diff options
Diffstat (limited to 'sound/pci')
83 files changed, 1506 insertions, 418 deletions
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index a2081803a82..d37346b12dc 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -216,14 +216,19 @@ config SND_CS46XX_NEW_DSP This works better than the old code, so say Y. config SND_CS5535AUDIO - tristate "CS5535 Audio" + tristate "CS5535/CS5536 Audio" depends on SND && X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help Say Y here to include support for audio on CS5535 chips. It is referred to as NS CS5535 IO or AMD CS5535 IO companion in - various literature. + various literature. This driver also supports the CS5536 audio + device. However, for both chips, on certain boards, you may + need to use ac97_quirk=hp_only if your board has physically + mapped headphone out to master output. If that works for you, + send lspci -vvv output to the mailing list so that your board + can be identified in the quirks list. To compile this driver as a module, choose M here: the module will be called snd-cs5535audio. diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index d05200741ac..0abf2808d59 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -253,6 +253,8 @@ void snd_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short va ac97->bus->ops->write(ac97, reg, value); } +EXPORT_SYMBOL(snd_ac97_write); + /** * snd_ac97_read - read a value from the given register * @@ -281,6 +283,8 @@ static inline unsigned short snd_ac97_read_cache(struct snd_ac97 *ac97, unsigned return ac97->regs[reg]; } +EXPORT_SYMBOL(snd_ac97_read); + /** * snd_ac97_write_cache - write a value on the given register and update the cache * @ac97: the ac97 instance @@ -302,6 +306,8 @@ void snd_ac97_write_cache(struct snd_ac97 *ac97, unsigned short reg, unsigned sh mutex_unlock(&ac97->reg_mutex); } +EXPORT_SYMBOL(snd_ac97_write_cache); + /** * snd_ac97_update - update the value on the given register * @ac97: the ac97 instance @@ -331,6 +337,8 @@ int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short va return change; } +EXPORT_SYMBOL(snd_ac97_update); + /** * snd_ac97_update_bits - update the bits on the given register * @ac97: the ac97 instance @@ -356,6 +364,8 @@ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned sho return change; } +EXPORT_SYMBOL(snd_ac97_update_bits); + /* no lock version - see snd_ac97_updat_bits() */ int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value) @@ -563,7 +573,7 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) }; static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = - AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0); + AC97_SINGLE("Mic Boost (+20dB) Switch", AC97_MIC, 6, 1, 0); static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; @@ -605,7 +615,7 @@ AC97_SINGLE("Simulated Stereo Enhancement", AC97_GENERAL_PURPOSE, 14, 1, 0), AC97_SINGLE("3D Control - Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), AC97_SINGLE("Loudness (bass boost)", AC97_GENERAL_PURPOSE, 12, 1, 0), AC97_ENUM("Mono Output Select", std_enum[2]), -AC97_ENUM("Mic Select", std_enum[3]), +AC97_ENUM("Mic Select Capture Switch", std_enum[3]), AC97_SINGLE("ADC/DAC Loopback", AC97_GENERAL_PURPOSE, 7, 1, 0) }; @@ -1226,7 +1236,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) ac97->regs[AC97_CENTER_LFE_MASTER] = 0x8080; /* build center controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) { + if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) + && !(ac97->flags & AC97_AD_MULTI)) { if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_center[0], ac97))) < 0) return err; if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_center[1], ac97))) < 0) @@ -1238,7 +1249,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build LFE controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER+1)) { + if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER+1)) + && !(ac97->flags & AC97_AD_MULTI)) { if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_lfe[0], ac97))) < 0) return err; if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_lfe[1], ac97))) < 0) @@ -1250,7 +1262,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build surround controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER)) { + if ((snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER)) + && !(ac97->flags & AC97_AD_MULTI)) { /* Surround Master (0x38) is with stereo mutes */ if ((err = snd_ac97_cmix_new_stereo(card, "Surround Playback", AC97_SURROUND_MASTER, 1, ac97)) < 0) return err; @@ -1335,9 +1348,11 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build Aux controls */ - if (snd_ac97_try_volume_mix(ac97, AC97_AUX)) { - if ((err = snd_ac97_cmix_new(card, "Aux Playback", AC97_AUX, ac97)) < 0) - return err; + if (!(ac97->flags & AC97_HAS_NO_AUX)) { + if (snd_ac97_try_volume_mix(ac97, AC97_AUX)) { + if ((err = snd_ac97_cmix_new(card, "Aux Playback", AC97_AUX, ac97)) < 0) + return err; + } } /* build PCM controls */ @@ -1682,6 +1697,7 @@ const char *snd_ac97_get_short_name(struct snd_ac97 *ac97) return "unknown codec"; } +EXPORT_SYMBOL(snd_ac97_get_short_name); /* wait for a while until registers are accessible after RESET * return 0 if ok, negative not ready @@ -1774,6 +1790,8 @@ int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops, return 0; } +EXPORT_SYMBOL(snd_ac97_bus); + /* stop no dev release warning */ static void ac97_device_release(struct device * dev) { @@ -2117,6 +2135,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, return 0; } +EXPORT_SYMBOL(snd_ac97_mixer); /* * Power down the chip. @@ -2166,6 +2185,8 @@ void snd_ac97_suspend(struct snd_ac97 *ac97) snd_ac97_powerdown(ac97); } +EXPORT_SYMBOL(snd_ac97_suspend); + /* * restore ac97 status */ @@ -2267,6 +2288,8 @@ __reset_ready: snd_ac97_restore_iec958(ac97); } } + +EXPORT_SYMBOL(snd_ac97_resume); #endif @@ -2590,29 +2613,7 @@ int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, cons return 0; } - -/* - * Exported symbols - */ - -EXPORT_SYMBOL(snd_ac97_write); -EXPORT_SYMBOL(snd_ac97_read); -EXPORT_SYMBOL(snd_ac97_write_cache); -EXPORT_SYMBOL(snd_ac97_update); -EXPORT_SYMBOL(snd_ac97_update_bits); -EXPORT_SYMBOL(snd_ac97_get_short_name); -EXPORT_SYMBOL(snd_ac97_bus); -EXPORT_SYMBOL(snd_ac97_mixer); -EXPORT_SYMBOL(snd_ac97_pcm_assign); -EXPORT_SYMBOL(snd_ac97_pcm_open); -EXPORT_SYMBOL(snd_ac97_pcm_close); -EXPORT_SYMBOL(snd_ac97_pcm_double_rate_rules); EXPORT_SYMBOL(snd_ac97_tune_hardware); -EXPORT_SYMBOL(snd_ac97_set_rate); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_ac97_resume); -EXPORT_SYMBOL(snd_ac97_suspend); -#endif /* * INIT part diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 4d9cf37300f..7f197c78081 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -464,6 +464,10 @@ int patch_wolfson05(struct snd_ac97 * ac97) { /* WM9705, WM9710 */ ac97->build_ops = &patch_wolfson_wm9705_ops; +#ifdef CONFIG_TOUCHSCREEN_WM9705 + /* WM9705 touchscreen uses AUX and VIDEO for touch */ + ac97->flags |=3D AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX; +#endif return 0; } @@ -1367,6 +1371,13 @@ static void ad18xx_resume(struct snd_ac97 *ac97) snd_ac97_restore_iec958(ac97); } + +static void ad1888_resume(struct snd_ac97 *ac97) +{ + ad18xx_resume(ac97); + snd_ac97_write_cache(ac97, AC97_CODEC_CLASS_REV, 0x8080); +} + #endif int patch_ad1819(struct snd_ac97 * ac97) @@ -1627,6 +1638,7 @@ static const struct snd_kcontrol_new snd_ac97_ad1981x_jack_sense[] = { * (SS vendor << 16 | device) */ static unsigned int ad1981_jacks_blacklist[] = { + 0x10140537, /* Thinkpad T41p */ 0x10140554, /* Thinkpad T42p/R50p */ 0 /* end */ }; @@ -1839,7 +1851,7 @@ static struct snd_ac97_build_ops patch_ad1888_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1888_specific, #ifdef CONFIG_PM - .resume = ad18xx_resume, + .resume = ad1888_resume, #endif .update_jacks = ad1888_update_jacks, }; @@ -2048,7 +2060,10 @@ int patch_alc650(struct snd_ac97 * ac97) /* Enable SPDIF-IN only on Rev.E and above */ val = snd_ac97_read(ac97, AC97_ALC650_CLOCK); /* SPDIF IN with pin 47 */ - if (ac97->spec.dev_flags) + if (ac97->spec.dev_flags && + /* ASUS A6KM requires EAPD */ + ! (ac97->subsystem_vendor == 0x1043 && + ac97->subsystem_device == 0x1103)) val |= 0x03; /* enable */ else val &= ~0x03; /* disable */ diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 512a3583b0c..f684aa2c006 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -317,6 +317,8 @@ int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate) return 0; } +EXPORT_SYMBOL(snd_ac97_set_rate); + static unsigned short get_pslots(struct snd_ac97 *ac97, unsigned char *rate_table, unsigned short *spdif_slots) { if (!ac97_is_audio(ac97)) @@ -550,6 +552,8 @@ int snd_ac97_pcm_assign(struct snd_ac97_bus *bus, return 0; } +EXPORT_SYMBOL(snd_ac97_pcm_assign); + /** * snd_ac97_pcm_open - opens the given AC97 pcm * @pcm: the ac97 pcm instance @@ -633,6 +637,8 @@ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate, return err; } +EXPORT_SYMBOL(snd_ac97_pcm_open); + /** * snd_ac97_pcm_close - closes the given AC97 pcm * @pcm: the ac97 pcm instance @@ -658,6 +664,8 @@ int snd_ac97_pcm_close(struct ac97_pcm *pcm) return 0; } +EXPORT_SYMBOL(snd_ac97_pcm_close); + static int double_rate_hw_constraint_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -709,3 +717,5 @@ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime) SNDRV_PCM_HW_PARAM_RATE, -1); return err; } + +EXPORT_SYMBOL(snd_ac97_pcm_double_rate_rules); diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 4d523df79cc..2118df50b9d 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -433,7 +433,7 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) prefix = ac97_is_audio(ac97) ? "ac97" : "mc97"; sprintf(name, "%s#%d-%d", prefix, ac97->addr, ac97->num); if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) { - snd_info_set_text_ops(entry, ac97, 1024, snd_ac97_proc_read); + snd_info_set_text_ops(entry, ac97, snd_ac97_proc_read); if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -442,10 +442,9 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) ac97->proc = entry; sprintf(name, "%s#%d-%d+regs", prefix, ac97->addr, ac97->num); if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) { - snd_info_set_text_ops(entry, ac97, 1024, snd_ac97_proc_regs_read); + snd_info_set_text_ops(entry, ac97, snd_ac97_proc_regs_read); #ifdef CONFIG_SND_DEBUG entry->mode |= S_IWUSR; - entry->c.text.write_size = 1024; entry->c.text.write = snd_ac97_proc_regs_write; #endif if (snd_info_register(entry) < 0) { diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c index 0fb7b340731..94c26ec0588 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ac97/ak4531_codec.c @@ -453,7 +453,7 @@ static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak453 struct snd_info_entry *entry; if (! snd_card_proc_new(card, "ak4531", &entry)) - snd_info_set_text_ops(entry, ak4531, 1024, snd_ak4531_proc_read); + snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read); } #endif diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index eece1c7e55a..d42bf457036 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -753,7 +753,7 @@ snd_ad1889_proc_init(struct snd_ad1889 *chip) struct snd_info_entry *entry; if (!snd_card_proc_new(chip->card, chip->card->driver, &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_ad1889_proc_read); + snd_info_set_text_ops(entry, chip, snd_ad1889_proc_read); } static struct ac97_quirk ac97_quirks[] = { diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index e2dbc211890..5dfdbf6657f 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -49,7 +49,7 @@ MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci},{ALI,M5451}}"); static int index = SNDRV_DEFAULT_IDX1; /* Index */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int pcm_channels = 32; -static int spdif = 0; +static int spdif; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for ALI M5451 PCI Audio."); @@ -2173,7 +2173,7 @@ static void __devinit snd_ali_proc_init(struct snd_ali *codec) { struct snd_info_entry *entry; if(!snd_card_proc_new(codec->card, "ali5451", &entry)) - snd_info_set_text_ops(entry, codec, 1024, snd_ali_proc_read); + snd_info_set_text_ops(entry, codec, snd_ali_proc_read); } static int __devinit snd_ali_resources(struct snd_ali *codec) diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 60423b1c678..a9f08066459 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -746,8 +746,8 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, card->shortname, chip->alt_port, chip->irq); if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000, - gcr+0x30, 1, pci->irq, 0, - &chip->rmidi)) < 0) { + gcr+0x30, MPU401_INFO_INTEGRATED, + pci->irq, 0, &chip->rmidi)) < 0) { printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n", gcr+0x30); goto out_err; } diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d0f759d86d3..f18a8c0e468 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1504,7 +1504,7 @@ static void __devinit snd_atiixp_proc_init(struct atiixp *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "atiixp", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_atiixp_proc_read); + snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read); } #else /* !CONFIG_PROC_FS */ #define snd_atiixp_proc_init(chip) diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 12a34c39caa..40739057076 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1177,7 +1177,7 @@ static void __devinit snd_atiixp_proc_init(struct atiixp_modem *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "atiixp-modem", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_atiixp_proc_read); + snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read); } #else #define snd_atiixp_proc_init(chip) diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 126870ec063..8a3b118989b 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -261,6 +261,13 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return err; } snd_vortex_workaround(pci, pcifix[dev]); + + // Card details needed in snd_vortex_midi + strcpy(card->driver, CARD_NAME_SHORT); + sprintf(card->shortname, "Aureal Vortex %s", CARD_NAME_SHORT); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->io, chip->irq); + // (4) Alloc components. // ADB pcm. if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_ADB)) < 0) { @@ -323,11 +330,6 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) #endif // (5) - strcpy(card->driver, CARD_NAME_SHORT); - strcpy(card->shortname, CARD_NAME_SHORT); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->io, chip->irq); - if ((err = pci_read_config_word(pci, PCI_DEVICE_ID, &(chip->device))) < 0) { snd_card_free(card); diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 873f486b07b..c75d368ea08 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -47,7 +47,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) struct snd_rawmidi *rmidi; int temp, mode; struct snd_mpu401 *mpu; - int port; + unsigned long port; #ifdef VORTEX_MPU401_LEGACY /* EnableHardCodedMPU401Port() */ @@ -70,9 +70,6 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) temp |= (MIDI_CLOCK_DIV << 8) | ((mode >> 24) & 0xff) << 4; hwwrite(vortex->mmio, VORTEX_CTRL2, temp); hwwrite(vortex->mmio, VORTEX_MIDI_CMD, MPU401_RESET); - /* Set some kind of mode */ - if (mode) - hwwrite(vortex->mmio, VORTEX_MIDI_CMD, MPU401_ENTER_UART); /* Check if anything is OK. */ temp = hwread(vortex->mmio, VORTEX_MIDI_DATA); @@ -98,7 +95,8 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) port = (unsigned long)(vortex->mmio + VORTEX_MIDI_DATA); if ((temp = snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port, - 1, 0, 0, &rmidi)) != 0) { + MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO, + 0, 0, &rmidi)) != 0) { hwwrite(vortex->mmio, VORTEX_CTRL, (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN); @@ -107,6 +105,9 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) mpu = rmidi->private_data; mpu->cport = (unsigned long)(vortex->mmio + VORTEX_MIDI_CMD); #endif + /* Overwrite MIDI name */ + snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI %d", CARD_NAME_SHORT , vortex->card->number); + vortex->rmidi = rmidi; return 0; } diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c index 4534e1882ad..b4151e208b7 100644 --- a/sound/pci/au88x0/au88x0_xtalk.c +++ b/sound/pci/au88x0/au88x0_xtalk.c @@ -66,31 +66,20 @@ static xtalk_gains_t const asXtalkGainsAllChan = { 0 //0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff,0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff }; -static xtalk_gains_t const asXtalkGainsZeros = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 -}; +static xtalk_gains_t const asXtalkGainsZeros; -static xtalk_dline_t const alXtalkDlineZeros = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0 -}; +static xtalk_dline_t const alXtalkDlineZeros; static xtalk_dline_t const