diff options
author | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2011-01-19 11:22:54 +0000 |
---|---|---|
committer | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2011-01-19 11:22:54 +0000 |
commit | a1926d1745114789687ac029ae8c58944b7d2256 (patch) | |
tree | c303e75615e378451a80b97bfd2c1ba54029d9bb /sound | |
parent | 492e917635a0fa05439bb562fd51577efc9cef30 (diff) | |
parent | 52fc43f7c1c416b114e88ff39635c36e67ef15b6 (diff) |
Merge branch 'for-2.6.38' into for-2.6.39
Diffstat (limited to 'sound')
72 files changed, 3833 insertions, 1790 deletions
diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index a351dd0a09c7..2b50cbe6aca9 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -19,8 +19,8 @@ /* * Let drivers decide whether they want to support given codec from their - * probe method. Drivers have direct access to the struct snd_ac97 structure and may - * decide based on the id field amongst other things. + * probe method. Drivers have direct access to the struct snd_ac97 + * structure and may decide based on the id field amongst other things. */ static int ac97_bus_match(struct device *dev, struct device_driver *drv) { diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 91852e49910e..3687a6cc9881 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1114,7 +1114,6 @@ static int onyx_i2c_remove(struct i2c_client *client) of_node_put(onyx->codec.node); if (onyx->codec_info) kfree(onyx->codec_info); - i2c_set_clientdata(client, onyx); kfree(onyx); return 0; } diff --git a/sound/core/control.c b/sound/core/control.c index 45a818002d99..9ce00ed20fba 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1488,7 +1488,7 @@ int snd_ctl_create(struct snd_card *card) } /* - * Frequently used control callbacks + * Frequently used control callbacks/helpers */ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1513,3 +1513,29 @@ int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL(snd_ctl_boolean_stereo_info); + +/** + * snd_ctl_enum_info - fills the info structure for an enumerated control + * @info: the structure to be filled + * @channels: the number of the control's channels; often one + * @items: the number of control values; also the size of @names + * @names: an array containing the names of all control values + * + * Sets all required fields in @info to their appropriate values. + * If the control's accessibility is not the default (readable and writable), + * the caller has to fill @info->access. + */ +int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, + unsigned int items, const char *const names[]) +{ + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = channels; + info->value.enumerated.items = items; + if (info->value.enumerated.item >= items) + info->value.enumerated.item = items - 1; + strlcpy(info->value.enumerated.name, + names[info->value.enumerated.item], + sizeof(info->value.enumerated.name)); + return 0; +} +EXPORT_SYMBOL(snd_ctl_enum_info); diff --git a/sound/core/init.c b/sound/core/init.c index 57b792e2439a..3e65da21a08c 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -642,7 +642,7 @@ static struct device_attribute card_number_attrs = * external accesses. Thus, you should call this function at the end * of the initialization of the card. * - * Returns zero otherwise a negative error code if the registrain failed. + * Returns zero otherwise a negative error code if the registration failed. */ int snd_card_register(struct snd_card *card) { diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index b753ec661fcf..a2e4eb324699 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -453,8 +453,10 @@ static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm, } else { *params = *save; max = snd_pcm_hw_param_max(pcm, params, var, max, &maxdir); - if (max < 0) + if (max < 0) { + kfree(save); return max; + } last = 1; } _end: diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index