00, 0x00, 0x14, 0x02, 0x76, 0x00, 0x60, 0x00, 0x80, 0x02, 0x00, 0x00, @@ -61,7 +61,7 @@ unsigned char page_zero[] __initdata = { 0x1d, 0x02, 0xdf }; -unsigned char page_one[] __initdata = { +static unsigned char page_one[] __initdata = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x19, 0x00, 0x1f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0xd8, 0x00, 0x00, 0x02, 0x20, 0x00, 0x19, 0x00, 0x00, 0x00, 0x00, 0x00, 0x18, 0x01, @@ -88,7 +88,7 @@ unsigned char page_one[] __initdata = { 0x60, 0x00, 0x1b }; -unsigned char page_two[] __initdata = { +static unsigned char page_two[] __initdata = { 0xc4, 0x00, 0x44, 0x07, 0x44, 0x00, 0x40, 0x25, 0x01, 0x06, 0xc4, 0x07, 0x40, 0x25, 0x01, 0x00, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, @@ -103,7 +103,7 @@ unsigned char page_two[] __initdata = { 0x46, 0x05, 0x46, 0x07, 0x46, 0x07, 0x44 }; -unsigned char page_three[] __initdata = { +static unsigned char page_three[] __initdata = { 0x07, 0x40, 0x00, 0x00, 0x00, 0x47, 0x00, 0x40, 0x00, 0x40, 0x06, 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, @@ -118,7 +118,7 @@ unsigned char page_three[] __initdata = { 0x02, 0x00, 0x42, 0x00, 0xc0, 0x00, 0x40 }; -unsigned char page_four[] __initdata = { +static unsigned char page_four[] __initdata = { 0x63, 0x03, 0x26, 0x02, 0x2c, 0x00, 0x24, 0x00, 0x2e, 0x02, 0x02, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, @@ -133,7 +133,7 @@ unsigned char page_four[] __initdata = { 0x02, 0x62, 0x02, 0x20, 0x01, 0x21, 0x01 }; -unsigned char page_six[] __initdata = { +static unsigned char page_six[] __initdata = { 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x04, 0x00, 0x00, 0x06, 0x00, 0x00, 0x08, 0x00, 0x00, 0x0a, 0x00, 0x00, 0x0c, 0x00, 0x00, 0x0e, 0x00, 0x00, 0x10, 0x00, 0x00, 0x12, 0x00, 0x00, 0x14, 0x00, 0x00, @@ -154,7 +154,7 @@ unsigned char page_six[] __initdata = { 0x80, 0x00, 0x7e, 0x80, 0x80 }; -unsigned char page_seven[] __initdata = { +static unsigned char page_seven[] __initdata = { 0x0f, 0xff, 0x00, 0x00, 0x08, 0x00, 0x08, 0x00, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x0f, 0xff, 0x00, 0x00, 0x00, 0x00, 0x08, 0x00, 0x08, 0x00, 0x00, 0x00, 0x0f, 0xff, 0x00, 0x00, 0x00, 0x00, 0x0f, @@ -181,7 +181,7 @@ unsigned char page_seven[] __initdata = { 0x00, 0x02, 0x00 }; -unsigned char page_zero_v2[] __initdata = { +static unsigned char page_zero_v2[] __initdata = { 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, @@ -193,7 +193,7 @@ unsigned char page_zero_v2[] __initdata = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 }; -unsigned char page_one_v2[] __initdata = { +static unsigned char page_one_v2[] __initdata = { 0x01, 0xc0, 0x01, 0xfa, 0x00, 0x1a, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, @@ -205,21 +205,21 @@ unsigned char page_one_v2[] __initdata = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 }; -unsigned char page_two_v2[] __initdata = { +static unsigned char page_two_v2[] __initdata = { 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 }; -unsigned char page_three_v2[] __initdata = { +static unsigned char page_three_v2[] __initdata = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 }; -unsigned char page_four_v2[] __initdata = { +static unsigned char page_four_v2[] __initdata = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, @@ -227,7 +227,7 @@ unsigned char page_four_v2[] __initdata = { 0x00, 0x00, 0x00, 0x00 }; -unsigned char page_seven_v2[] __initdata = { +static unsigned char page_seven_v2[] __initdata = { 0x0f, 0xff, 0x0f, 0xff, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, @@ -239,7 +239,7 @@ unsigned char page_seven_v2[] __initdata = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 }; -unsigned char mod_v2[] __initdata = { +static unsigned char mod_v2[] __initdata = { 0x01, 0x00, 0x02, 0x00, 0x01, 0x01, 0x02, 0x00, 0x01, 0x02, 0x02, 0x00, 0x01, 0x03, 0x02, 0x00, 0x01, 0x04, 0x02, 0x00, 0x01, 0x05, 0x02, 0x00, 0x01, 0x06, 0x02, 0x00, 0x01, 0x07, 0x02, 0x00, 0xb0, @@ -269,7 +269,7 @@ unsigned char mod_v2[] __initdata = { 0x02, 0x01, 0x01, 0x04, 0x02, 0x01, 0x01, 0x05, 0x02, 0x01, 0x01, 0x06, 0x02, 0x01, 0x01, 0x07, 0x02, 0x01 }; -unsigned char coefficients[] __initdata = { +static unsigned char coefficients[] __initdata = { 0x07, 0x46, 0x00, 0x00, 0x07, 0x49, 0x00, 0x00, 0x00, 0x4b, 0x03, 0x11, 0x00, 0x4d, 0x01, 0x32, 0x07, 0x46, 0x00, 0x00, 0x07, 0x49, 0x00, 0x00, 0x07, 0x40, 0x00, 0x00, 0x07, 0x41, 0x00, 0x00, 0x01, @@ -305,14 +305,14 @@ unsigned char coefficients[] __initdata = { 0x06, 0x6c, 0x4c, 0x6c, 0x06, 0x50, 0x52, 0xe2, 0x06, 0x42, 0x02, 0xba }; -unsigned char coefficients2[] __initdata = { +static unsigned char coefficients2[] __initdata = { 0x07, 0x46, 0x00, 0x00, 0x07, 0x49, 0x00, 0x00, 0x07, 0x45, 0x0f, 0xff, 0x07, 0x48, 0x0f, 0xff, 0x07, 0x7b, 0x04, 0xcc, 0x07, 0x7d, 0x04, 0xcc, 0x07, 0x7c, 0x00, 0x00, 0x07, 0x7e, 0x00, 0x00, 0x07, 0x46, 0x00, 0x00, 0x07, 0x49, 0x00, 0x00, 0x07, 0x47, 0x00, 0x00, 0x07, 0x4a, 0x00, 0x00, 0x07, 0x4c, 0x00, 0x00, 0x07, 0x4e, 0x00, 0x00 }; -unsigned char coefficients3[] __initdata = { +static unsigned char coefficients3[] __initdata = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x28, 0x00, 0x28, 0x00, 0x51, 0x00, 0x51, 0x00, 0x7a, 0x00, 0x7a, 0x00, 0xa3, 0x00, 0xa3, 0x00, 0xcc, 0x00, 0xcc, 0x00, 0xf5, 0x00, 0xf5, 0x01, 0x1e, 0x01, 0x1e, 0x01, -- cgit v1.2.3