alXtalkDlineTest = { 0xFC18, 0x03E8FFFF, 0x186A0, 0x7960FFFE, 1, 0xFFFFFFFF, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }; -static xtalk_instate_t const asXtalkInStateZeros = { 0, 0, 0, 0 }; +static xtalk_instate_t const asXtalkInStateZeros; static xtalk_instate_t const asXtalkInStateTest = { 0xFF80, 0x0080, 0xFFFF, 0x0001 }; -static xtalk_state_t const asXtalkOutStateZeros = { - {0, 0, 0, 0}, - {0, 0, 0, 0}, - {0, 0, 0, 0}, - {0, 0, 0, 0}, - {0, 0, 0, 0} -}; +static xtalk_state_t const asXtalkOutStateZeros; + static short const sDiamondKLeftEq = 0x401d; static short const sDiamondKRightEq = 0x401d; static short const sDiamondKLeftXt = 0xF90E; @@ -162,13 +151,7 @@ static xtalk_coefs_t const asXtalkNarrowCoefsRightXt = { {0, 0, 0, 0, 0} }; -static xtalk_coefs_t const asXtalkCoefsZeros = { - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0}, - {0, 0, 0, 0, 0} -}; +static xtalk_coefs_t const asXtalkCoefsZeros; static xtalk_coefs_t const asXtalkCoefsPipe = { {0, 0, 0x0FA0, 0, 0}, {0, 0, 0x0FA0, 0, 0}, diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 52a36452426..6e62dafb66c 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -33,14 +33,21 @@ * in the first place >:-P}), * I was forced to base this driver on reverse engineering * (3 weeks' worth of evenings filled with driver work). - * (and no, I did NOT go the easy way: to pick up a PCI128 for 9 Euros) + * (and no, I did NOT go the easy way: to pick up a SB PCI128 for 9 Euros) * * The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name * for compatibility reasons) has the following features: * * - builtin AC97 conformant codec (SNR over 80dB) - * (really AC97 compliant?? I really doubt it when looking - * at the mixer register layout) + * Note that "conformant" != "compliant"!! this chip's mixer register layout + * *differs* from the standard AC97 layout: + * they chose to not implement the headphone register (which is not a + * problem since it's merely optional), yet when doing this, they committed + * the grave sin of letting other registers follow immediately instead of + * keeping a headphone dummy register, thereby shifting the mixer register + * addresses illegally. So far unfortunately it looks like the very flexible + * ALSA AC97 support is still not enough to easily compensate for such a + * grave layout violation despite all tweaks and quirks mechanisms it offers. * - builtin genuine OPL3 * - full duplex 16bit playback/record at independent sampling rate * - MPU401 (+ legacy address support) FIXME: how to enable legacy addr?? @@ -90,10 +97,15 @@ * * TODO * - test MPU401 MIDI playback etc. - * - power management. See e.g. intel8x0 or cs4281. - * This would be nice since the chip runs a bit hot, and it's *required* - * anyway for proper ACPI power management. + * - add some power micro-management (disable various units of the card + * as long as they're unused). However this requires I/O ports which I + * haven't figured out yet and which thus might not even exist... + * The standard suspend/resume functionality could probably make use of + * some improvement, too... * - figure out what all unknown port bits are responsible for + * - figure out some cleverly evil scheme to possibly make ALSA AC97 code + * fully accept our quite incompatible ""AC97"" mixer and thus save some + * code (but I'm not too optimistic that doing this is possible at all) */ #include <sound/driver.h> @@ -214,6 +226,16 @@ struct snd_azf3328 { struct pci_dev *pci; int irq; + +#ifdef CONFIG_PM + /* register value containers for power management + * Note: not always full I/O range preserved (just like Win driver!) */ + u16 saved_regs_codec [AZF_IO_SIZE_CODEC_PM / 2]; + u16 saved_regs_io2 [AZF_IO_SIZE_IO2_PM / 2]; + u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; + u16 saved_regs_synth[AZF_IO_SIZE_SYNTH_PM / 2]; + u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2]; +#endif }; static const struct pci_device_id snd_azf3328_ids[] __devinitdata = { @@ -317,10 +339,8 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg else dst_vol_left &= ~0x80; - do - { - if (!left_done) - { + do { + if (!left_done) { if (curr_vol_left > dst_vol_left) curr_vol_left--; else @@ -330,8 +350,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg left_done = 1; outb(curr_vol_left, portbase + 1); } - if (!right_done) - { + if (!right_done) { if (curr_vol_right > dst_vol_right) curr_vol_right--; else @@ -346,8 +365,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg } if (delay) mdelay(delay); - } - while ((!left_done) || (!right_done)); + } while ((!left_done) || (!right_done)); snd_azf3328_dbgcallleave(); } @@ -514,15 +532,18 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static const char * const texts1[] = { - "ModemOut1", "ModemOut2" + "Mic1", "Mic2" }; static const char * const texts2[] = { - "MonoSelectSource1", "MonoSelectSource2" + "Mix", "Mic" }; static const char * const texts3[] = { "Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone" }; + static const char * const texts4[] = { + "pre 3D", "post 3D" + }; struct azf3328_mixer_reg reg; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); @@ -531,14 +552,19 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, uinfo->value.enumerated.items = reg.enum_c; if (uinfo->value.enumerated.item > reg.enum_c - 1U) uinfo->value.enumerated.item = reg.enum_c - 1U; - if (reg.reg == IDX_MIXER_ADVCTL2) - { - if (reg.lchan_shift == 8) /* modem out sel */ + if (reg.reg == IDX_MIXER_ADVCTL2) { + switch(reg.lchan_shift) { + case 8: /* modem out sel */ strcpy(uinfo->value.enumerated.name, texts1[uinfo->value.enumerated.item]); - else /* mono sel source */ + break; + case 9: /* mono sel source */ strcpy(uinfo->value.enumerated.name, texts2[uinfo->value.enumerated.item]); - } - else + break; + case 15: /* PCM Out Path */ + strcpy(uinfo->value.enumerated.name, texts4[uinfo->value.enumerated.item]); + break; + } + } else strcpy(uinfo->value.enumerated.name, texts3[uinfo->value.enumerated.item] ); return 0; @@ -554,12 +580,10 @@ snd_azf3328_get_mixer_enum(struct snd_kcontrol *kcontrol, snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); val = snd_azf3328_mixer_inw(chip, reg.reg); - if (reg.reg == IDX_MIXER_REC_SELECT) - { + if (reg.reg == IDX_MIXER_REC_SELECT) { ucontrol->value.enumerated.item[0] = (val >> 8) & (reg.enum_c - 1); ucontrol->value.enumerated.item[1] = (val >> 0) & (reg.enum_c - 1); - } - else + } else ucontrol->value.enumerated.item[0] = (val >> reg.lchan_shift) & (reg.enum_c - 1); snd_azf3328_dbgmixer("get_enum: %02x is %04x -> %d|%d (shift %02d, enum_c %d)\n", @@ -579,16 +603,13 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); oreg = snd_azf3328_mixer_inw(chip, reg.reg); val = oreg; - if (reg.reg == IDX_MIXER_REC_SELECT) - { + if (reg.reg == IDX_MIXER_REC_SELECT) { if (ucontrol->value.enumerated.item[0] > reg.enum_c - 1U || ucontrol->value.enumerated.item[1] > reg.enum_c - 1U) return -EINVAL; val = (ucontrol->value.enumerated.item[0] << 8) | (ucontrol->value.enumerated.item[1] << 0); - } - else - { + } else { if (ucontrol->value.enumerated.item[0] > reg.enum_c - 1U) return -EINVAL; val &= ~((reg.enum_c - 1) << reg.lchan_shift); @@ -629,13 +650,14 @@ static const struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata AZF3328_MIXER_VOL_MONO("Modem Playback Volume", IDX_MIXER_MODEMOUT, 0x1f, 1), AZF3328_MIXER_SWITCH("Modem Capture Switch", IDX_MIXER_MODEMIN, 15, 1), AZF3328_MIXER_VOL_MONO("Modem Capture Volume", IDX_MIXER_MODEMIN, 0x1f, 1), - AZF3328_MIXER_ENUM("Modem Out Select", IDX_MIXER_ADVCTL2, 2, 8), - AZF3328_MIXER_ENUM("Mono Select Source", IDX_MIXER_ADVCTL2, 2, 9), + AZF3328_MIXER_ENUM("Mic Select", IDX_MIXER_ADVCTL2, 2, 8), + AZF3328_MIXER_ENUM("Mono Output Select", IDX_MIXER_ADVCTL2, 2, 9), + AZF3328_MIXER_ENUM("PCM", IDX_MIXER_ADVCTL2, 2, 15), /* PCM Out Path, place in front since it controls *both* 3D and Bass/Treble! */ AZF3328_MIXER_VOL_SPECIAL("Tone Control - Treble", IDX_MIXER_BASSTREBLE, 0x07, 1, 0), AZF3328_MIXER_VOL_SPECIAL("Tone Control - Bass", IDX_MIXER_BASSTREBLE, 0x07, 9, 0), AZF3328_MIXER_SWITCH("3D Control - Switch", IDX_MIXER_ADVCTL2, 13, 0), - AZF3328_MIXER_VOL_SPECIAL("3D Control - Wide", IDX_MIXER_ADVCTL1, 0x07, 1, 0), /* "3D Width" */ - AZF3328_MIXER_VOL_SPECIAL("3D Control - Space", IDX_MIXER_ADVCTL1, 0x03, 8, 0), /* "Hifi 3D" */ + AZF3328_MIXER_VOL_SPECIAL("3D Control - Width", IDX_MIXER_ADVCTL1, 0x07, 1, 0), /* "3D Width" */ + AZF3328_MIXER_VOL_SPECIAL("3D Control - Depth", IDX_MIXER_ADVCTL1, 0x03, 8, 0), /* "Hifi 3D" */ #if MIXER_TESTING AZF3328_MIXER_SWITCH("0", IDX_MIXER_ADVCTL2, 0, 0), AZF3328_MIXER_SWITCH("1", IDX_MIXER_ADVCTL2, 1, 0), @@ -813,22 +835,18 @@ snd_azf3328_setdmaa(struct snd_azf3328 *chip, unsigned int is_running; snd_azf3328_dbgcallenter(); - if (do_recording) - { + if (do_recording) { /* access capture registers, i.e. skip playback reg section */ portbase = chip->codec_port + 0x20; is_running = chip->is_recording; - } - else - { + } else { /* access the playback register section */ portbase = chip->codec_port + 0x00; is_running = chip->is_playing; } /* AZF3328 uses a two buffer pointer DMA playback approach */ - if (!is_running) - { + if (!is_running) { unsigned long addr_area2; unsigned long count_areas, count_tmp; /* width 32bit -- overflow!! */ count_areas = size/2; @@ -961,6 +979,13 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) chip->is_playing = 1; snd_azf3328_dbgplay("STARTED PLAYBACK\n"); break; + case SNDRV_PCM_TRIGGER_RESUME: + snd_azf3328_dbgplay("RESUME PLAYBACK\n"); + /* resume playback if we were active */ + if (chip->is_playing) + snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME); + break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP PLAYBACK\n"); @@ -988,6 +1013,12 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) chip->is_playing = 0; snd_azf3328_dbgplay("STOPPED PLAYBACK\n"); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + snd_azf3328_dbgplay("SUSPEND PLAYBACK\n"); + /* make sure playback is stopped */ + snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) & ~DMA_RESUME); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n"); break; @@ -995,6 +1026,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n"); break; default: + printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } @@ -1068,6 +1100,13 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) chip->is_recording = 1; snd_azf3328_dbgplay("STARTED CAPTURE\n"); break; + case SNDRV_PCM_TRIGGER_RESUME: + snd_azf3328_dbgplay("RESUME CAPTURE\n"); + /* resume recording if we were active */ + if (chip->is_recording) + snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, + snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME); + break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP CAPTURE\n"); @@ -1088,6 +1127,12 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) chip->is_recording = 0; snd_azf3328_dbgplay("STOPPED CAPTURE\n"); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + snd_azf3328_dbgplay("SUSPEND CAPTURE\n"); + /* make sure recording is stopped */ + snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, + snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) & ~DMA_RESUME); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n"); break; @@ -1095,6 +1140,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n"); break; default: + printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } @@ -1163,8 +1209,7 @@ snd_azf3328_interrupt(int irq, void *dev_id, struct pt_regs *regs) snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), status); - if (status & IRQ_TIMER) - { + if (status & IRQ_TIMER) { /* snd_azf3328_dbgplay("timer %ld\n", inl(chip->codec_port+IDX_IO_TIMER_VALUE) & TIMER_VALUE_MASK); */ if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); @@ -1174,50 +1219,43 @@ snd_azf3328_interrupt(int irq, void *dev_id, struct pt_regs *regs) spin_unlock(&chip->reg_lock); snd_azf3328_dbgplay("azt3328: timer IRQ\n"); } - if (status & IRQ_PLAYBACK) - { + if (status & IRQ_PLAYBACK) { spin_lock(&chip->reg_lock); which = snd_azf3328_codec_inb(chip, IDX_IO_PLAY_IRQTYPE); /* ack all IRQ types immediately */ snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->playback_substream) - { + if (chip->pcm && chip->playback_substream) { snd_pcm_period_elapsed(chip->playback_substream); snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n", which, inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS)); - } - else + } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) snd_azf3328_dbgplay("azt3328: unknown play IRQ type occurred, please report!\n"); } - if (status & IRQ_RECORDING) - { + if (status & IRQ_RECORDING) { spin_lock(&chip->reg_lock); which = snd_azf3328_codec_inb(chip, IDX_IO_REC_IRQTYPE); /* ack all IRQ types immediately */ snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->capture_substream) - { + if (chip->pcm && chip->capture_substream) { snd_pcm_period_elapsed(chip->capture_substream); snd_azf3328_dbgplay("REC period done (#%x), @ %x\n", which, inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS)); - } - else + } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); } /* MPU401 has less critical IRQ requirements * than timer and playback/recording, right? */ - if (status & IRQ_MPU401) - { + if (status & IRQ_MPU401) { snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data, regs); /* hmm, do we have to ack the IRQ here somehow? @@ -1511,8 +1549,7 @@ snd_azf3328_timer_start(struct snd_timer *timer) snd_azf3328_dbgcallenter(); chip = snd_timer_chip(timer); delay = ((timer->sticks * seqtimer_scaling) - 1) & TIMER_VALUE_MASK; - if (delay < 49) - { + if (delay < 49) { /* uhoh, that's not good, since user-space won't know about * this timing tweak * (we need to do it to avoid a lockup, though) */ @@ -1766,9 +1803,11 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) goto out_err; } + card->private_data = chip; + if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_MPU401, - chip->mpu_port, 1, pci->irq, 0, - &chip->rmidi)) < 0) { + chip->mpu_port, MPU401_INFO_INTEGRATED, + pci->irq, 0, &chip->rmidi)) < 0) { snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", chip->mpu_port); goto out_err; } @@ -1791,6 +1830,8 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) } } + opl3->private_data = chip; + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->codec_port, chip->irq); @@ -1834,11 +1875,80 @@ snd_azf3328_remove(struct pci_dev *pci) snd_azf3328_dbgcallleave(); } +#ifdef CONFIG_PM +static int +snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_azf3328 *chip = card->private_data; + int reg; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + snd_pcm_suspend_all(chip->pcm); + + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) + chip->saved_regs_mixer[reg] = inw(chip->mixer_port + reg * 2); + + /* make sure to disable master volume etc. to prevent looping sound */ + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) + chip->saved_regs_codec[reg] = inw(chip->codec_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) + chip->saved_regs_io2[reg] = inw(chip->io2_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) + chip->saved_regs_mpu[reg] = inw(chip->mpu_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) + chip->saved_regs_synth[reg] = inw(chip->synth_port + reg * 2); + + pci_set_power_state(pci, PCI_D3hot); + pci_disable_device(pci); + pci_save_state(pci); + return 0; +} + +static int +snd_azf3328_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_azf3328 *chip = card->private_data; + int reg; + + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_power_state(pci, PCI_D0); + pci_set_master(pci); + + for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) + outw(chip->saved_regs_io2[reg], chip->io2_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) + outw(chip->saved_regs_mpu[reg], chip->mpu_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) + outw(chip->saved_regs_synth[reg], chip->synth_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) + outw(chip->saved_regs_mixer[reg], chip->mixer_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) + outw(chip->saved_regs_codec[reg], chip->codec_port + reg * 2); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + + + + static struct pci_driver driver = { .name = "AZF3328", .id_table = snd_azf3328_ids, .probe = snd_azf3328_probe, .remove = __devexit_p(snd_azf3328_remove), +#ifdef CONFIG_PM + .suspend = snd_azf3328_suspend, + .resume = snd_azf3328_resume, +#endif }; static int __init diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index f489bdaf6d4..b4f3e3cd006 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -5,6 +5,9 @@ /*** main I/O area port indices ***/ /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */ +#define AZF_IO_SIZE_CODEC 0x80 +#define AZF_IO_SIZE_CODEC_PM 0x70 + /* the driver initialisation suggests a layout of 4 main