b75db8e9cc0f..a82e3756a72d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -373,6 +373,27 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, (unsigned long)new_hw_ptr, (unsigned long)runtime->hw_ptr_base); } + + if (runtime->no_period_wakeup) { + /* + * Without regular period interrupts, we have to check + * the elapsed time to detect xruns. + */ + jdelta = jiffies - runtime->hw_ptr_jiffies; + if (jdelta < runtime->hw_ptr_buffer_jiffies / 2) + goto no_delta_check; + hdelta = jdelta - delta * HZ / runtime->rate; + while (hdelta > runtime->hw_ptr_buffer_jiffies / 2 + 1) { + delta += runtime->buffer_size; + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + hdelta -= runtime->hw_ptr_buffer_jiffies; + } + goto no_delta_check; + } + /* something must be really wrong */ if (delta >= runtime->buffer_size + runtime->period_size) { hw_ptr_error(substream, @@ -442,6 +463,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, (long)old_hw_ptr); } + no_delta_check: if (runtime->status->hw_ptr == new_hw_ptr) return 0; @@ -1070,8 +1092,10 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, struct snd_pcm_hw_rule *new; unsigned int new_rules = constrs->rules_all + 16; new = kcalloc(new_rules, sizeof(*c), GFP_KERNEL); - if (!new) + if (!new) { + va_end(args); return -ENOMEM; + } if (constrs->rules) { memcpy(new, constrs->rules, constrs->rules_num * sizeof(*c)); @@ -1087,8 +1111,10 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, c->private = private; k = 0; while (1) { - if (snd_BUG_ON(k >= ARRAY_SIZE(c->deps))) + if (snd_BUG_ON(k >= ARRAY_SIZE(c->deps))) { + va_end(args); return -EINVAL; + } c->deps[k++] = dep; if (dep < 0) break; @@ -1097,7 +1123,7 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, constrs->rules_num++; va_end(args); return 0; -} +} EXPORT_SYMBOL(snd_pcm_hw_rule_add); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e82c1f97d99e..4be45e7be8ad 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -422,6 +422,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->info = params->info; runtime->rate_num = params->rate_num; runtime->rate_den = params->rate_den; + runtime->no_period_wakeup = + (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && + (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); bits = snd_pcm_format_physical_width(runtime->format); runtime->sample_bits = bits; @@ -984,7 +987,7 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push) if (push) snd_pcm_update_hw_ptr(substream); /* The jiffies check in snd_pcm_update_hw_ptr*() is done by - * a delta betwen the current jiffies, this gives a large enough + * a delta between the current jiffies, this gives a large enough * delta, effectively to skip the check once. */ substream->runtime->hw_ptr_jiffies = jiffies - HZ * 1000; diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index bf09a5ad1865..119fddb6fc99 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -32,6 +32,7 @@ #include "seq_timer.h" #include "seq_system.h" #include "seq_info.h" +#include <sound/minors.h> #include <sound/seq_device.h> #if defined(CONFIG_SND_SEQ_DUMMY_MODULE) @@ -73,6 +74,9 @@ MODULE_PARM_DESC(seq_default_timer_subdevice, "The default timer subdevice numbe module_param(seq_default_timer_resolution, int, 0644); MODULE_PARM_DESC(seq_default_timer_resolution, "The default timer resolution in Hz."); +MODULE_ALIAS_CHARDEV(CONFIG_SND_MAJOR, SNDRV_MINOR_SEQUENCER); +MODULE_ALIAS("devname:snd/seq"); + /* * INIT PART */ diff --git a/sound/core/sound.c b/sound/core/sound.c index 66691fe437e6..1c7a3efe1778 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -188,14 +188,22 @@ static const struct file_operations snd_fops = }; #ifdef CONFIG_SND_DYNAMIC_MINORS -static int snd_find_free_minor(void) +static int snd_find_free_minor(int type) { int minor; + /* static minors for module auto loading */ + if (type == SNDRV_DEVICE_TYPE_SEQUENCER) + return SNDRV_MINOR_SEQUENCER; + if (type == SNDRV_DEVICE_TYPE_TIMER) + return SNDRV_MINOR_TIMER; + for (minor = 0; minor < ARRAY_SIZE(snd_minors); ++minor) { - /* skip minors still used statically for autoloading devices */ - if (SNDRV_MINOR_DEVICE(minor) == SNDRV_MINOR_CONTROL || - minor == SNDRV_MINOR_SEQUENCER) + /* skip static minors still used for module auto loading */ + if (SNDRV_MINOR_DEVICE(minor) == SNDRV_MINOR_CONTROL) + continue; + if (minor == SNDRV_MINOR_SEQUENCER || + minor == SNDRV_MINOR_TIMER) continue; if (!snd_minors[minor]) return minor; @@ -269,7 +277,7 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, preg->private_data = private_data; mutex_lock(&sound_mutex); #ifdef CONFIG_SND_DYNAMIC_MINORS - minor = snd_find_free_minor(); + minor = snd_find_free_minor(type); #else minor = snd_kernel_minor(type, card, dev); if (minor >= 0 && snd_minors[minor]) diff --git a/sound/core/timer.c b/sound/core/timer.c index 13afb60999b9..ed016329e911 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -34,8 +34,8 @@ #include <sound/initval.h> #include <linux/kmod.h> -#if defined(CONFIG_SND_HPET) || defined(CONFIG_SND_HPET_MODULE) -#define DEFAULT_TIMER_LIMIT 3 +#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE) +#define DEFAULT_TIMER_LIMIT 4 #elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE) #define DEFAULT_TIMER_LIMIT 2 #else @@ -52,6 +52,9 @@ MODULE_PARM_DESC(timer_limit, "Maximum global timers in system."); module_param(timer_tstamp_monotonic, int, 0444); MODULE_PARM_DESC(timer_tstamp_monotonic, "Use posix monotonic clock source for timestamps (default)."); +MODULE_ALIAS_CHARDEV(CONFIG_SND_MAJOR, SNDRV_MINOR_TIMER); +MODULE_ALIAS("devname:snd/timer"); + struct snd_timer_user { struct snd_timer_instance *timeri; int tread; /* enhanced read with timestamps and events */ diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index a1282c1c0591..5cfcb908c430 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1143,8 +1143,8 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, (resource->start) + 1); if (ml403_ac97cr->port == NULL) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " - "unable to remap memory region (%x to %x)\n", - resource->start, resource->end); + "unable to remap memory region (%pR)\n", + resource); snd_ml403_ac97cr_free(ml403_ac97cr); return -EBUSY; } diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 265abcce9dba..9b915e27b5bd 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -264,7 +264,7 @@ static int __devinit snd_opl3sa2_detect(struct snd_card *card) snd_printd("OPL3-SA [0x%lx] detect (1) = 0x%x (0x%x)\n", port, tmp, tmp1); return -ENODEV; } - /* try if the MIC register is accesible */ + /* try if the MIC register is accessible */ tmp = snd_opl3sa2_read(chip, OPL3SA2_MIC); snd_opl3sa2_write(chip, OPL3SA2_MIC, 0x8a); if (((tmp1 = snd_opl3sa2_read(chip, OPL3SA2_MIC)) & 0x9f) != 0x8a) { diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 46c0d03dbecc..fcb14a099822 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -87,7 +87,7 @@ int *load_mixer_volumes(char *name, int *levels, int present) int i, n; for (i = 0; i < num_mixer_volumes; i++) { - if (strcmp(name, mixer_vols[i].name) == 0) { + if (strncmp(name, mixer_vols[i].name, 32) == 0) { if (present) mixer_vols[i].num = i; return mixer_vols[i].levels; @@ -99,7 +99,7 @@ int *load_mixer_volumes(char *name, int *levels, int present) } n = num_mixer_volumes++; - strcpy(mixer_vols[n].name, name); + strncpy(mixer_vols[n].name, name, 32); if (present) mixer_vols[n].num = n; diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 12e34653b8a8..9823d59d7ad7 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -209,7 +209,7 @@ config SND_OXYGEN_LIB tristate config SND_OXYGEN - tristate "C-Media 8788 (Oxygen)" + tristate "C-Media 8786, 8787, 8788 (Oxygen)" select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART @@ -217,13 +217,18 @@ config SND_OXYGEN Say Y here to include support for sound cards based on the C-Media CMI8788 (Oxygen HD Audio) chip: * Asound A-8788 + * Asus Xonar DG * AuzenTech X-Meridian + * AuzenTech X-Meridian 2G * Bgears b-Enspirer * Club3D Theatron DTS * HT-Omega Claro (plus) * HT-Omega Claro halo (XT) + * Kuroutoshikou CMI8787-HG2PCI * Razer Barracuda AC-1 * Sondigo Inferno + * TempoTec/MediaTek HiFier Fantasia + * TempoTec/MediaTek HiFier Serenade To compile this driver as a module, choose M here: the module will be called snd-oxygen. @@ -578,18 +583,6 @@ config SND_HDSPM To compile this driver as a module, choose M here: the module will be called snd-hdspm. -config SND_HIFIER - tristate "TempoTec HiFier Fantasia" - select SND_OXYGEN_LIB - select SND_PCM - select SND_MPU401_UART - help - Say Y here to include support for the MediaTek/TempoTec HiFier - Fantasia sound card. - - To compile this driver as a module, choose M here: the module - will be called snd-hifier. - config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" select SND_MPU401_UART @@ -826,8 +819,8 @@ config SND_VIRTUOSO Say Y here to include support for sound cards based on the Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), and Essence STX. - Support for the HDAV1.3 (Deluxe) is incomplete; for the - HDAV1.3 Slim and Xense, missing. + Support for the HDAV1.3 (Deluxe) and HDAV1.3 Slim is experimental; + for the Xense, missing. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 0fc614ce16c1..cb62d178b3e0 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1961,7 +1961,7 @@ static int snd_ac97_dev_disconnect(struct snd_device *device) } /* build_ops to do nothing */ -static struct snd_ac97_build_ops null_build_ops; +static const struct snd_ac97_build_ops null_build_ops; #ifdef CONFIG_SND_AC97_POWER_SAVE static void do_update_power(struct work_struct *work) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index e68c98ef4041..bf47574ca1f0 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -371,7 +371,7 @@ static int patch_yamaha_ymf743_build_spdif(struct snd_ac97 *ac97) return 0; } -static struct snd_ac97_build_ops patch_yamaha_ymf743_ops = { +static const struct snd_ac97_build_ops patch_yamaha_ymf743_ops = { .build_spdif = patch_yamaha_ymf743_build_spdif, .build_3d = patch_yamaha_ymf7x3_3d, }; @@ -455,7 +455,7 @@ static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = { +static const struct snd_ac97_build_ops patch_yamaha_ymf753_ops = { .build_3d = patch_yamaha_ymf7x3_3d, .build_post_spdif = patch_yamaha_ymf753_post_spdif }; @@ -502,7 +502,7 @@ static int patch_wolfson_wm9703_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_wolfson_wm9703_ops = { +static const struct snd_ac97_build_ops patch_wolfson_wm9703_ops = { .build_specific = patch_wolfson_wm9703_specific, }; @@ -533,7 +533,7 @@ static int patch_wolfson_wm9704_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_wolfson_wm9704_ops = { +static const struct snd_ac97_build_ops patch_wolfson_wm9704_ops = { .build_specific = patch_wolfson_wm9704_specific, }; @@ -677,7 +677,7 @@ static int patch_wolfson_wm9711_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_wolfson_wm9711_ops = { +static const struct snd_ac97_build_ops patch_wolfson_wm9711_ops = { .build_specific = patch_wolfson_wm9711_specific, }; @@ -871,7 +871,7 @@ static void patch_wolfson_wm9713_resume (struct snd_ac97 * ac97) } #endif -static struct snd_ac97_build_ops patch_wolfson_wm9713_ops = { +static const struct snd_ac97_build_ops patch_wolfson_wm9713_ops = { .build_specific = patch_wolfson_wm9713_specific, .build_3d = patch_wolfson_wm9713_3d, #ifdef CONFIG_PM @@ -976,7 +976,7 @@ static int patch_sigmatel_stac97xx_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_sigmatel_stac9700_ops = { +static const struct snd_ac97_build_ops patch_sigmatel_stac9700_ops = { .build_3d = patch_sigmatel_stac9700_3d, .build_specific = patch_sigmatel_stac97xx_specific }; @@ -1023,7 +1023,7 @@ static int patch_sigmatel_stac9708_specific(struct snd_ac97 *ac97) return patch_sigmatel_stac97xx_specific(ac97); } -static struct snd_ac97_build_ops patch_sigmatel_stac9708_ops = { +static const struct snd_ac97_build_ops patch_sigmatel_stac9708_ops = { .build_3d = patch_sigmatel_stac9708_3d, .build_specific = patch_sigmatel_stac9708_specific }; @@ -1252,7 +1252,7 @@ static int patch_sigmatel_stac9758_specific(struct snd_ac97 *ac97) return 0; } -static struct snd_ac97_build_ops patch_sigmatel_stac9758_ops = { +static const struct snd_ac97_build_ops patch_sigmatel_stac9758_ops = { .build_3d = patch_sigmatel_stac9700_3d, .build_specific = patch_sigmatel_stac9758_specific }; @@ -1327,7 +1327,7 @@ static int patch_cirrus_build_spdif(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_cirrus_ops = { +static const struct snd_ac97_build_ops patch_cirrus_ops = { .build_spdif = patch_cirrus_build_spdif }; @@ -1384,7 +1384,7 @@ static int patch_conexant_build_spdif(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_conexant_ops = { +static const struct snd_ac97_build_ops patch_conexant_ops = { .build_spdif = patch_conexant_build_spdif }; @@ -1560,7 +1560,7 @@ static void patch_ad1881_chained(struct snd_ac97 * ac97, int unchained_idx, int } } -static struct snd_ac97_build_ops patch_ad1881_build_ops = { +static const struct snd_ac97_build_ops patch_ad1881_build_ops = { #ifdef CONFIG_PM .resume = ad18xx_resume #endif @@ -1647,7 +1647,7 @@ static int patch_ad1885_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_ad1885_build_ops = { +static const struct snd_ac97_build_ops patch_ad1885_build_ops = { .build_specific = &patch_ad1885_specific, #ifdef CONFIG_PM .resume = ad18xx_resume @@ -1674,7 +1674,7 @@ static int patch_ad1886_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_ad1886_build_ops = { +static const struct snd_ac97_build_ops patch_ad1886_build_ops = { .build_specific = &patch_ad1886_specific, #ifdef CONFIG_PM .resume = ad18xx_resume @@ -1881,7 +1881,7 @@ static int patch_ad1981a_specific(struct snd_ac97 * ac97) ARRAY_SIZE(snd_ac97_ad1981x_jack_sense)); } -static struct snd_ac97_build_ops patch_ad1981a_build_ops = { +static const struct snd_ac97_build_ops patch_ad1981a_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1981a_specific, #ifdef CONFIG_PM @@ -1936,7 +1936,7 @@ static int patch_ad1981b_specific(struct snd_ac97 *ac97) ARRAY_SIZE(snd_ac97_ad1981x_jack_sense)); } -static struct snd_ac97_build_ops patch_ad1981b_build_ops = { +static const struct snd_ac97_build_ops patch_ad1981b_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1981b_specific, #ifdef CONFIG_PM @@ -2075,7 +2075,7 @@ static int patch_ad1888_specific(struct snd_ac97 *ac97) return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls)); } -static struct snd_ac97_build_ops patch_ad1888_build_ops = { +static const struct snd_ac97_build_ops patch_ad1888_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1888_specific, #ifdef CONFIG_PM @@ -2124,7 +2124,7 @@ static int patch_ad1980_specific(struct snd_ac97 *ac97) return patch_build_controls(ac97, &snd_ac97_ad198x_2cmic, 1); } -static struct snd_ac97_build_ops patch_ad1980_build_ops = { +static const struct snd_ac97_build_ops patch_ad1980_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1980_specific, #ifdef CONFIG_PM @@ -2239,7 +2239,7 @@ static int patch_ad1985_specific(struct snd_ac97 *ac97) ARRAY_SIZE(snd_ac97_ad1985_controls)); } -static struct snd_ac97_build_ops patch_ad1985_build_ops = { +static const struct snd_ac97_build_ops patch_ad1985_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1985_specific, #ifdef CONFIG_PM @@ -2531,7 +2531,7 @@ static int patch_ad1986_specific(struct snd_ac97 *ac97) ARRAY_SIZE(snd_ac97_ad1985_controls)); } -static struct snd_ac97_build_ops patch_ad1986_build_ops = { +static const struct snd_ac97_build_ops patch_ad1986_build_ops = { .build_post_spdif = patch_ad198x_post_spdif, .build_specific = patch_ad1986_specific, #ifdef CONFIG_PM @@ -2636,7 +2636,7 @@ static int patch_alc650_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_alc650_ops = { +static const struct snd_ac97_build_ops patch_alc650_ops = { .build_specific = patch_alc650_specific, .update_jacks = alc650_update_jacks }; @@ -2788,7 +2788,7 @@ static int patch_alc655_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_alc655_ops = { +static const struct snd_ac97_build_ops patch_alc655_ops = { .build_specific = patch_alc655_specific, .update_jacks = alc655_update_jacks }; @@ -2900,7 +2900,7 @@ static int patch_alc850_specific(struct snd_ac97 *ac97) return 0; } -static struct snd_ac97_build_ops patch_alc850_ops = { +static const struct snd_ac97_build_ops patch_alc850_ops = { .build_specific = patch_alc850_specific, .update_jacks = alc850_update_jacks }; @@ -2962,7 +2962,7 @@ static int patch_cm9738_specific(struct snd_ac97 * ac97) return patch_build_controls(ac97, snd_ac97_cm9738_controls, ARRAY_SIZE(snd_ac97_cm9738_controls)); } -static struct snd_ac97_build_ops patch_cm9738_ops = { +static const struct snd_ac97_build_ops patch_cm9738_ops = { .build_specific = patch_cm9738_specific, .update_jacks = cm9738_update_jacks }; @@ -3053,7 +3053,7 @@ static int patch_cm9739_post_spdif(struct snd_ac97 * ac97) return patch_build_controls(ac97, snd_ac97_cm9739_controls_spdif, ARRAY_SIZE(snd_ac97_cm9739_controls_spdif)); } -static struct snd_ac97_build_ops patch_cm9739_ops = { +static const struct snd_ac97_build_ops patch_cm9739_ops = { .build_specific = patch_cm9739_specific, .build_post_spdif = patch_cm9739_post_spdif, .update_jacks = cm9739_update_jacks @@ -3227,7 +3227,7 @@ static int patch_cm9761_specific(struct snd_ac97 * ac97) return patch_build_controls(ac97, snd_ac97_cm9761_controls, ARRAY_SIZE(snd_ac97_cm9761_controls)); } -static struct snd_ac97_build_ops patch_cm9761_ops = { +static const struct snd_ac97_build_ops patch_cm9761_ops = { .build_specific = patch_cm9761_specific, .build_post_spdif = patch_cm9761_post_spdif, .update_jacks = cm9761_update_jacks @@ -3323,7 +3323,7 @@ static int patch_cm9780_specific(struct snd_ac97 *ac97) return patch_build_controls(ac97, cm9780_controls, ARRAY_SIZE(cm9780_controls)); } -static struct snd_ac97_build_ops patch_cm9780_ops = { +static const struct snd_ac97_build_ops patch_cm9780_ops = { .build_specific = patch_cm9780_specific, .build_post_spdif = patch_cm9761_post_spdif /* identical with CM9761 */ }; @@ -3443,7 +3443,7 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_vt1616_ops = { +static const struct snd_ac97_build_ops patch_vt1616_ops = { .build_specific = patch_vt1616_specific }; @@ -3797,7 +3797,7 @@ static int patch_it2646_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_it2646_ops = { +static const struct snd_ac97_build_ops patch_it2646_ops = { .build_specific = patch_it2646_specific, .update_jacks = it2646_update_jacks }; @@ -3831,7 +3831,7 @@ static int patch_si3036_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_si3036_ops = { +static const struct snd_ac97_build_ops patch_si3036_ops = { .build_specific = patch_si3036_specific, }; @@ -3898,7 +3898,7 @@ static int patch_ucb1400_specific(struct snd_ac97 * ac97) return 0; } -static struct snd_ac97_build_ops patch_ucb1400_ops = { +static const struct snd_ac97_build_ops patch_ucb1400_ops = { .build_specific = patch_ucb1400_specific, }; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index b9d2f202cf9b..5439d662d104 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -42,11 +42,7 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = { .rate_min = 5000, .rate_max = 48000, .channels_min = 1, -#ifdef CHIP_AU8830 - .channels_max = 