areas: * from 0x00 (playback), from 0x20 (recording) and from 0x40 (maybe MPU401??). * And another area from 0x60 to 0x6f (DirectX timer, IRQ management, @@ -87,7 +90,7 @@ #define IDX_IO_REC_DMA_CURROFS 0x34 /* PU:0x00000000 */ #define IDX_IO_REC_SOUNDFORMAT 0x36 /* PU:0x0000 */ -/** hmm, what is this I/O area for? MPU401?? (after playback, recording, ???, timer) **/ +/** hmm, what is this I/O area for? MPU401?? or external DAC via I2S?? (after playback, recording, ???, timer) **/ #define IDX_IO_SOMETHING_FLAGS 0x40 /* gets set to 0x34 just like port 0x0 and 0x20 on card init, PU:0x0000 */ /* general */ #define IDX_IO_42H 0x42 /* PU:0x0001 */ @@ -107,7 +110,8 @@ #define IRQ_UNKNOWN2 0x0080 /* probably unused */ #define IDX_IO_66H 0x66 /* writing 0xffff returns 0x0000 */ #define IDX_IO_SOME_VALUE 0x68 /* this is set to e.g. 0x3ff or 0x300, and writable; maybe some buffer limit, but I couldn't find out more, PU:0x00ff */ -#define IDX_IO_6AH 0x6A /* this WORD can be set to have bits 0x0028 activated; actually inhibits PCM playback!!! maybe power management?? */ +#define IDX_IO_6AH 0x6A /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); actually inhibits PCM playback!!! maybe power management?? */ + #define IO_6A_PAUSE_PLAYBACK 0x0200 /* bit 9; sure, this pauses playback, but what the heck is this really about?? */ #define IDX_IO_6CH 0x6C #define IDX_IO_6EH 0x6E /* writing 0xffff returns 0x83fe */ /* further I/O indices not saved/restored, so probably not used */ @@ -115,15 +119,25 @@ /*** I/O 2 area port indices ***/ /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ +#define AZF_IO_SIZE_IO2 0x08 +#define AZF_IO_SIZE_IO2_PM 0x06 + #define IDX_IO2_LEGACY_ADDR 0x04 #define LEGACY_SOMETHING 0x01 /* OPL3?? */ #define LEGACY_JOY 0x08 +#define AZF_IO_SIZE_MPU 0x04 +#define AZF_IO_SIZE_MPU_PM 0x04 + +#define AZF_IO_SIZE_SYNTH 0x08 +#define AZF_IO_SIZE_SYNTH_PM 0x06 /*** mixer I/O area port indices ***/ /* (only 0x22 of 0x40 bytes saved/restored by Windows driver) - * generally spoken: AC97 register index = AZF3328 mixer reg index + 2 - * (in other words: AZF3328 NOT fully AC97 compliant) */ + * UNFORTUNATELY azf3328 is NOT truly AC97 compliant: see main file intro */ +#define AZF_IO_SIZE_MIXER 0x40 +#define AZF_IO_SIZE_MIXER_PM 0x22 + #define MIXER_VOLUME_RIGHT_MASK 0x001f #define MIXER_VOLUME_LEFT_MASK 0x1f00 #define MIXER_MUTE_MASK 0x8000 @@ -156,14 +170,14 @@ #define IDX_MIXER_ADVCTL1 0x1e /* unlisted bits are unmodifiable */ #define MIXER_ADVCTL1_3DWIDTH_MASK 0x000e - #define MIXER_ADVCTL1_HIFI3D_MASK 0x0300 -#define IDX_MIXER_ADVCTL2 0x20 /* resembles AC97_GENERAL_PURPOSE reg! */ + #define MIXER_ADVCTL1_HIFI3D_MASK 0x0300 /* yup, this is missing the high bit that official AC97 contains, plus it doesn't have linear bit value range behaviour but instead acts weirdly (possibly we're dealing with two *different* 3D settings here??) */ +#define IDX_MIXER_ADVCTL2 0x20 /* subset of AC97_GENERAL_PURPOSE reg! */ /* unlisted bits are unmodifiable */ - #define MIXER_ADVCTL2_BIT7 0x0080 /* WaveOut 3D Bypass? mutes WaveOut at LineOut */ - #define MIXER_ADVCTL2_BIT8 0x0100 /* is this Modem Out Select? */ - #define MIXER_ADVCTL2_BIT9 0x0200 /* Mono Select Source? */ - #define MIXER_ADVCTL2_BIT13 0x2000 /* 3D enable? */ - #define MIXER_ADVCTL2_BIT15 0x8000 /* unknown */ + #define MIXER_ADVCTL2_LPBK 0x0080 /* Loopback mode -- Win driver: "WaveOut3DBypass"? mutes WaveOut at LineOut */ + #define MIXER_ADVCTL2_MS 0x0100 /* Mic Select 0=Mic1, 1=Mic2 -- Win driver: "ModemOutSelect"?? */ + #define MIXER_ADVCTL2_MIX 0x0200 /* Mono output select 0=Mix, 1=Mic; Win driver: "MonoSelectSource"?? */ + #define MIXER_ADVCTL2_3D 0x2000 /* 3D Enhancement 1=on */ + #define MIXER_ADVCTL2_POP 0x8000 /* Pcm Out Path, 0=pre 3D, 1=post 3D */ #define IDX_MIXER_SOMETHING30H 0x30 /* used, but unknown??? */ diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 9ee07d4aac1..c33642d8d9a 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -44,7 +44,7 @@ MODULE_SUPPORTED_DEVICE("{{Brooktree,Bt878}," static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* Exclude the first card */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int digital_rate[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; /* digital input rate */ +static int digital_rate[SNDRV_CARDS]; /* digital input rate */ static int load_all; /* allow to load the non-whitelisted cards */ module_param_array(index, int, NULL, 0444); @@ -781,10 +781,12 @@ static struct pci_device_id snd_bt87x_ids[] __devinitdata = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, 32000), /* Viewcast Osprey 200 */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, 44100), - /* AVerMedia Studio No. 103, 203, ...? */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000), /* Leadtek Winfast tv 2000xp delux */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, 32000), + /* Voodoo TV 200 */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, 32000), + /* AVerMedia Studio No. 103, 203, ...? */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000), { } }; MODULE_DEVICE_TABLE(pci, snd_bt87x_ids); diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index c8131ea92ed..9cb66c59f52 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -537,9 +537,9 @@ #endif #define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux +#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used) #define ADC_MUX_MIC 0x00000002 //Value to select Mic at ADC Mux #define ADC_MUX_LINEIN 0x00000004 //Value to select LineIn at ADC Mux -#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used) #define ADC_MUX_AUX 0x00000008 //Value to select Aux at ADC Mux #define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */ @@ -604,6 +604,8 @@ struct snd_ca0106 { u32 spdif_bits[4]; /* s/pdif out setup */ int spdif_enable; int capture_source; + int i2c_capture_source; + u8 i2c_capture_volume[4][2]; int capture_mic_line_in; struct snd_dma_buffer buffer; diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index fd8bfebfbd5..59bf9bd0253 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -186,8 +186,8 @@ static struct snd_ca0106_details ca0106_chip_details[] = { /* New Audigy SE. Has a different DAC. */ /* SB0570: * CTRL:CA0106-DAT - * ADC: WM8768GEDS - * DAC: WM8775EDS + * ADC: WM8775EDS + * DAC: WM8768GEDS */ { .serial = 0x100a1102, .name = "Audigy SE [SB0570]", @@ -195,9 +195,14 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .i2c_adc = 1, .spi_dac = 1 } , /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */ + /* SB0438 + * CTRL:CA0106-DAT + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ { .serial = 0x10091462, .name = "MSI K8N Diamond MB [SB0438]", - .gpio_type = 1, + .gpio_type = 2, .i2c_adc = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX @@ -326,6 +331,7 @@ int snd_ca0106_spi_write(struct snd_ca0106 * emu, return 0; } +/* The ADC does not support i2c read, so only write is implemented */ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value) @@ -340,6 +346,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, } tmp = reg << 25 | value << 16; + // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); /* Not sure what this I2C channel controls. */ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */ @@ -348,8 +355,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, for (retry = 0; retry < 10; retry++) { /* Send the data to i2c */ - tmp = snd_ca0106_ptr_read(emu, I2C_A, 0); - tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK); + //tmp = snd_ca0106_ptr_read(emu, I2C_A, 0); + //tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK); + tmp = 0; tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD); snd_ca0106_ptr_write(emu, I2C_A, 0, tmp); @@ -1181,7 +1189,7 @@ static unsigned int spi_dac_init[] = { 0x02ff, 0x0400, 0x0520, - 0x0600, + 0x0620, /* Set 24 bit. Was 0x0600 */ 0x08ff, 0x0aff, 0x0cff, @@ -1200,6 +1208,22 @@ static unsigned int spi_dac_init[] = { 0x1400, }; +static unsigned int i2c_adc_init[][2] = { + { 0x17, 0x00 }, /* Reset */ + { 0x07, 0x00 }, /* Timeout */ + { 0x0b, 0x22 }, /* Interface control */ + { 0x0c, 0x22 }, /* Master mode control */ + { 0x0d, 0x08 }, /* Powerdown control */ + { 0x0e, 0xcf }, /* Attenuation Left 0x01 = -103dB, 0xff = 24dB */ + { 0x0f, 0xcf }, /* Attenuation Right 0.5dB steps */ + { 0x10, 0x7b }, /* ALC Control 1 */ + { 0x11, 0x00 }, /* ALC Control 2 */ + { 0x12, 0x32 }, /* ALC Control 3 */ + { 0x13, 0x00 }, /* Noise gate control */ + { 0x14, 0xa6 }, /* Limiter control */ + { 0x15, ADC_MUX_LINEIN }, /* ADC Mixer control */ +}; + static int __devinit snd_ca0106_create(struct snd_card *card, struct pci_dev *pci, struct snd_ca0106 **rchip) @@ -1361,7 +1385,12 @@ static int __devinit snd_ca0106_create(struct snd_card *card, snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */ chip->capture_source = 3; /* Set CAPTURE_SOURCE */ - if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */ + if (chip->details->gpio_type == 2) { /* The SB0438 use GPIO differently. */ + /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ + outl(0x0, chip->port+GPIO); + //outl(0x00f0e000, chip->port+GPIO); /* Analog */ + outl(0x005f5301, chip->port+GPIO); /* Analog */ + } else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */ /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ outl(0x0, chip->port+GPIO); //outl(0x00f0e000, chip->port+GPIO); /* Analog */ @@ -1379,7 +1408,19 @@ static int __devinit snd_ca0106_create(struct snd_card *card, outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */ if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */ - snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */ + int size, n; + + size = ARRAY_SIZE(i2c_adc_init); + //snd_printk("I2C:array size=0x%x\n", size); + for (n=0; n < size; n++) { + snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]); + } + for (n=0; n < 4; n++) { + chip->i2c_capture_volume[n][0]= 0xcf; + chip->i2c_capture_volume[n][1]= 0xcf; + } + chip->i2c_capture_source=2; /* Line in */ + //snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */ } if (chip->details->spi_dac == 1) { /* The SB0570 use SPI to control DAC. */ int size, n; diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 06fe055674f..146eed70dce 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -171,6 +171,76 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol, return change; } +static int snd_ca0106_i2c_capture_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[6] = { + "Phone", "Mic", "Line in", "Aux" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_ca0106_i2c_capture_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->i2c_capture_source; + return 0; +} + +static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int source_id; + unsigned int ngain, ogain; + int change = 0; + u32 source; + /* If the capture source has changed, + * update the capture volume from the cached value + * for the particular source. + */ + source_id = ucontrol->value.enumerated.item[0] ; + change = (emu->i2c_capture_source != source_id); + if (change) { + snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ + ngain = emu->i2c_capture_volume[source_id][0]; /* Left */ + ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */ + if (ngain != ogain) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff)); + ngain = emu->i2c_capture_volume[source_id][1]; /* Left */ + ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Left */ + if (ngain != ogain) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff)); + source = 1 << source_id; + snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */ + emu->i2c_capture_source = source_id; + } + return change; +} + +static int snd_ca0106_capture_line_in_side_out_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Side out", "Line in" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item > 1) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -207,16 +277,16 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, if (change) { emu->capture_mic_line_in = val; if (val) { - snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_PHONE); /* Mute input */ + //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ tmp = inl(emu->port+GPIO) & ~0x400; tmp = tmp | 0x400; outl(tmp, emu->port+GPIO); - snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); + //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); } else { - snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_PHONE); /* Mute input */ + //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ tmp = inl(emu->port+GPIO) & ~0x400; outl(tmp, emu->port+GPIO); - snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); + //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); } } return change; @@ -225,12 +295,22 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Mic/Line in Capture", + .name = "Shared Mic/Line in Capture Switch", .info = snd_ca0106_capture_mic_line_in_info, .get = snd_ca0106_capture_mic_line_in_get, .put = snd_ca0106_capture_mic_line_in_put }; +static struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out __devinitdata = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Shared Line in/Side out Capture Switch", + .info = snd_ca0106_capture_line_in_side_out_info, + .get = snd_ca0106_capture_mic_line_in_get, + .put = snd_ca0106_capture_mic_line_in_put +}; + + static int snd_ca0106_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -329,15 +409,81 @@ static int snd_ca0106_volume_put(struct snd_kcontrol *kcontrol, return 1; } +static int snd_ca0106_i2c_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 255; + return 0; +} + +static int snd_ca0106_i2c_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + int source_id; + + source_id = kcontrol->private_value; + + ucontrol->value.integer.value[0] = emu->i2c_capture_volume[source_id][0]; + ucontrol->value.integer.value[1] = emu->i2c_capture_volume[source_id][1]; + return 0; +} + +static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int ogain; + unsigned int ngain; + int source_id; + int change = 0; + + source_id = kcontrol->private_value; + ogain = emu->i2c_capture_volume[source_id][0]; /* Left */ + ngain = ucontrol->value.integer.value[0]; + if (ngain > 0xff) + return 0; + if (ogain != ngain) { + if (emu->i2c_capture_source == source_id) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) ); + emu->i2c_capture_volume[source_id][0] = ucontrol->value.integer.value[0]; + change = 1; + } + ogain = emu->i2c_capture_volume[source_id][1]; /* Right */ + ngain = ucontrol->value.integer.value[1]; + if (ngain > 0xff) + return 0; + if (ogain != ngain) { + if (emu->i2c_capture_source == source_id) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff)); + emu->i2c_capture_volume[source_id][1] = ucontrol->value.integer.value[1]; + change = 1; + } + + return change; +} + #define CA_VOLUME(xname,chid,reg) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_ca0106_volume_info, \ - .get = snd_ca0106_volume_get, \ - .put = snd_ca0106_volume_put, \ + .info = snd_ca0106_volume_info, \ + .get = snd_ca0106_volume_get, \ + .put = snd_ca0106_volume_put, \ .private_value = ((chid) << 8) | (reg) \ } +#define I2C_VOLUME(xname,chid) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_ca0106_i2c_volume_info, \ + .get = snd_ca0106_i2c_volume_get, \ + .put = snd_ca0106_i2c_volume_put, \ + .private_value = chid \ +} + static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = { CA_VOLUME("Analog Front Playback Volume", @@ -361,6 +507,11 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = { CA_VOLUME("CAPTURE feedback Playback Volume", 1, CAPTURE_CONTROL), + I2C_VOLUME("Phone Capture Volume", 0), + I2C_VOLUME("Mic Capture Volume", 1), + I2C_VOLUME("Line in Capture Volume", 2), + I2C_VOLUME("Aux Capture Volume", 3), + { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -378,12 +529,19 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", + .name = "Digital Capture Source", .info = snd_ca0106_capture_source_info, .get = snd_ca0106_capture_source_get, .put = snd_ca0106_capture_source_put }, { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_ca0106_i2c_capture_source_info, + .get = snd_ca0106_i2c_capture_source_get, + .put = snd_ca0106_i2c_capture_source_put + }, + { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .count = 4, @@ -477,7 +635,10 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) return err; } if (emu->details->i2c_adc == 1) { - err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu)); + if (emu->details->gpio_type == 1) + err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu)); + else /* gpio_type == 2 */ + err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_line_in_side_out, emu)); if (err < 0) return err; } diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 63757273bfb..75ca421eb3a 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -431,33 +431,30 @@ int __devinit snd_ca0106_proc_init(struct snd_ca0106 * emu) struct snd_info_entry *entry; if(! snd_card_proc_new(emu->card, "iec958", &entry)) - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_iec958); + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_iec958); if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read32); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32); entry->c.text.write = snd_ca0106_proc_reg_write32; entry->mode |= S_IWUSR; } if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read16); + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16); if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry)) - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read8); + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read8); if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read1); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1); entry->c.text.write = snd_ca0106_proc_reg_write; entry->mode |= S_IWUSR; // entry->private_data = emu; } if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_i2c_write); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_i2c_write); entry->c.text.write = snd_ca0106_proc_i2c_write; entry->mode |= S_IWUSR; // entry->private_data = emu; } if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) - snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read2); + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2); return 0; } diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index e5ce2dabd08..0938c158b5c 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2121,7 +2121,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = { 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("PC Speaker Playback 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), }; @@ -2602,7 +2602,7 @@ static void __devinit snd_cmipci_proc_init(struct cmipci *cm) struct snd_info_entry *entry; if (! snd_card_proc_new(cm->card, "cmipci", &entry)) - snd_info_set_text_ops(entry, cm, 1024, snd_cmipci_proc_read); + snd_info_set_text_ops(entry, cm, snd_cmipci_proc_read); } #else /* !CONFIG_PROC_FS */ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} @@ -2932,7 +2932,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc } integrated_midi = snd_cmipci_read_b(cm, CM_REG_MPU_PCI) != 0xff; - if (integrated_midi) + if (integrated_midi && mpu_port[dev] == 1) iomidi = cm->iobase + CM_REG_MPU_PCI; else { iomidi = mpu_port[dev]; @@ -2981,7 +2981,9 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc if (iomidi > 0) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, - iomidi, integrated_midi, + iomidi, + (integrated_midi ? + MPU401_INFO_INTEGRATED : 0), cm->irq, 0, &cm->rmidi)) < 0) { printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi); } diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index b3c94d83450..e77a4ce314b 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1184,7 +1184,7 @@ static void __devinit snd_cs4281_proc_init(struct cs4281 * chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "cs4281", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_cs4281_proc_read); + snd_info_set_text_ops(entry, chip, snd_cs4281_proc_read); if (! snd_card_proc_new(chip->card, "cs4281_BA0", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = chip; @@ -1379,6 +1379,13 @@ static int __devinit snd_cs4281_create(struct snd_card *card, chip->ba0_addr = pci_resource_start(pci, 0); chip->ba1_addr = pci_resource_start(pci, 1); + chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); + chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); + if (!chip->ba0 || !chip->ba1) { + snd_cs4281_free(chip); + return -ENOMEM; + } + if (request_irq(pci->irq, snd_cs4281_interrupt, SA_INTERRUPT|SA_SHIRQ, "CS4281", chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); @@ -1387,13 +1394,6 @@ static int __devinit snd_cs4281_create(struct snd_card *card, } chip->irq = pci->irq; - chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); - chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); - if (!chip->ba0 || !chip->ba1) { - snd_cs4281_free(chip); - return -ENOMEM; - } - tmp = snd_cs4281_chip_init(chip); if (tmp) { snd_cs4281_free(chip); diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 848d772ae3c..772dc52bfeb 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -48,8 +48,8 @@ MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)}," 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 */ -static int external_amp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; -static int thinkpad[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int external_amp[SNDRV_CARDS]; +static int thinkpad[SNDRV_CARDS]; static int mmap_valid[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; module_param_array(index, int, NULL, 0444); diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 69dbf542a6d..5c211443920 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2877,14 +2877,15 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) if (chip->region.idx[0].resource) snd_cs46xx_hw_stop(chip); + if (chip->irq >= 0) + free_irq(chip->irq, chip); + for (idx = 0; idx < 5; idx++) { struct snd_cs46xx_region *region = &chip->region.idx[idx]; if (region->remap_addr) iounmap(region->remap_addr); release_and_free_resource(region->resource); } - if (chip->irq >= 0) - free_irq(chip->irq, chip); if (chip->active_ctrl) chip->active_ctrl(chip, -chip->amplifier); diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index f407d2a5ce3..5c9711c0265 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -767,7 +767,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "dsp", card->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; - entry->c.text.read_size = 512; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -784,7 +783,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_symbol_table_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -797,7 +795,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_modules_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -810,7 +807,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_parameter_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -823,7 +819,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_sample_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -836,7 +831,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_task_tree_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -849,7 +843,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 1024; entry->c.text.read = cs46xx_dsp_proc_scb_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 2c4ee45fe10..3844d18af19 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -267,7 +267,6 @@ void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip, entry->private_data = scb_info; entry->mode = S_IFREG | S_IRUGO | S_IWUSR; - entry->c.text.read_size = 512; entry->c.text.read = cs46xx_dsp_proc_scb_info_read; if (snd_info_register(entry) < 0) { diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile index 08d8ee6547d..2911a8adc1f 100644 --- a/sound/pci/cs5535audio/Makefile +++ b/sound/pci/cs5535audio/Makefile @@ -4,5 +4,9 @@ snd-cs5535audio-objs := cs5535audio.o cs5535audio_pcm.o +ifdef CONFIG_PM +snd-cs5535audio-objs += cs5535audio_pm.o +endif + # Toplevel Module Dependency obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 2c1213a35dc..91c18a11fe8 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -1,5 +1,5 @@ /* - * Driver for audio on multifunction CS5535 companion device + * Driver for audio on multifunction CS5535/6 companion device * Copyright (C) Jaya Kumar * * Based on Jaroslav Kysela and Takashi Iwai's examples. @@ -40,16 +40,36 @@ #define DRIVER_NAME "cs5535audio" +static char *ac97_quirk; +module_param(ac97_quirk, charp, 0444); +MODULE_PARM_DESC(ac97_quirk, "AC'97 board specific workarounds."); + +static struct ac97_quirk ac97_quirks[] __devinitdata = { +#if 0 /* Not yet confirmed if all 5536 boards are HP only */ + { + .subvendor = PCI_VENDOR_ID_AMD, + .subdevice = PCI_DEVICE_ID_AMD_CS5536_AUDIO, + .name = "AMD RDK", + .type = AC97_TUNE_HP_ONLY + }, +#endif + {} +}; static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " DRIVER_NAME); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); + static struct pci_device_id snd_cs5535audio_ids[] __devinitdata = { - { PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, + { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) }, {} }; @@ -90,7 +110,8 @@ static unsigned short snd_cs5535audio_codec_read(struct cs5535audio *cs5535au, udelay(1); } while (--timeout); if (!timeout) - snd_printk(KERN_ERR "Failure reading cs5535 codec\n"); + snd_printk(KERN_ERR "Failure reading codec reg 0x%x," + "Last value=0x%x\n", reg, val); return (unsigned short) val; } @@ -148,6 +169,8 @@ static int snd_cs5535audio_mixer(struct cs5535audio *cs5535au) return err; } + snd_ac97_tune_hardware(cs5535au->ac97, ac97_quirks, ac97_quirk); + return 0; } @@ -347,6 +370,8 @@ static int __devinit snd_cs5535audio_probe(struct pci_dev *pci, if ((err = snd_cs5535audio_create(card, pci, &cs5535au)) < 0) goto probefail_out; + card->private_data = cs5535au; + if ((err = snd_cs5535audio_mixer(cs5535au)) < 0) goto probefail_out; @@ -383,6 +408,10 @@ static struct pci_driver driver = { .id_table = snd_cs5535audio_ids, .probe = snd_cs5535audio_probe, .remove = __devexit_p(snd_cs5535audio_remove), +#ifdef CONFIG_PM + .suspend = snd_cs5535audio_suspend, + .resume = snd_cs5535audio_resume, +#endif }; static int __init alsa_card_cs5535audio_init(void) diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h index 5e55a1a1ed6..4fd1f31a6cf 100644 --- a/sound/pci/cs5535audio/cs5535audio.h +++ b/sound/pci/cs5535audio/cs5535audio.h @@ -74,6 +74,8 @@ #define PRM_RDY_STS 0x00800000 #define ACC_CODEC_CNTL_WR_CMD (~0x80000000) #define ACC_CODEC_CNTL_RD_CMD 0x80000000 +#define ACC_CODEC_CNTL_LNK_SHUTDOWN 0x00040000 +#define ACC_CODEC_CNTL_LNK_WRM_RST 0x00020000 #define PRD_JMP 0x2000 #define PRD_EOP 0x4000 #define PRD_EOT 0x8000 @@ -88,6 +90,7 @@ struct cs5535audio_dma_ops { void (*disable_dma)(struct cs5535audio *cs5535au); void (*pause_dma)(struct cs5535audio *cs5535au); void (*setup_prd)(struct cs5535audio *cs5535au, u32 prd_addr); + u32 (*read_prd)(struct cs5535audio *cs5535au); u32 (*read_dma_pntr)(struct cs5535audio *cs5535au); }; @@ -103,11 +106,14 @@ struct cs5535audio_dma { struct snd_pcm_substream *substream; unsigned int buf_addr, buf_bytes; unsigned int period_bytes, periods; + int suspended; + u32 saved_prd; }; struct cs5535audio { struct snd_card *card; struct snd_ac97 *ac97; + struct snd_pcm *pcm; int irq; struct pci_dev *pci; unsigned long port; @@ -117,6 +123,8 @@ struct cs5535audio { struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS]; }; +int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state); +int snd_cs5535audio_resume(struct pci_dev *pci); int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535audio); #endif /* __SOUND_CS5535AUDIO_H */ diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 60bb82b2ff4..f0a48693d68 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -43,7 +43,8 @@ static struct snd_pcm_hardware snd_cs5535audio_playback = SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME ), .formats = ( SNDRV_PCM_FMTBIT_S16_LE @@ -193,6 +194,11 @@ static void cs5535audio_playback_setup_prd(struct cs5535audio *cs5535au, cs_writel(cs5535au, ACC_BM0_PRD, prd_addr); } +static u32 cs5535audio_playback_read_prd(struct cs5535audio *cs5535au) +{ + return cs_readl(cs5535au, ACC_BM0_PRD); +} + static u32 cs5535audio_playback_read_dma_pntr(struct cs5535audio *cs5535au) { return cs_readl(cs5535au, ACC_BM0_PNTR); @@ -219,6 +225,11 @@ static void cs5535audio_capture_setup_prd(struct cs5535audio *cs5535au, cs_writel(cs5535au, ACC_BM1_PRD, prd_addr); } +static u32 cs5535audio_capture_read_prd(struct cs5535audio *cs5535au) +{ + return cs_readl(cs5535au, ACC_BM1_PRD); +} + static u32 cs5535audio_capture_read_dma_pntr(struct cs5535audio *cs5535au) { return cs_readl(cs5535au, ACC_BM1_PNTR); @@ -285,9 +296,17 @@ static int snd_cs5535audio_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: dma->ops->enable_dma(cs5535au); break; + case SNDRV_PCM_TRIGGER_RESUME: + dma->ops->enable_dma(cs5535au); + dma->suspended = 0; + break; case SNDRV_PCM_TRIGGER_STOP: dma->ops->disable_dma(cs5535au); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + dma->ops->disable_dma(cs5535au); + dma->suspended = 1; + break; default: snd_printk(KERN_ERR "unhandled trigger\n"); err = -EINVAL; @@ -375,6 +394,7 @@ static struct cs5535audio_dma_ops snd_cs5535audio_playback_dma_ops = { .enable_dma = cs5535audio_playback_enable_dma, .disable_dma = cs5535audio_playback_disable_dma, .setup_prd = cs5535audio_playback_setup_prd, + .read_prd = cs5535audio_playback_read_prd, .pause_dma = cs5535audio_playback_pause_dma, .read_dma_pntr = cs5535audio_playback_read_dma_pntr, }; @@ -384,6 +404,7 @@ static struct cs5535audio_dma_ops snd_cs5535audio_capture_dma_ops = { .enable_dma = cs5535audio_capture_enable_dma, .disable_dma = cs5535audio_capture_disable_dma, .setup_prd = cs5535audio_capture_setup_prd, + .read_prd = cs5535audio_capture_read_prd, .pause_dma = cs5535audio_capture_pause_dma, .read_dma_pntr = cs5535audio_capture_read_dma_pntr, }; @@ -413,6 +434,7 @@ int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535au) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(cs5535au->pci), 64*1024, 128*1024); + cs5535au->pcm = pcm; return 0; } diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c new file mode 100644 index 00000000000..aad0e69db9c --- /dev/null +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -0,0 +1,123 @@ +/* + * Power management for audio on multifunction CS5535 companion device + * Copyright (C) Jaya Kumar + * + * 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 <linux/init.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/initval.h> +#include <sound/asoundef.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include "cs5535audio.h" + +static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au) +{ + /* + we depend on snd_ac97_suspend to tell the + AC97 codec to shutdown. the amd spec suggests + that the LNK_SHUTDOWN be done at the same time + that the codec power-down is issued. instead, + we do it just after rather than at the same + time. excluding codec specific build_ops->suspend + ac97 powerdown hits: + 0x8000 EAPD + 0x4000 Headphone amplifier + 0x0300 ADC & DAC + 0x0400 Analog Mixer powerdown (Vref on) + I am not sure if this is the best that we can do. + The remainder to be investigated are: + - analog mixer (vref off) 0x0800 + - AC-link powerdown 0x1000 + - codec internal clock 0x2000 + */ + + /* set LNK_SHUTDOWN to shutdown AC link */ + cs_writel(cs5535au, ACC_CODEC_CNTL, ACC_CODEC_CNTL_LNK_SHUTDOWN); + +} + +int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct cs5535audio *cs5535au = card->private_data; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) { + struct cs5535audio_dma *dma = &cs5535au->dmas[i]; + if (dma && dma->substream && !dma->suspended) + dma->saved_prd = dma->ops->read_prd(cs5535au); + } + snd_pcm_suspend_all(cs5535au->pcm); + snd_ac97_suspend(cs5535au->ac97); + /* save important regs, then disable aclink in hw */ + snd_cs5535audio_stop_hardware(cs5535au); + pci_disable_device(pci); + pci_save_state(pci); + + return 0; +} + +int snd_cs5535audio_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct cs5535audio *cs5535au = card->private_data; + u32 tmp; + int timeout; + int i; + + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_master(pci); + + /* set LNK_WRM_RST to reset AC link */ + cs_writel(cs5535au, ACC_CODEC_CNTL, ACC_CODEC_CNTL_LNK_WRM_RST); + + timeout = 50; + do { + tmp = cs_readl(cs5535au, ACC_CODEC_STATUS); + if (tmp & PRM_RDY_STS) + break; + udelay(1); + } while (--timeout); + + if (!timeout) + snd_printk(KERN_ERR "Failure getting AC Link ready\n"); + + /* we depend on ac97 to perform the codec power up */ + snd_ac97_resume(cs5535au->ac97); + /* set up rate regs, dma. actual initiation is done in trig */ + for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) { + struct cs5535audio_dma *dma = &cs5535au->dmas[i]; + if (dma && dma->substream && dma->suspended) { + dma->substream->ops->prepare(dma->substream); + dma->ops->setup_prd(cs5535au, dma->saved_prd); + } + } + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} + diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 42b11ba1d21..549673ea14a 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -46,13 +46,13 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS}," 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 */ -static int extin[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; -static int extout[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int extin[SNDRV_CARDS]; +static int extout[SNDRV_CARDS]; 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 */ +static int enable_ir[SNDRV_CARDS]; +static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard."); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 6bfa08436ef..42a358f989c 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -777,14 +777,6 @@ static int snd_emu10k1_dev_free(struct snd_device *device) static struct snd_emu_chip_details emu_chip_details[] = { /* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/ - /* Audigy4 SB0400 */ - {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102, - .driver = "Audigy2", .name = "Audigy 4 [SB0400]", - .id = "Audigy2", - .emu10k2_chip = 1, - .ca0108_chip = 1, - .spk71 = 1, - .ac97_chip = 1} , /* Tested by James@superbug.co.uk 3rd July 2005 */ /* DSP: CA0108-IAT * DAC: CS4382-KQ @@ -799,13 +791,59 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0108_chip = 1, .spk71 = 1, .ac97_chip = 1} , + /* Audigy4 (Not PRO) SB0610 */ + /* Tested by James@superbug.co.uk 4th April 2006 */ + /* A_IOCFG bits + * Output + * 0: ? + * 1: ? + * 2: ? + * 3: 0 - Digital Out, 1 - Line in + * 4: ? + * 5: ? + * 6: ? + * 7: ? + * Input + * 8: ? + * 9: ? + * A: Green jack sense (Front) + * B: ? + * C: Black jack sense (Rear/Side Right) + * D: Yellow jack sense (Center/LFE/Side Left) + * E: ? + * F: ? + * + * Digital Out/Line in switch using A_IOCFG bit 3 (0x08) + * 0 - Digital Out + * 1 - Line in + */ + /* Mic input not tested. + * Analog CD input not tested + * Digital Out not tested. + * Line in working. + * Audio output 5.1 working. Side outputs not working. + */ + /* DSP: CA10300-IAT LF + * DAC: Cirrus Logic CS4382-KQZ + * ADC: Philips 1361T + * AC97: Sigmatel STAC9750 + * CA0151: None + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102, + .driver = "Audigy2", .name = "Audigy 4 [SB0610]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .adc_1361t = 1, /* 24 bit capture instead of 16bit */ + .ac97_chip = 1} , /* Audigy 2 ZS Notebook Cardbus card.