4, -#else .channels_max = 2, -#endif .buffer_bytes_max = 0x10000, .period_bytes_min = 0x1, .period_bytes_max = 0x1000, @@ -115,6 +111,17 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_wt = { .periods_max = 64, }; #endif +#ifdef CHIP_AU8830 +static unsigned int au8830_channels[3] = { + 1, 2, 4, +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_au8830_channels = { + .count = ARRAY_SIZE(au8830_channels), + .list = au8830_channels, + .mask = 0, +}; +#endif /* open callback */ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) { @@ -156,6 +163,15 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB || VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_I2S) runtime->hw = snd_vortex_playback_hw_adb; +#ifdef CHIP_AU8830 + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { + runtime->hw.channels_max = 4; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &hw_constraints_au8830_channels); + } +#endif substream->runtime->private_data = NULL; } #ifndef CHIP_AU8810 diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 2f3cacbd5528..6117595fc075 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1,6 +1,6 @@ /* * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005 - 2009 by Andreas Mohr <andi AT lisas.de> + * Copyright (C) 2002, 2005 - 2010 by Andreas Mohr <andi AT lisas.de> * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -175,6 +175,7 @@ #include <asm/io.h> #include <linux/init.h> +#include <linux/bug.h> /* WARN_ONCE */ #include <linux/pci.h> #include <linux/delay.h> #include <linux/slab.h> @@ -201,14 +202,15 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); /* === Debug settings === Further diagnostic functionality than the settings below - does not need to be provided, since one can easily write a bash script + does not need to be provided, since one can easily write a POSIX shell script to dump the card's I/O ports (those listed in lspci -v -v): - function dump() + dump() { local descr=$1; local addr=$2; local count=$3 echo "${descr}: ${count} @ ${addr}:" - dd if=/dev/port skip=$[${addr}] count=${count} bs=1 2>/dev/null| hexdump -C + dd if=/dev/port skip=`printf %d ${addr}` count=${count} bs=1 \ + 2>/dev/null| hexdump -C } and then use something like "dump joy200 0x200 8", "dump mpu388 0x388 4", "dump joy 0xb400 8", @@ -216,14 +218,14 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); possibly within a "while true; do ... sleep 1; done" loop. Tweaking ports could be done using VALSTRING="`printf "%02x" $value`" - printf "\x""$VALSTRING"|dd of=/dev/port seek=$[${addr}] bs=1 2>/dev/null + printf "\x""$VALSTRING"|dd of=/dev/port seek=`printf %d ${addr}` bs=1 \ + 2>/dev/null */ #define DEBUG_MISC 0 #define DEBUG_CALLS 0 #define DEBUG_MIXER 0 #define DEBUG_CODEC 0 -#define DEBUG_IO 0 #define DEBUG_TIMER 0 #define DEBUG_GAME 0 #define DEBUG_PM 0 @@ -291,19 +293,23 @@ static int seqtimer_scaling = 128; module_param(seqtimer_scaling, int, 0444); MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128."); -struct snd_azf3328_codec_data { - unsigned long io_base; - struct snd_pcm_substream *substream; - bool running; - const char *name; -}; - enum snd_azf3328_codec_type { + /* warning: fixed indices (also used for bitmask checks!) */ AZF_CODEC_PLAYBACK = 0, AZF_CODEC_CAPTURE = 1, AZF_CODEC_I2S_OUT = 2, }; +struct snd_azf3328_codec_data { + unsigned long io_base; /* keep first! (avoid offset calc) */ + unsigned int dma_base; /* helper to avoid an indirection in hotpath */ + spinlock_t *lock; /* TODO: convert to our own per-codec lock member */ + struct snd_pcm_substream *substream; + bool running; + enum snd_azf3328_codec_type type; + const char *name; +}; + struct snd_azf3328 { /* often-used fields towards beginning, then grouped */ @@ -362,6 +368,9 @@ MODULE_DEVICE_TABLE(pci, snd_azf3328_ids); static int snd_azf3328_io_reg_setb(unsigned reg, u8 mask, bool do_set) { + /* Well, strictly