*/ /* Tested by James@superbug.co.uk 22th December 2005 */ /* Audio output 7.1/Headphones working. * Digital output working. (AC3 not checked, only PCM) * Audio inputs not tested. */ - /* DSP: Tiny2 + /* DSP: Tina2 * DAC: Wolfson WM8768/WM8568 * ADC: Wolfson WM8775 * AC97: None @@ -1421,16 +1459,3 @@ void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu) } } #endif - -/* memory.c */ -EXPORT_SYMBOL(snd_emu10k1_synth_alloc); -EXPORT_SYMBOL(snd_emu10k1_synth_free); -EXPORT_SYMBOL(snd_emu10k1_synth_bzero); -EXPORT_SYMBOL(snd_emu10k1_synth_copy_from_user); -EXPORT_SYMBOL(snd_emu10k1_memblk_map); -/* voice.c */ -EXPORT_SYMBOL(snd_emu10k1_voice_alloc); -EXPORT_SYMBOL(snd_emu10k1_voice_free); -/* io.c */ -EXPORT_SYMBOL(snd_emu10k1_ptr_read); -EXPORT_SYMBOL(snd_emu10k1_ptr_write); diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index d51290c1816..0fb27e4be07 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1055,8 +1055,7 @@ static int __devinit snd_emu10k1x_proc_init(struct emu10k1x * emu) struct snd_info_entry *entry; if(! snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_emu10k1x_proc_reg_read); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu10k1x_proc_reg_read); entry->c.text.write = snd_emu10k1x_proc_reg_write; entry->mode |= S_IWUSR; entry->private_data = emu; diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 2a9d12d1068..c31f3d0877f 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -777,6 +777,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, }; static char *audigy_remove_ctls[] = { /* Master/PCM controls on ac97 of Audigy has no effect */ + /* On the Audigy2 the AC97 playback is piped into + * the Philips ADC for 24bit capture */ "PCM Playback Switch", "PCM Playback Volume", "Master Mono Playback Switch", @@ -804,6 +806,47 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, "AMic Playback Volume", "Mic Playback Volume", NULL }; + static char *audigy_remove_ctls_1361t_adc[] = { + /* On the Audigy2 the AC97 playback is piped into + * the Philips ADC for 24bit capture */ + "PCM Playback Switch", + "PCM Playback Volume", + "Master Mono Playback Switch", + "Master Mono Playback Volume", + "Capture Source", + "Capture Switch", + "Capture Volume", + "Mic Capture Volume", + "Headphone Playback Switch", + "Headphone Playback Volume", + "3D Control - Center", + "3D Control - Depth", + "3D Control - Switch", + "Line2 Playback Volume", + "Line2 Capture Volume", + NULL + }; + static char *audigy_rename_ctls_1361t_adc[] = { + "Master Playback Switch", "Master Capture Switch", + "Master Playback Volume", "Master Capture Volume", + "Wave Master Playback Volume", "Master Playback Volume", + "PC Speaker Playback Switch", "PC Speaker Capture Switch", + "PC Speaker Playback Volume", "PC Speaker Capture Volume", + "Phone Playback Switch", "Phone Capture Switch", + "Phone Playback Volume", "Phone Capture Volume", + "Mic Playback Switch", "Mic Capture Switch", + "Mic Playback Volume", "Mic Capture Volume", + "Line Playback Switch", "Line Capture Switch", + "Line Playback Volume", "Line Capture Volume", + "CD Playback Switch", "CD Capture Switch", + "CD Playback Volume", "CD Capture Volume", + "Aux Playback Switch", "Aux Capture Switch", + "Aux Playback Volume", "Aux Capture Volume", + "Video Playback Switch", "Video Capture Switch", + "Video Playback Volume", "Video Capture Volume", + + NULL + }; if (emu->card_capabilities->ac97_chip) { struct snd_ac97_bus *pbus; @@ -834,7 +877,10 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ac97_write_cache(emu->ac97, AC97_MASTER, 0x0000); /* set capture source to mic */ snd_ac97_write_cache(emu->ac97, AC97_REC_SEL, 0x0000); - c = audigy_remove_ctls; + if (emu->card_capabilities->adc_1361t) + c = audigy_remove_ctls_1361t_adc; + else + c = audigy_remove_ctls; } else { /* * Credits for cards based on STAC9758: @@ -863,11 +909,15 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, } if (emu->audigy) - c = audigy_rename_ctls; + if (emu->card_capabilities->adc_1361t) + c = audigy_rename_ctls_1361t_adc; + else + c = audigy_rename_ctls; else c = emu10k1_rename_ctls; for (; *c; c += 2) rename_ctl(card, c[0], c[1]); + if (emu->card_capabilities->subsystem == 0x20071102) { /* Audigy 4 Pro */ rename_ctl(card, "Line2 Capture Volume", "Line1/Mic Capture Volume"); rename_ctl(card, "Analog Mix Capture Volume", "Line2 Capture Volume"); diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 90f1c52703a..b939e03aaed 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -532,57 +532,51 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu) struct snd_info_entry *entry; #ifdef CONFIG_SND_DEBUG if (! snd_card_proc_new(emu->card, "io_regs", &entry)) { - snd_info_set_text_ops(entry, emu, 1024, snd_emu_proc_io_reg_read); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read); entry->c.text.write = snd_emu_proc_io_reg_write; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read00a); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00a); entry->c.text.write = snd_emu_proc_ptr_reg_write00; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read00b); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00b); entry->c.text.write = snd_emu_proc_ptr_reg_write00; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20a); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20a); entry->c.text.write = snd_emu_proc_ptr_reg_write20; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20b); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20b); entry->c.text.write = snd_emu_proc_ptr_reg_write20; entry->mode |= S_IWUSR; } if (! snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) { - snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20c); - entry->c.text.write_size = 64; + snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20c); entry->c.text.write = snd_emu_proc_ptr_reg_write20; entry->mode |= S_IWUSR; } #endif if (! snd_card_proc_new(emu->card, "emu10k1", &entry)) - snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_read); + snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_read); if (emu->card_capabilities->emu10k2_chip) { if (! snd_card_proc_new(emu->card, "spdif-in", &entry)) - snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_spdif_read); + snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_spdif_read); } if (emu->card_capabilities->ca0151_chip) { if (! snd_card_proc_new(emu->card, "capture-rates", &entry)) - snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_rates_read); + snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_rates_read); } if (! snd_card_proc_new(emu->card, "voices", &entry)) - snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_voices_read); + snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_voices_read); if (! snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; @@ -616,7 +610,6 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; - entry->c.text.read_size = 128*1024; entry->c.text.read = snd_emu10k1_proc_acode_read; } return 0; diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index ef5304df8c1..029e7856c43 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -62,6 +62,8 @@ unsigned int snd_emu10k1_ptr_read(struct snd_emu10k1 * emu, unsigned int reg, un } } +EXPORT_SYMBOL(snd_emu10k1_ptr_read); + void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data) { unsigned int regptr; @@ -92,6 +94,8 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i } } +EXPORT_SYMBOL(snd_emu10k1_ptr_write); + unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg, unsigned int chn) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index e7ec98649f0..4fcaefe5a3c 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -287,6 +287,8 @@ int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *b return err; } +EXPORT_SYMBOL(snd_emu10k1_memblk_map); + /* * page allocation for DMA */ @@ -387,6 +389,7 @@ snd_emu10k1_synth_alloc(struct snd_emu10k1 *hw, unsigned int size) return (struct snd_util_memblk *)blk; } +EXPORT_SYMBOL(snd_emu10k1_synth_alloc); /* * free a synth sample area @@ -409,6 +412,7 @@ snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *memblk) return 0; } +EXPORT_SYMBOL(snd_emu10k1_synth_free); /* check new allocation range */ static void get_single_page_range(struct snd_util_memhdr *hdr, @@ -540,6 +544,8 @@ int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk return 0; } +EXPORT_SYMBOL(snd_emu10k1_synth_bzero); + /* * copy_from_user(blk + offset, data, size) */ @@ -568,3 +574,5 @@ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_me } while (offset < end_offset); return 0; } + +EXPORT_SYMBOL(snd_emu10k1_synth_copy_from_user); diff --git a/sound/pci/emu10k1/p17v.h b/sound/pci/emu10k1/p17v.h new file mode 100644 index 00000000000..7ddb5be632c --- /dev/null +++ b/sound/pci/emu10k1/p17v.h @@ -0,0 +1,111 @@ +/* + * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk> + * Driver p17v chips + * Version: 0.01 + * + * 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 + * + */ + +/******************************************************************************/ +/* Audigy2Value Tina (P17V) pointer-offset register set, + * accessed through the PTR20 and DATA24 registers */ +/******************************************************************************/ + +/* 00 - 07: Not used */ +#define P17V_PLAYBACK_FIFO_PTR 0x08 /* Current playback fifo pointer + * and number of sound samples in cache. + */ +/* 09 - 12: Not used */ +#define P17V_CAPTURE_FIFO_PTR 0x13 /* Current capture fifo pointer + * and number of sound samples in cache. + */ +/* 14 - 17: Not used */ +#define P17V_PB_CHN_SEL 0x18 /* P17v playback channel select */ +#define P17V_SE_SLOT_SEL_L 0x19 /* Sound Engine slot select low */ +#define P17V_SE_SLOT_SEL_H 0x1a /* Sound Engine slot select high */ +/* 1b - 1f: Not used */ +/* 20 - 2f: Not used */ +/* 30 - 3b: Not used */ +#define P17V_SPI 0x3c /* SPI interface register */ +#define P17V_I2C_ADDR 0x3d /* I2C Address */ +#define P17V_I2C_0 0x3e /* I2C Data */ +#define P17V_I2C_1 0x3f /* I2C Data */ + +#define P17V_START_AUDIO 0x40 /* Start Audio bit */ +/* 41 - 47: Reserved */ +#define P17V_START_CAPTURE 0x48 /* Start Capture bit */ +#define P17V_CAPTURE_FIFO_BASE 0x49 /* Record FIFO base address */ +#define P17V_CAPTURE_FIFO_SIZE 0x4a /* Record FIFO buffer size */ +#define P17V_CAPTURE_FIFO_INDEX 0x4b /* Record FIFO capture index */ +#define P17V_CAPTURE_VOL_H 0x4c /* P17v capture volume control */ +#define P17V_CAPTURE_VOL_L 0x4d /* P17v capture volume control */ +/* 4e - 4f: Not used */ +/* 50 - 5f: Not used */ +#define P17V_SRCSel 0x60 /* SRC48 and SRCMulti sample rate select + * and output select + */ +#define P17V_MIXER_AC97_10K1_VOL_L 0x61 /* 10K to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_10K1_VOL_H 0x62 /* 10K to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_P17V_VOL_L 0x63 /* P17V to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_P17V_VOL_H 0x64 /* P17V to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_SRP_REC_VOL_L 0x65 /* SRP Record to Mixer_AC97 input volume control */ +#define P17V_MIXER_AC97_SRP_REC_VOL_H 0x66 /* SRP Record to Mixer_AC97 input volume control */ +/* 67 - 68: Reserved */ +#define P17V_MIXER_Spdif_10K1_VOL_L 0x69 /* 10K to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_10K1_VOL_H 0x6A /* 10K to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_P17V_VOL_L 0x6B /* P17V to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_P17V_VOL_H 0x6C /* P17V to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_SRP_REC_VOL_L 0x6D /* SRP Record to Mixer_Spdif input volume control */ +#define P17V_MIXER_Spdif_SRP_REC_VOL_H 0x6E /* SRP Record to Mixer_Spdif input volume control */ +/* 6f - 70: Reserved */ +#define P17V_MIXER_I2S_10K1_VOL_L 0x71 /* 10K to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_10K1_VOL_H 0x72 /* 10K to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_P17V_VOL_L 0x73 /* P17V to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_P17V_VOL_H 0x74 /* P17V to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_SRP_REC_VOL_L 0x75 /* SRP Record to Mixer_I2S input volume control */ +#define P17V_MIXER_I2S_SRP_REC_VOL_H 0x76 /* SRP Record to Mixer_I2S input volume control */ +/* 77 - 78: Reserved */ +#define P17V_MIXER_AC97_ENABLE 0x79 /* Mixer AC97 input audio enable */ +#define P17V_MIXER_SPDIF_ENABLE 0x7A /* Mixer SPDIF input audio enable */ +#define P17V_MIXER_I2S_ENABLE 0x7B /* Mixer I2S input audio enable */ +#define P17V_AUDIO_OUT_ENABLE 0x7C /* Audio out enable */ +#define P17V_MIXER_ATT 0x7D /* SRP Mixer Attenuation Select */ +#define P17V_SRP_RECORD_SRR 0x7E /* SRP Record channel source Select */ +#define P17V_SOFT_RESET_SRP_MIXER 0x7F /* SRP and mixer soft reset */ + +#define P17V_AC97_OUT_MASTER_VOL_L 0x80 /* AC97 Output master volume control */ +#define P17V_AC97_OUT_MASTER_VOL_H 0x81 /* AC97 Output master volume control */ +#define P17V_SPDIF_OUT_MASTER_VOL_L 0x82 /* SPDIF Output master volume control */ +#define P17V_SPDIF_OUT_MASTER_VOL_H 0x83 /* SPDIF Output master volume control */ +#define P17V_I2S_OUT_MASTER_VOL_L 0x84 /* I2S Output master volume control */ +#define P17V_I2S_OUT_MASTER_VOL_H 0x85 /* I2S Output master volume control */ +/* 86 - 87: Not used */ +#define P17V_I2S_CHANNEL_SWAP_PHASE_INVERSE 0x88 /* I2S out mono channel swap + * and phase inverse */ +#define P17V_SPDIF_CHANNEL_SWAP_PHASE_INVERSE 0x89 /* SPDIF out mono channel swap + * and phase inverse */ +/* 8A: Not used */ +#define P17V_SRP_P17V_ESR 0x8B /* SRP_P17V estimated sample rate and rate lock */ +#define P17V_SRP_REC_ESR 0x8C /* SRP_REC estimated sample rate and rate lock */ +#define P17V_SRP_BYPASS 0x8D /* srps channel bypass and srps bypass */ +/* 8E - 92: Not used */ +#define P17V_I2S_SRC_SEL 0x93 /* I2SIN mode sel */ + + + + + + diff --git a/sound/pci/emu10k1/tina2.h b/sound/pci/emu10k1/tina2.h index 5c43abf03e8..f2d8eb6c89e 100644 --- a/sound/pci/emu10k1/tina2.h +++ b/sound/pci/emu10k1/tina2.h @@ -1,11 +1,7 @@ /* * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk> - * Driver p16v chips - * Version: 0.21 - * - * - * This code was initally based on code from ALSA's emu10k1x.c which is: - * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> + * Driver tina2 chips + * Version: 0.1 * * 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 diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c index 56ffb7dc3ee..94eca82dd4f 100644 --- a/sound/pci/emu10k1/voice.c +++ b/sound/pci/emu10k1/voice.c @@ -139,6 +139,8 @@ int snd_emu10k1_voice_alloc(struct snd_emu10k1 *emu, int type, int number, return result; } +EXPORT_SYMBOL(snd_emu10k1_voice_alloc); + int snd_emu10k1_voice_free(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *pvoice) { @@ -153,3 +155,5 @@ int snd_emu10k1_voice_free(struct snd_emu10k1 *emu, spin_unlock_irqrestore(&emu->voice_lock, flags); return 0; } + +EXPORT_SYMBOL(snd_emu10k1_voice_free); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index ca9e34e88f6..9d46bbee2a4 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1915,7 +1915,7 @@ static void __devinit snd_ensoniq_proc_init(struct ensoniq * ensoniq) struct snd_info_entry *entry; if (! snd_card_proc_new(ensoniq->card, "audiopci", &entry)) - snd_info_set_text_ops(entry, ensoniq, 1024, snd_ensoniq_proc_read); + snd_info_set_text_ops(entry, ensoniq, snd_ensoniq_proc_read); } /* diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 6f9094ca4fb..ca6603fe0b1 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1756,7 +1756,8 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci, } } if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, 1, chip->irq, 0, &chip->rmidi) < 0) { + chip->mpu_port, MPU401_INFO_INTEGRATED, + chip->irq, 0, &chip->rmidi) < 0) { printk(KERN_ERR "es1938: unable to initialize MPU-401\n"); } else { // this line is vital for MIDI interrupt handling on ess-solo1 diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 5ff4175c7b6..bfa0876e715 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -132,7 +132,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * static int total_bufsize[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1024 }; static int pcm_substreams_p[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4 }; static int pcm_substreams_c[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1 }; -static int clock[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int clock[SNDRV_CARDS]; static int use_pm[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; static int enable_mpu[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; #ifdef SUPPORT_JOYSTICK @@ -2727,7 +2727,8 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci, } if (enable_mpu[dev]) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->io_port + ESM_MPU401_PORT, 1, + chip->io_port + ESM_MPU401_PORT, + MPU401_INFO_INTEGRATED, chip->irq, 0, &chip->rmidi)) < 0) { printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n"); } diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index d72fc28c580..0afa573dd24 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -56,7 +56,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * * 3 = MediaForte 64-PCR * High 16-bits are video (radio) device number + 1 */ -static int tea575x_tuner[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; +static int tea575x_tuner[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the FM801 soundcard."); @@ -1448,7 +1448,8 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, return err; } if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801, - FM801_REG(chip, MPU401_DATA), 1, + FM801_REG(chip, MPU401_DATA), + MPU401_INFO_INTEGRATED, chip->irq, 0, &chip->rmidi)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index ddfb5ff7fb8..dbacba6177d 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,5 +1,5 @@ snd-hda-intel-objs := hda_intel.o -snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o patch_sigmatel.o patch_si3054.o +snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o patch_sigmatel.o patch_si3054.o patch_atihdmi.o ifdef CONFIG_PROC_FS snd-hda-codec-objs += hda_proc.o endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5bee3b53647..8c2a8174ece 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -86,6 +86,8 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int dire return res; } +EXPORT_SYMBOL(snd_hda_codec_read); + /** * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec @@ -108,6 +110,8 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, return err; } +EXPORT_SYMBOL(snd_hda_codec_write); + /** * snd_hda_sequence_write - sequence writes * @codec: the HDA codec @@ -122,6 +126,8 @@ void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq) snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param); } +EXPORT_SYMBOL(snd_hda_sequence_write); + /** * snd_hda_get_sub_nodes - get the range of sub nodes * @codec: the HDA codec @@ -140,6 +146,8 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *sta return (int)(parm & 0x7fff); } +EXPORT_SYMBOL(snd_hda_get_sub_nodes); + /** * snd_hda_get_connections - get connection list * @codec: the HDA codec @@ -256,6 +264,8 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) return 0; } +EXPORT_SYMBOL(snd_hda_queue_unsol_event); + /* * process queueud unsolicited events */ @@ -384,6 +394,7 @@ int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, return 0; } +EXPORT_SYMBOL(snd_hda_bus_new); /* * find a matching codec preset @@ -587,6 +598,8 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, return 0; } +EXPORT_SYMBOL(snd_hda_codec_new); + /** * snd_hda_codec_setup_stream - set up the codec for streaming * @codec: the CODEC to set up @@ -609,6 +622,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stre snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format); } +EXPORT_SYMBOL(snd_hda_codec_setup_stream); /* * amp access functions @@ -1294,6 +1308,7 @@ int snd_hda_build_controls(struct hda_bus *bus) return 0; } +EXPORT_SYMBOL(snd_hda_build_controls); /* * stream formats @@ -1382,6 +1397,8 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, return val; } +EXPORT_SYMBOL(snd_hda_calc_stream_format); + /** * snd_hda_query_supported_pcm - query the supported PCM rates and formats * @codec: the HDA codec @@ -1663,6 +1680,7 @@ int snd_hda_build_pcms(struct hda_bus *bus) return 0; } +EXPORT_SYMBOL(snd_hda_build_pcms); /** * snd_hda_check_board_config - compare the current codec with the config table @@ -2165,6 +2183,8 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state) return 0; } +EXPORT_SYMBOL(snd_hda_suspend); + /** * snd_hda_resume - resume the codecs * @bus: the HDA bus @@ -2187,6 +2207,8 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } +EXPORT_SYMBOL(snd_hda_resume); + /** * snd_hda_resume_ctls - resume controls in the new control list * @codec: the HDA codec @@ -2247,25 +2269,6 @@ int snd_hda_resume_spdif_in(struct hda_codec *codec) #endif /* - * symbols exported for controller modules - */ -EXPORT_SYMBOL(snd_hda_codec_read); -EXPORT_SYMBOL(snd_hda_codec_write); -EXPORT_SYMBOL(snd_hda_sequence_write); -EXPORT_SYMBOL(snd_hda_get_sub_nodes); -EXPORT_SYMBOL(snd_hda_queue_unsol_event); -EXPORT_SYMBOL(snd_hda_bus_new); -EXPORT_SYMBOL(snd_hda_codec_new); -EXPORT_SYMBOL(snd_hda_codec_setup_stream); -EXPORT_SYMBOL(snd_hda_calc_stream_format); -EXPORT_SYMBOL(snd_hda_build_pcms); -EXPORT_SYMBOL(snd_hda_build_controls); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_hda_suspend); -EXPORT_SYMBOL(snd_hda_resume); -#endif - -/* * INIT part */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e821d65afa1..4070b5cd9b6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -82,6 +82,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH8}," "{ATI, SB450}," "{ATI, SB600}," + "{ATI, RS600}," "{VIA, VT8251}," "{VIA, VT8237A}," "{SiS, SIS966}," @@ -167,6 +168,12 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ULI_PLAYBACK_INDEX 5 #define ULI_NUM_PLAYBACK 6 +/* ATI HDMI has 1 playback and 0 capture */ +#define ATIHDMI_CAPTURE_INDEX 0 +#define ATIHDMI_NUM_CAPTURE 0 +#define ATIHDMI_PLAYBACK_INDEX 0 +#define ATIHDMI_NUM_PLAYBACK 1 + /* this number is statically defined for simplicity */ #define MAX_AZX_DEV 16 @@ -331,6 +338,7 @@ struct azx { enum { AZX_DRIVER_ICH, AZX_DRIVER_ATI, + AZX_DRIVER_ATIHDMI, AZX_DRIVER_VIA, AZX_DRIVER_SIS, AZX_DRIVER_ULI, @@ -340,6 +348,7 @@ enum { static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_ATI] = "HDA ATI SB", + [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI", [AZX_DRIVER_VIA] = "HDA VIA VT82xx", [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", @@ -1393,10 +1402,10 @@ static int azx_free(struct azx *chip) msleep(1); } - if (chip->remap_addr) - iounmap(chip->remap_addr); if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); + if (chip->remap_addr) + iounmap(chip->remap_addr); if (chip->bdl.area) snd_dma_free_pages(&chip->bdl); @@ -1495,6 +1504,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->playback_index_offset = ULI_PLAYBACK_INDEX; chip->capture_index_offset = ULI_CAPTURE_INDEX; break; + case AZX_DRIVER_ATIHDMI: + chip->playback_streams = ATIHDMI_NUM_PLAYBACK; + chip->capture_streams = ATIHDMI_NUM_CAPTURE; + chip->playback_index_offset = ATIHDMI_PLAYBACK_INDEX; + chip->capture_index_offset = ATIHDMI_CAPTURE_INDEX; + break; default: chip->playback_streams = ICH6_NUM_PLAYBACK; chip->capture_streams = ICH6_NUM_CAPTURE; @@ -1621,6 +1636,7 @@ static struct pci_device_id azx_ids[] __devinitdata = { { 0x8086, 0x284b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH8 */ { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */ { 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */ + { 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h index acaef3c811b..0b668793fac 100644 --- a/sound/pci/hda/hda_patch.h +++ b/sound/pci/hda/hda_patch.h @@ -12,6 +12,8 @@ extern struct hda_codec_preset snd_hda_preset_analog[]; extern struct hda_codec_preset snd_hda_preset_sigmatel[]; /* SiLabs 3054/3055 modem codecs */ extern struct hda_codec_preset snd_hda_preset_si3054[]; +/* ATI HDMI codecs */ +extern struct hda_codec_preset snd_hda_preset_atihdmi[]; static const struct hda_codec_preset *hda_preset_tables[] = { snd_hda_preset_realtek, @@ -19,5 +21,6 @@ static const struct hda_codec_preset *hda_preset_tables[] = { snd_hda_preset_analog, snd_hda_preset_sigmatel, snd_hda_preset_si3054, + snd_hda_preset_atihdmi, NULL }; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index ca514a6a587..c2f0fe85bf3 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -182,6 +182,10 @@ static void print_pin_caps(struct snd_info_buffer *buffer, snd_iprintf(buffer, " OUT"); if (caps & AC_PINCAP_HP_DRV) snd_iprintf(buffer, " HP"); + if (caps & AC_PINCAP_EAPD) + snd_iprintf(buffer, " EAPD"); + if (caps & AC_PINCAP_PRES_DETECT) + snd_iprintf(buffer, " Detect"); snd_iprintf(buffer, "\n"); caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, @@ -318,7 +322,7 @@ int snd_hda_codec_proc_new(struct hda_codec *codec) if (err < 0) return err; - snd_info_set_text_ops(entry, codec, 32 * 1024, print_codec_info); + snd_info_set_text_ops(entry, codec, print_codec_info); return 0; } diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 40f000ba136..dd4e00a82b5 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -789,6 +789,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { { .modelname = "3stack", .config = AD1986A_3STACK }, { .pci_subvendor = 0x10de, .pci_subdevice = 0xcb84, .config = AD1986A_3STACK }, /* ASUS A8N-VM CSM */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x81b3, + .config = AD1986A_3STACK }, /* ASUS P5RD2-VM / P5GPL-X SE */ { .modelname = "laptop", .config = AD1986A_LAPTOP }, { .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e, .config = AD1986A_LAPTOP }, /* FSC V2060 */ @@ -809,6 +811,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { .config = AD1986A_LAPTOP_EAPD }, /* ASUS Z62F */ { .pci_subvendor = 0x103c, .pci_subdevice = 0x30af, .config = AD1986A_LAPTOP_EAPD }, /* HP Compaq Presario B2800 */ + { .pci_subvendor = 0x17aa, .pci_subdevice = 0x2066, + .config = AD1986A_LAPTOP_EAPD }, /* Lenovo 3000 N100-07684JU */ {} }; @@ -963,7 +967,7 @@ static struct snd_kcontrol_new ad1983_mixers[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", .info = ad1983_spdif_route_info, .get = ad1983_spdif_route_get, .put = ad1983_spdif_route_put, @@ -1103,7 +1107,7 @@ static struct snd_kcontrol_new ad1981_mixers[] = { /* identical with AD1983 */ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route", + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", .info = ad1983_spdif_route_info, .get = ad1983_spdif_route_get, .put = ad1983_spdif_route_put, @@ -1329,13 +1333,60 @@ static int ad1981_hp_init(struct hda_codec *codec) return 0; } +/* configuration for Lenovo Thinkpad T60 */ +static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + /* identical with AD1983 */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, + { } /* end */ +}; + +static struct hda_input_mux ad1981_thinkpad_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Mix", 0x2 }, + { "CD", 0x4 }, + }, +}; + /* models */ -enum { AD1981_BASIC, AD1981_HP }; +enum { AD1981_BASIC, AD1981_HP, AD1981_THINKPAD }; static struct hda_board_config ad1981_cfg_tbl[] = { { .modelname = "hp", .config = AD1981_HP }, /* All HP models */ { .pci_subvendor = 0x103c, .config = AD1981_HP }, + { .pci_subvendor = 0x30b0, .pci_subdevice = 0x103c, + .config = AD1981_HP }, /* HP nx6320 (reversed SSID, H/W bug) */ + { .modelname = "thinkpad", .config = AD1981_THINKPAD }, + /* Lenovo Thinkpad T60/X60/Z6xx */ + { .pci_subvendor = 0x17aa, .config = AD1981_THINKPAD }, + { .pci_subvendor = 0x1014, .pci_subdevice = 0x0597, + .config = AD1981_THINKPAD }, /* Z60m/t */ { .modelname = "basic", .config = AD1981_BASIC }, {} }; @@ -1381,6 +1432,10 @@ static int patch_ad1981(struct hda_codec *codec) codec->patch_ops.init = ad1981_hp_init; codec->patch_ops.unsol_event = ad1981_hp_unsol_event; break; + case AD1981_THINKPAD: + spec->mixers[0] = ad1981_thinkpad_mixers; + spec->input_mux = &ad1981_thinkpad_capture_source; + break; } return 0; diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c new file mode 100644 index 00000000000..a27440ffd1c --- /dev/null +++ b/sound/pci/hda/patch_atihdmi.c @@ -0,0 +1,165 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * HD audio interface patch for ATI HDMI codecs + * + * Copyright (c) 2006 ATI Technologies Inc. + * + * + * This driver 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 driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + +struct atihdmi_spec { + struct hda_multi_out multiout; + + struct hda_pcm pcm_rec; +}; + +static struct hda_verb atihdmi_basic_init[] = { + /* enable digital output on pin widget */ + { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + {} /* terminator */ +}; + +/* + * Controls + */ +static int atihdmi_build_controls(struct hda_codec *codec) +{ + struct atihdmi_spec *spec = codec->spec; + int err; + + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); + if (err < 0) + return err; + + return 0; +} + +static int atihdmi_init(struct hda_codec *codec) +{ + snd_hda_sequence_write(codec, atihdmi_basic_init); + return 0; +} + +#ifdef CONFIG_PM +/* + * resume + */ +static int atihdmi_resume(struct hda_codec *codec) +{ + atihdmi_init(codec); + snd_hda_resume_spdif_out(codec); + + return 0; +} +#endif + +/* + * Digital out + */ +static int atihdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct atihdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int atihdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct atihdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static struct hda_pcm_stream atihdmi_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = 0x2, /* NID to query formats and rates and setup streams */ + .ops = { + .open = atihdmi_dig_playback_pcm_open, + .close = atihdmi_dig_playback_pcm_close + }, +}; + +static int atihdmi_build_pcms(struct hda_codec *codec) +{ + struct atihdmi_spec *spec = codec->spec; + struct hda_pcm *info = &spec->pcm_rec; + + codec->num_pcms = 1; + codec->pcm_info = info; + + info->name = "ATI HDMI"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback; + + return 0; +} + +static void atihdmi_free(struct hda_codec *codec) +{ + kfree(codec->spec); +} + +static struct hda_codec_ops atihdmi_patch_ops = { + .build_controls = atihdmi_build_controls, + .build_pcms = atihdmi_build_pcms, + .init = atihdmi_init, + .free = atihdmi_free, +#ifdef CONFIG_PM + .resume = atihdmi_resume, +#endif +}; + +static int patch_atihdmi(struct hda_codec *codec) +{ + struct atihdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 2; + spec->multiout.dig_out_nid = 0x2; /* NID for copying analog to digital, + * seems to be unused in pure-digital + * case. */ + + codec->patch_ops = atihdmi_patch_ops; + + return 0; +} + +/* + * patch entries + */ +struct hda_codec_preset snd_hda_preset_atihdmi[] = { + { .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, + {} /* terminator */ +}; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f0e9a9c9078..98b9f16c26f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2174,6 +2174,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .modelname = "lg", .config = ALC880_LG }, { .pci_subvendor = 0x1854, .pci_subdevice = 0x003b, .config = ALC880_LG }, + { .pci_subvendor = 0x1854, .pci_subdevice = 0x0068, .config = ALC880_LG }, { .modelname = "lg-lw", .config = ALC880_LG_LW }, { .pci_subvendor = 0x1854, .pci_subdevice = 0x0018, .config = ALC880_LG_LW }, @@ -3105,6 +3106,7 @@ static struct hda_verb alc260_init_verbs[] = { { } }; +#if 0 /* should be identical with alc260_init_verbs? */ static struct hda_verb alc260_hp_init_verbs[] = { /* Headphone and output */ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, @@ -3151,6 +3153,7 @@ static struct hda_verb alc260_hp_init_verbs[] = { {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, { } }; +#endif static struct hda_verb alc260_hp_3013_init_verbs[] = { /* Line out and output */ @@ -3822,12 +3825,16 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .modelname = "basic", .config = ALC260_BASIC }, { .pci_subvendor = 0x104d, .pci_subdevice = 0x81bb, .config = ALC260_BASIC }, /* Sony VAIO */ + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81cc, + .config = ALC260_BASIC }, /* Sony VAIO VGN-S3HP */ + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81cd, + .config = ALC260_BASIC }, /* Sony VAIO */ { .pci_subvendor = 0x152d, .pci_subdevice = 0x0729, .config = ALC260_BASIC }, /* CTL Travel Master U553W */ { .modelname = "hp", .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP }, - { .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP_3013 }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3013, .config = ALC260_HP_3013 }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3014, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3015, .config = ALC260_HP }, @@ -3862,7 +3869,7 @@ static struct alc_config_preset alc260_presets[] = { .mixers = { alc260_base_output_mixer, alc260_input_mixer, alc260_capture_alt_mixer }, - .init_verbs = { alc260_hp_init_verbs }, + .init_verbs = { alc260_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), @@ -4094,21 +4101,6 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 3, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, { } /* end */ }; @@ -4342,8 +4334,6 @@ static struct alc_config_preset alc882_presets[] = { .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, .dig_in_nid = ALC882_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), .channel_mode = alc882_ch_modes, @@ -4355,8 +4345,6 @@ static struct alc_config_preset alc882_presets[] = { .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, .dig_in_nid = ALC882_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), .channel_mode = alc882_sixstack_modes, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c440fb9860..36f199442fd 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -41,6 +41,7 @@ #define STAC_REF 0 #define STAC_D945GTP3 1 #define STAC_D945GTP5 2 +#define STAC_MACMINI 3 struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; @@ -52,6 +53,7 @@ struct sigmatel_spec { unsigned int mic_switch: 1; unsigned int alt_switch: 1; unsigned int hp_detect: 1; + unsigned int gpio_mute: 1; /* playback */ struct hda_multi_out multiout; @@ -293,6 +295,7 @@ static unsigned int *stac922x_brd_tbl[] = { ref922x_pin_configs, d945gtp3_pin_configs, d945gtp5_pin_configs, + NULL, /* STAC_MACMINI */ }; static struct hda_board_config stac922x_cfg_tbl[] = { @@ -324,6 +327,9 @@ static struct hda_board_config stac922x_cfg_tbl[] = { { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x0417, .config = STAC_D945GTP5 }, /* Intel D975XBK - 5 Stack */ + { .pci_subvendor = 0x8384, + .pci_subdevice = 0x7680, + .config = STAC_MACMINI }, /* Apple Mac Mini (early 2006) */ {} /* terminator */ }; @@ -841,6 +847,19 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const } } + if (imux->num_items == 1) { + /* + * Set the current input for the muxes. + * The STAC9221 has two input muxes with identical source + * NID lists. Hopefully this won't get confused. + */ + for (i = 0; i < spec->num_muxes; i++) { + snd_hda_codec_write(codec, spec->mux_nids[i], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[0].index); + } + } + return 0; } @@ -946,6 +965,45 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) return 1; } +/* + * Early 2006 Intel Macintoshes with STAC9220X5 codecs seem to have a + * funky external mute control using GPIO pins. + */ + +static void stac922x_gpio_mute(struct hda_codec *codec, int pin, int muted) +{ + unsigned int gpiostate, gpiomask, gpiodir; + + gpiostate = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + + if (!muted) + gpiostate |= (1 << pin); + else + gpiostate &= ~(1 << pin); + + gpiomask = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_MASK, 0); + gpiomask |= (1 << pin); + + gpiodir = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DIRECTION, 0); + gpiodir |= (1 << pin); + + /* AppleHDA seems to do this -- WTF is this verb?? */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); + + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_MASK, gpiomask); + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DIRECTION, gpiodir); + + msleep(1); + + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DATA, gpiostate); +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -982,6 +1040,11 @@ static int stac92xx_init(struct hda_codec *codec) stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, AC_PINCTL_IN_EN); + if (spec->gpio_mute) { + stac922x_gpio_mute(codec, 0, 0); + stac922x_gpio_mute(codec, 1, 0); + } + return 0; } @@ -1132,7 +1195,7 @@ static int patch_stac922x(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, stac922x_cfg_tbl); if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, using BIOS defaults\n"); - else { + else if (stac922x_brd_tbl[spec->board_config] != NULL) { spec->num_pins = 10; spec->pin_nids = stac922x_pin_nids; spec->pin_configs = stac922x_brd_tbl[spec->board_config]; @@ -1154,6 +1217,9 @@ static int patch_stac922x(struct hda_codec *codec) return err; } + if (spec->board_config == STAC_MACMINI) + spec->gpio_mute = 1; + codec->patch_ops = stac92xx_patch_ops; return 0; @@ -1262,13 +1328,13 @@ static int vaio_master_sw_put(struct snd_kcontrol *kcontrol, int change; change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0, - 0x80, valp[0] & 0x80); + 0x80, (valp[0] ? 0 : 0x80)); change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0, - 0x80, valp[1] & 0x80); + 0x80, (valp[1] ? 0 : 0x80)); snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, - 0x80, valp[0] & 0x80); + 0x80, (valp[0] ? 0 : 0x80)); snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, - 0x80, valp[1] & 0x80); + 0x80, (valp[1] ? 0 : 0x80)); return change; } @@ -1370,6 +1436,12 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x }, { .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x }, { .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x }, + { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac922x }, + { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac922x }, + { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac922x }, + { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac922x }, + { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac922x }, + { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac922x }, { .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x }, { .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x }, { .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x }, diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 336dc489aee..ca74f5b85f4 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -1281,9 +1281,15 @@ static int aureon_set_headphone_amp(struct snd_ice1712 *ice, int enable) tmp2 = tmp = snd_ice1712_gpio_read(ice); if (enable) - tmp |= AUREON_HP_SEL; + if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT) + tmp |= AUREON_HP_SEL; + else + tmp |= PRODIGY_HP_SEL; else - tmp &= ~ AUREON_HP_SEL; + if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT) + tmp &= ~ AUREON_HP_SEL; + else + tmp &= ~ PRODIGY_HP_SEL; if (tmp != tmp2) { snd_ice1712_gpio_write(ice, tmp); return 1; @@ -2079,16 +2085,16 @@ static unsigned char prodigy71_eeprom[] __devinitdata = { }; static unsigned char prodigy71lt_eeprom[] __devinitdata = { - 0x0b, /* SYSCINF: clock 512, spdif-in/ADC, 4DACs */ + 0x4b, /* SYSCINF: clock 512, spdif-in/ADC, 4DACs */ 0x80, /* ACLINK: I2S */ 0xfc, /* I2S: vol, 96k, 24bit, 192k */ - 0xc3, /* SPDUF: out-en, out-int */ - 0x00, /* GPIO_DIR */ - 0x07, /* GPIO_DIR1 */ - 0x00, /* GPIO_DIR2 */ - 0xff, /* GPIO_MASK */ - 0xf8, /* GPIO_MASK1 */ - 0xff, /* GPIO_MASK2 */ + 0xc3, /* SPDIF: out-en, out-int, spdif-in */ + 0xff, /* GPIO_DIR */ + 0xff, /* GPIO_DIR1 */ + 0x5f, /* GPIO_DIR2 */ + 0x00, /* GPIO_MASK */ + 0x00, /* GPIO_MASK1 */ + 0x00, /* GPIO_MASK2 */ 0x00, /* GPIO_STATE */ 0x00, /* GPIO_STATE1 */ 0x00, /* GPIO_STATE2 */ diff --git a/sound/pci/ice1712/aureon.h b/sound/pci/ice1712/aureon.h index 98a6752280f..3b7bea656c5 100644 --- a/sound/pci/ice1712/aureon.h +++ b/sound/pci/ice1712/aureon.h @@ -58,5 +58,6 @@ extern struct snd_ice1712_card_info snd_vt1724_aureon_cards[]; #define PRODIGY_WM_CS (1 << 8) #define PRODIGY_SPI_MOSI (1 << 10) #define PRODIGY_SPI_CLK (1 << 9) +#define PRODIGY_HP_SEL (1 << 5) #endif /* __SOUND_AUREON_H */ diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index 2c529e74138..b135389fec6 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -1031,6 +1031,9 @@ struct snd_ice1712_card_info snd_ice1712_ews_cards[] __devinitdata = { .model = "dmx6fire", .chip_init = snd_ice1712_ews_init, .build_controls = snd_ice1712_ews_add_controls, + .mpu401_1_name = "MIDI-Front DMX6fire", + .mpu401_2_name = "Wavetable DMX6fire", + .mpu401_2_info_flags = MPU401_INFO_OUTPUT, }, { } /* terminator */ }; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index c56793b381e..845907159b7 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -61,7 +61,6 @@ #include <sound/core.h> #include <sound/cs8427.h> #include <sound/info.h> -#include <sound/mpu401.h> #include <sound/initval.h> #include <sound/asoundef.h> @@ -1596,7 +1595,7 @@ static void __devinit snd_ice1712_proc_init(struct snd_ice1712 * ice) struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "ice1712", &entry)) - snd_info_set_text_ops(entry, ice, 1024, snd_ice1712_proc_read); + snd_info_set_text_ops(entry, ice, snd_ice1712_proc_read); } /* @@ -2398,13 +2397,14 @@ static int __devinit snd_ice1712_chip_init(struct snd_ice1712 *ice) udelay(200); outb(ICE1712_NATIVE, ICEREG(ice, CONTROL)); udelay(200); - if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DMX6FIRE && !ice->dxr_enable) { - /* Limit active ADCs and DACs to 6; */ - /* Note: DXR extension not supported */ - pci_write_config_byte(ice->pci, 0x60, 0x2a); - } else { - pci_write_config_byte(ice->pci, 0x60, ice->eeprom.data[ICE_EEP1_CODEC]); - } + if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DMX6FIRE && + !ice->dxr_enable) + /* Set eeprom value to limit active ADCs and DACs to 6; + * Also disable AC97 as no hardware in standard 6fire card/box + * Note: DXR extensions are not currently supported + */ + ice->eeprom.data[ICE_EEP1_CODEC] = 0x3a; + pci_write_config_byte(ice->pci, 0x60, ice->eeprom.data[ICE_EEP1_CODEC]); pci_write_config_byte(ice->pci, 0x61, ice->eeprom.data[ICE_EEP1_ACLINK]); pci_write_config_byte(ice->pci, 0x62, ice->eeprom.data[ICE_EEP1_I2SID]); pci_write_config_byte(ice->pci, 0x63, ice->eeprom.data[ICE_EEP1_SPDIF]); @@ -2737,21 +2737,38 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, if (! c->no_mpu401) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, - ICEREG(ice, MPU1_CTRL), 1, + ICEREG(ice, MPU1_CTRL), + (c->mpu401_1_info_flags | + MPU401_INFO_INTEGRATED), ice->irq, 0, &ice->rmidi[0])) < 0) { snd_card_free(card); return err; } - - if (ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) + if (c->mpu401_1_name) + /* Prefered name available in card_info */ + snprintf(ice->rmidi[0]->name, + sizeof(ice->rmidi[0]->name), + "%s %d", c->mpu401_1_name, card->number); + + if (ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) { + /* 2nd port used */ if ((err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712, - ICEREG(ice, MPU2_CTRL), 1, + ICEREG(ice, MPU2_CTRL), + (c->mpu401_2_info_flags | + MPU401_INFO_INTEGRATED), ice->irq, 0, &ice->rmidi[1])) < 0) { snd_card_free(card); return err; } + if (c->mpu401_2_name) + /* Prefered name available in card_info */ + snprintf(ice->rmidi[1]->name, + sizeof(ice->rmidi[1]->name), + "%s %d", c->mpu401_2_name, + card->number); + } } snd_ice1712_set_input_clock_source(ice, 0); diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 053f8e56fd6..ce27eac40d4 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -29,6 +29,7 @@ #include <sound/ak4xxx-adda.h> #include <sound/ak4114.h> #include <sound/pcm.h> +#include <sound/mpu401.h> /* @@ -495,6 +496,10 @@ struct snd_ice1712_card_info { int (*chip_init)(struct snd_ice1712 *); int (*build_controls)(struct snd_ice1712 *); unsigned int no_mpu401: 1; + unsigned int mpu401_1_info_flags; + unsigned int mpu401_2_info_flags; + const char *mpu401_1_name; + const char *mpu401_2_name; unsigned int eeprom_size; unsigned char *eeprom_data; }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index b1c007e022d..34a58c629f4 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1293,7 +1293,7 @@ static void __devinit snd_vt1724_proc_init(struct snd_ice1712 * ice) struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "ice1724", &entry)) - snd_info_set_text_ops(entry, ice, 1024, snd_vt1724_proc_read); + snd_info_set_text_ops(entry, ice, snd_vt1724_proc_read); } /* @@ -2388,7 +2388,8 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, if (! c->no_mpu401) { if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, - ICEREG1724(ice, MPU_CTRL), 1, + ICEREG1724(ice, MPU_CTRL), + MPU401_INFO_INTEGRATED, ice->irq, 0, &ice->rmidi[0])) < 0) { snd_card_free(card); diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index d23fb3fc213..0efcad9260a 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -680,9 +680,8 @@ static void wm_proc_init(struct snd_ice1712 *ice) { struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) { - snd_info_set_text_ops(entry, ice, 1024, wm_proc_regs_read); + snd_info_set_text_ops(entry, ice, wm_proc_regs_read); entry->mode |= S_IWUSR; - entry->c.text.write_size = 1024; entry->c.text.write = wm_proc_regs_write; } } @@ -705,9 +704,8 @@ static void cs_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buff static void cs_proc_init(struct snd_ice1712 *ice) { struct snd_info_entry *entry; - if (! snd_card_proc_new(ice->card, "cs_codec", &entry)) { - snd_info_set_text_ops(entry, ice, 1024, cs_proc_regs_read); - } + if (! snd_card_proc_new(ice->card, "cs_codec", &entry)) + snd_info_set_text_ops(entry, ice, cs_proc_regs_read); } diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 0df7602568e..edc14475ef8 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -66,7 +66,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ -static int ac97_clock = 0; +static int ac97_clock; static char *ac97_quirk; static int buggy_semaphore; static int buggy_irq = -1; /* auto-check */ @@ -1807,6 +1807,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x1028, + .subdevice = 0x014e, + .name = "Dell D800", /* STAC9750/51 */ + .type = AC97_TUNE_HP_ONLY + }, + { + .subvendor = 0x1028, .subdevice = 0x0163, .name = "Dell Unknown", /* STAC9750/51 */ .type = AC97_TUNE_HP_ONLY @@ -2645,7 +2651,7 @@ static void __devinit snd_intel8x0_proc_init(struct intel8x0 * chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "intel8x0", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_intel8x0_proc_read); + snd_info_set_text_ops(entry, chip, snd_intel8x0_proc_read); } #else #define snd_intel8x0_proc_init(x) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 720635f0cb8..24703d75b65 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -59,7 +59,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH}," static int index = -2; /* Exclude the first card */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ -static int ac97_clock = 0; +static int ac97_clock; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for Intel i8x0 modemcard."); @@ -1092,7 +1092,7 @@ static void __devinit snd_intel8x0m_proc_init(struct intel8x0m * chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "intel8x0m", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_intel8x0m_proc_read); + snd_info_set_text_ops(entry, chip, snd_intel8x0m_proc_read); } #else /* !CONFIG_PROC_FS */ #define snd_intel8x0m_proc_init(chip) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index e39fad1a420..6e97932de34 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2085,7 +2085,7 @@ static void __devinit snd_korg1212_proc_init(struct snd_korg1212 *korg1212) struct snd_info_entry *entry; if (! snd_card_proc_new(korg1212->card, "korg1212", &entry)) - snd_info_set_text_ops(entry, korg1212, 1024, snd_korg1212_proc_read); + snd_info_set_text_ops(entry, korg1212, snd_korg1212_proc_read); } static int diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 1928e06b6d8..1c344fbd964 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2861,7 +2861,8 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) #if 0 /* TODO: not supported yet */ /* TODO enable MIDI IRQ and I/O */ err = snd_mpu401_uart_new(chip->card, 0, MPU401_HW_MPU401, - chip->iobase + MPU401_DATA_PORT, 1, + chip->iobase + MPU401_DATA_PORT, + MPU401_INFO_INTEGRATED, chip->irq, 0, &chip->rmidi); if (err < 0) printk(KERN_WARNING "maestro3: no MIDI support.\n"); diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 09cc0786495..366c4a7e65c 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1244,7 +1244,6 @@ static void __devinit snd_mixart_proc_init(struct snd_mixart *chip) /* text interface to read perf and temp meters */ if (! snd_card_proc_new(chip->card, "board_info", &entry)) { entry->private_data = chip; - entry->c.text.read_size = 1024; entry->c.text.read = snd_mixart_proc_read; } diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index dafa2235aba..8198884b51e 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1150,9 +1150,9 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "info", &entry)) - snd_info_set_text_ops(entry, chip, 1024, pcxhr_proc_info); + snd_info_set_text_ops(entry, chip, pcxhr_proc_info); if (! snd_card_proc_new(chip->card, "sync", &entry)) - snd_info_set_text_ops(entry, chip, 1024, pcxhr_proc_sync); + snd_info_set_text_ops(entry, chip, pcxhr_proc_sync); } /* end of proc interface */ diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index d8cc985d724..5618ec9740b 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1836,11 +1836,11 @@ static int snd_riptide_free(struct snd_riptide *chip) UNSET_GRESET(cif->hwport); kfree(chip->cif); } + if (chip->irq >= 0) + free_irq(chip->irq, chip); if (chip->fw_entry) release_firmware(chip->fw_entry); release_and_free_resource(chip->res_port); - if (chip->irq >= 0) - free_irq(chip->irq, chip); kfree(chip); return 0; } @@ -1992,7 +1992,7 @@ static void __devinit snd_riptide_proc_init(struct snd_riptide *chip) struct snd_info_entry *entry; if (!snd_card_proc_new(chip->card, "riptide", &entry)) - snd_info_set_text_ops(entry, chip, 4096, snd_riptide_proc_read); + snd_info_set_text_ops(entry, chip, snd_riptide_proc_read); } static int __devinit snd_riptide_mixer(struct snd_riptide *chip) diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 55b1d4838d9..2cb9fe98db2 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1368,18 +1368,18 @@ static int __devinit snd_rme32_create(struct rme32 * rme32) return err; rme32->port = pci_resource_start(rme32->pci, 0); - if (request_irq(pci->irq, snd_rme32_interrupt, SA_INTERRUPT | SA_SHIRQ, "RME32", (void *) rme32)) { - snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); - return -EBUSY; - } - rme32->irq = pci->irq; - if ((rme32->iobase = ioremap_nocache(rme32->port, RME32_IO_SIZE)) == 0) { snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme32->port, rme32->port + RME32_IO_SIZE - 1); return -ENOMEM; } + if (request_irq(pci->irq, snd_rme32_interrupt, SA_INTERRUPT | SA_SHIRQ, "RME32", (void *) rme32)) { + snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); + return -EBUSY; + } + rme32->irq = pci->irq; + /* read the card's revision number */ pci_read_config_byte(pci, 8, &rme32->rev); @@ -1578,7 +1578,7 @@ static void __devinit snd_rme32_proc_init(struct rme32 * rme32) struct snd_info_entry *entry; if (! snd_card_proc_new(rme32->card, "rme32", &entry)) - snd_info_set_text_ops(entry, rme32, 1024, snd_rme32_proc_read); + snd_info_set_text_ops(entry, rme32, snd_rme32_proc_read); } /* diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 3c1bc533d51..991cb18c14f 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1151,6 +1151,25 @@ static struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = { .mask = 0 }; +static void +rme96_set_buffer_size_constraint(struct rme96 *rme96, + struct snd_pcm_runtime *runtime) +{ + unsigned int size; + + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); + if ((size = rme96->playback_periodsize) != 0 || + (size = rme96->capture_periodsize) != 0) + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + size, size); + else + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + &hw_constraints_period_bytes); +} + static int snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream) { @@ -1180,8 +1199,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; } - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); + rme96_set_buffer_size_constraint(rme96, runtime); rme96->wcreg_spdif_stream = rme96->wcreg_spdif; rme96->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -1219,9 +1237,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream) rme96->capture_substream = substream; spin_unlock_irq(&rme96->lock); - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); - + rme96_set_buffer_size_constraint(rme96, runtime); return 0; } @@ -1254,8 +1270,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; } - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); + rme96_set_buffer_size_constraint(rme96, runtime); return 0; } @@ -1291,8 +1306,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream) rme96->capture_substream = substream; spin_unlock_irq(&rme96->lock); - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); + rme96_set_buffer_size_constraint(rme96, runtime); return 0; } @@ -1569,17 +1583,17 @@ snd_rme96_create(struct rme96 *rme96) return err; rme96->port = pci_resource_start(rme96->pci, 0); + if ((rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE)) == 0) { + snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme96->port, rme96->port + RME96_IO_SIZE - 1); + return -ENOMEM; + } + if (request_irq(pci->irq, snd_rme96_interrupt, SA_INTERRUPT|SA_SHIRQ, "RME96", (void *)rme96)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); return -EBUSY; } rme96->irq = pci->irq; - if ((rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE)) == 0) { - snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme96->port, rme96->port + RME96_IO_SIZE - 1); - return -ENOMEM; - } - /* read the card's revision number */ pci_read_config_byte(pci, 8, &rme96->rev); @@ -1805,7 +1819,7 @@ snd_rme96_proc_init(struct rme96 *rme96) struct snd_info_entry *entry; if (! snd_card_proc_new(rme96->card, "rme96", &entry)) - snd_info_set_text_ops(entry, rme96, 1024, snd_rme96_proc_read); + snd_info_set_text_ops(entry, rme96, snd_rme96_proc_read); } /* diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 61f82f0d5cc..eaf3c22449a 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -389,7 +389,7 @@ MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP}," /* use hotplug firmeare loader? */ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#ifndef HDSP_USE_HWDEP_LOADER +#if !defined(HDSP_USE_HWDEP_LOADER) && !defined(CONFIG_SND_HDSP) #define HDSP_FW_LOADER #endif #endif @@ -3169,9 +3169,10 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) char *clock_source; int x; - if (hdsp_check_for_iobox (hdsp)) + if (hdsp_check_for_iobox (hdsp)) { snd_iprintf(buffer, "No I/O box connected.\nPlease connect one and upload firmware.\n"); return; + } if (hdsp_check_for_firmware(hdsp, 0)) { if (hdsp->state & HDSP_FirmwareCached) { @@ -3470,7 +3471,7 @@ static void __devinit snd_hdsp_proc_init(struct hdsp *hdsp) struct snd_info_entry *entry; if (! snd_card_proc_new(hdsp->card, "hdsp", &entry)) - snd_info_set_text_ops(entry, hdsp, 1024, snd_hdsp_proc_read); + snd_info_set_text_ops(entry, hdsp, snd_hdsp_proc_read); } static void snd_hdsp_free_buffers(struct hdsp *hdsp) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 722b9e6ce54..bba1615504d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2489,7 +2489,7 @@ static void __devinit snd_hdspm_proc_init(struct hdspm * hdspm) struct snd_info_entry *entry; if (!snd_card_proc_new(hdspm->card, "hdspm", &entry)) - snd_info_set_text_ops(entry, hdspm, 1024, + snd_info_set_text_ops(entry, hdspm, snd_hdspm_proc_read); } diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 75d6406303d..3b945e8c1b1 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -41,7 +41,7 @@ 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 */ -static int precise_ptr[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; /* Enable precise pointer */ +static int precise_ptr[SNDRV_CARDS]; /* Enable precise pointer */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for RME Digi9652 (Hammerfall) soundcard."); @@ -1787,7 +1787,7 @@ static void __devinit snd_rme9652_proc_init(struct snd_rme9652 *rme9652) struct snd_info_entry *entry; if (! snd_card_proc_new(rme9652->card, "rme9652", &entry)) - snd_info_set_text_ops(entry, rme9652, 1024, snd_rme9652_proc_read); + snd_info_set_text_ops(entry, rme9652, snd_rme9652_proc_read); } static void snd_rme9652_free_buffers(struct snd_rme9652 *rme9652) diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 91f8bf3ae9f..dcf40294834 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -54,8 +54,8 @@ MODULE_SUPPORTED_DEVICE("{{S3,SonicVibes PCI}}"); 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 */ -static int reverb[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; -static int mge[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; +static int reverb[SNDRV_CARDS]; +static int mge[SNDRV_CARDS]; static unsigned int dmaio = 0x7a00; /* DDMA i/o address */ module_param_array(index, int, NULL, 0444); @@ -1144,7 +1144,7 @@ static void __devinit snd_sonicvibes_proc_init(struct sonicvibes * sonic) struct snd_info_entry *entry; if (! snd_card_proc_new(sonic->card, "sonicvibes", &entry)) - snd_info_set_text_ops(entry, sonic, 1024, snd_sonicvibes_proc_read); + snd_info_set_text_ops(entry, sonic, snd_sonicvibes_proc_read); } /* @@ -1456,7 +1456,7 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci, return err; } if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES, - sonic->midi_port, 1, + sonic->midi_port, MPU401_INFO_INTEGRATED, sonic->irq, 0, &midi_uart)) < 0) { snd_card_free(card); diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 9624a5f2b87..5629b7eba96 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -148,7 +148,8 @@ static int __devinit snd_trident_probe(struct pci_dev *pci, } if (trident->device != TRIDENT_DEVICE_ID_SI7018 && (err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE, - trident->midi_port, 1, + trident->midi_port, + MPU401_INFO_INTEGRATED, trident->irq, 0, &trident->rmidi)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 52178b8ad49..d99ed723775 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -306,6 +306,8 @@ void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice) outl(mask, TRID_REG(trident, reg)); } +EXPORT_SYMBOL(snd_trident_start_voice); + /*--------------------------------------------------------------------------- void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice) @@ -328,6 +330,8 @@ void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice) outl(mask, TRID_REG(trident, reg)); } +EXPORT_SYMBOL(snd_trident_stop_voice); + /*--------------------------------------------------------------------------- int snd_trident_allocate_pcm_channel(struct snd_trident *trident) @@ -502,6 +506,8 @@ void snd_trident_write_voice_regs(struct snd_trident * trident, #endif } +EXPORT_SYMBOL(snd_trident_write_voice_regs); + /*--------------------------------------------------------------------------- snd_trident_write_cso_reg @@ -3332,7 +3338,7 @@ static void __devinit snd_trident_proc_init(struct snd_trident * trident) if (trident->device == TRIDENT_DEVICE_ID_SI7018) s = "sis7018"; if (! snd_card_proc_new(trident->card, s, &entry)) - snd_info_set_text_ops(entry, trident, 1024, snd_trident_proc_read); + snd_info_set_text_ops(entry, trident, snd_trident_proc_read); } static int snd_trident_dev_free(struct snd_device *device) @@ -3884,6 +3890,8 @@ struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident, return NULL; } +EXPORT_SYMBOL(snd_trident_alloc_voice); + void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voice *voice) { unsigned long flags; @@ -3912,6 +3920,8 @@ void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voi private_free(voice); } +EXPORT_SYMBOL(snd_trident_free_voice); + static void snd_trident_clear_voices(struct snd_trident * trident, unsigned short v_min, unsigned short v_max) { unsigned int i, val, mask[2] = { 0, 0 }; @@ -3993,13 +4003,3 @@ int snd_trident_resume(struct pci_dev *pci) return 0; } #endif /* CONFIG_PM */ - -EXPORT_SYMBOL(snd_trident_alloc_voice); -EXPORT_SYMBOL(snd_trident_free_voice); -EXPORT_SYMBOL(snd_trident_start_voice); -EXPORT_SYMBOL(snd_trident_stop_voice); -EXPORT_SYMBOL(snd_trident_write_voice_regs); -/* trident_memory.c symbols */ -EXPORT_SYMBOL(snd_trident_synth_alloc); -EXPORT_SYMBOL(snd_trident_synth_free); -EXPORT_SYMBOL(snd_trident_synth_copy_from_user); diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index 46c6982c9e8..aff3f874131 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -349,6 +349,7 @@ snd_trident_synth_alloc(struct snd_trident *hw, unsigned int size) return blk; } +EXPORT_SYMBOL(snd_trident_synth_alloc); /* * free a synth sample area @@ -365,6 +366,7 @@ snd_trident_synth_free(struct snd_trident *hw, struct snd_util_memblk *blk) return 0; } +EXPORT_SYMBOL(snd_trident_synth_free); /* * reset TLB entry and free kernel page @@ -486,3 +488,4 @@ int snd_trident_synth_copy_from_user(struct snd_trident *trident, return 0; } +EXPORT_SYMBOL(snd_trident_synth_copy_from_user); diff --git a/sound/pci/trident/trident_synth.c b/sound/pci/trident/trident_synth.c index cc7af8bc55a..9b7dee84743 100644 --- a/sound/pci/trident/trident_synth.c +++ b/sound/pci/trident/trident_synth.c @@ -914,7 +914,9 @@ static int snd_trident_synth_create_port(struct snd_trident * trident, int idx) &callbacks, SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE, SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE | - SNDRV_SEQ_PORT_TYPE_SYNTH, + SNDRV_SEQ_PORT_TYPE_SYNTH | + SNDRV_SEQ_PORT_TYPE_HARDWARE | + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER, 16, 0, name); if (p->chset->port < 0) { diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 39daf62d2ba..2527bbd958c 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1775,6 +1775,12 @@ static struct ac97_quirk ac97_quirks[] = { .name = "Targa Traveller 811", .type = AC97_TUNE_HP_ONLY, }, + { + .subvendor = 0x161f, + .subdevice = 0x2032, + .name = "m680x", + .type = AC97_TUNE_HP_ONLY, /* http://launchpad.net/bugs/38546 */ + }, { } /* terminator */ }; @@ -1973,7 +1979,7 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip) pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, legacy_cfg); if (chip->mpu_res) { if (snd_mpu401_uart_new(chip->card, 0, MPU401_HW_VIA686A, - mpu_port, 1, + mpu_port, MPU401_INFO_INTEGRATED, chip->irq, 0, &chip->rmidi) < 0) { printk(KERN_WARNING "unable to initialize MPU-401" " at 0x%lx, skipping\n", mpu_port); @@ -2015,7 +2021,7 @@ static void __devinit snd_via82xx_proc_init(struct via82xx *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "via82xx", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_via82xx_proc_read); + snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read); } /* @@ -2365,7 +2371,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci, int revision) { .subvendor = 0x1462, .subdevice = 0x0470, .action = VIA_DXS_SRC }, /* MSI KT880 Delta-FSR */ { .subvendor = 0x1462, .subdevice = 0x3800, .action = VIA_DXS_ENABLE }, /* MSI KT266 */ { .subvendor = 0x1462, .subdevice = 0x5901, .action = VIA_DXS_NO_VRA }, /* MSI KT6 Delta-SR */ - { .subvendor = 0x1462, .subdevice = 0x7023, .action = VIA_DXS_NO_VRA }, /* MSI K8T Neo2-FI */ + { .subvendor = 0x1462, .subdevice = 0x7023, .action = VIA_DXS_SRC }, /* MSI K8T Neo2-FI */ { .subvendor = 0x1462, .subdevice = 0x7120, .action = VIA_DXS_ENABLE }, /* MSI KT4V */ { .subvendor = 0x1462, .subdevice = 0x7142, .action = VIA_DXS_ENABLE }, /* MSI K8MM-V */ { .subvendor = 0x1462, .subdevice = 0xb012, .action = VIA_DXS_SRC }, /* P4M800/VIA8237R */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index ef97e50cd6c..577a2b03759 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -929,7 +929,7 @@ static void __devinit snd_via82xx_proc_init(struct via82xx_modem *chip) struct snd_info_entry *entry; if (! snd_card_proc_new(chip->card, "via82xx", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_via82xx_proc_read); + snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read); } /* diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 65ebf5f1933..26aa775b7b6 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -308,7 +308,8 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci, } if (chip->mpu_res) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI, - mpu_port[dev], 1, + mpu_port[dev], + MPU401_INFO_INTEGRATED, pci->irq, 0, &chip->rawmidi)) < 0) { printk(KERN_WARNING "ymfpci: cannot initialize MPU401 at 0x%lx, skipping...\n", mpu_port[dev]); legacy_ctrl &= ~YMFPCI_LEGACY_MIEN; /* disable MPU401 irq */ diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 8ac5ab50b5c..f894752523b 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1919,7 +1919,7 @@ static int __devinit snd_ymfpci_proc_init(struct snd_card *card, struct snd_ymfp struct snd_info_entry *entry; if (! snd_card_proc_new(card, "ymfpci", &entry)) - snd_info_set_text_ops(entry, chip, 1024, snd_ymfpci_proc_read); + snd_info_set_text_ops(entry, chip, snd_ymfpci_proc_read); return 0; } |