spoken, the inb/outb sequence isn't atomic + and would need locking. However we currently don't care + since it potentially complicates matters. */ u8 prev = inb(reg), new; new = (do_set) ? (prev|mask) : (prev & ~mask); @@ -413,6 +422,21 @@ snd_azf3328_codec_outl(const struct snd_azf3328_codec_data *codec, outl(value, codec->io_base + reg); } +static inline void +snd_azf3328_codec_outl_multi(const struct snd_azf3328_codec_data *codec, + unsigned reg, const void *buffer, int count +) +{ + unsigned long addr = codec->io_base + reg; + if (count) { + const u32 *buf = buffer; + do { + outl(*buf++, addr); + addr += 4; + } while (--count); + } +} + static inline u32 snd_azf3328_codec_inl(const struct snd_azf3328_codec_data *codec, unsigned reg) { @@ -943,38 +967,43 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream) } static void -snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, - enum snd_azf3328_codec_type codec_type, +snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, enum azf_freq_t bitrate, unsigned int format_width, unsigned int channels ) { unsigned long flags; - const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; u16 val = 0xff00; + u8 freq = 0; snd_azf3328_dbgcallenter(); switch (bitrate) { - case AZF_FREQ_4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; - case AZF_FREQ_4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; - case AZF_FREQ_5512: - /* the AZF3328 names it "5510" for some strange reason */ - val |= SOUNDFORMAT_FREQ_5510; break; - case AZF_FREQ_6620: val |= SOUNDFORMAT_FREQ_6620; break; - case AZF_FREQ_8000: val |= SOUNDFORMAT_FREQ_8000; break; - case AZF_FREQ_9600: val |= SOUNDFORMAT_FREQ_9600; break; - case AZF_FREQ_11025: val |= SOUNDFORMAT_FREQ_11025; break; - case AZF_FREQ_13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; - case AZF_FREQ_16000: val |= SOUNDFORMAT_FREQ_16000; break; - case AZF_FREQ_22050: val |= SOUNDFORMAT_FREQ_22050; break; - case AZF_FREQ_32000: val |= SOUNDFORMAT_FREQ_32000; break; +#define AZF_FMT_XLATE(in_freq, out_bits) \ + do { \ + case AZF_FREQ_ ## in_freq: \ + freq = SOUNDFORMAT_FREQ_ ## out_bits; \ + break; \ + } while (0); + AZF_FMT_XLATE(4000, SUSPECTED_4000) + AZF_FMT_XLATE(4800, SUSPECTED_4800) + /* the AZF3328 names it "5510" for some strange reason: */ + AZF_FMT_XLATE(5512, 5510) + AZF_FMT_XLATE(6620, 6620) + AZF_FMT_XLATE(8000, 8000) + AZF_FMT_XLATE(9600, 9600) + AZF_FMT_XLATE(11025, 11025) + AZF_FMT_XLATE(13240, SUSPECTED_13240) + AZF_FMT_XLATE(16000, 16000) + AZF_FMT_XLATE(22050, 22050) + AZF_FMT_XLATE(32000, 32000) default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); /* fall-through */ - case AZF_FREQ_44100: val |= SOUNDFORMAT_FREQ_44100; break; - case AZF_FREQ_48000: val |= SOUNDFORMAT_FREQ_48000; break; - case AZF_FREQ_66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; + AZF_FMT_XLATE(44100, 44100) + AZF_FMT_XLATE(48000, 48000) + AZF_FMT_XLATE(66200, SUSPECTED_66200) +#undef AZF_FMT_XLATE } /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */ /* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */ @@ -986,13 +1015,15 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, /* val = 0xff0d; 41m23.135s (5523,600Hz; -> 5512Hz???) */ /* val = 0xff0e; 28m30.777s (8017Hz; -> 8000Hz???) */ + val |= freq; + if (channels == 2) val |= SOUNDFORMAT_FLAG_2CHANNELS; if (format_width == 16) val |= SOUNDFORMAT_FLAG_16BIT; - spin_lock_irqsave(&chip->reg_lock, flags); + spin_lock_irqsave(codec->lock, flags); /* set bitrate/format */ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_SOUNDFORMAT, val); @@ -1004,7 +1035,8 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, * (FIXME: yes, it works, but what exactly am I doing here?? :) * FIXME: does this have some side effects for full-duplex * or other dramatic side effects? */ - if (codec_type == AZF_CODEC_PLAYBACK) /* only do it for playback */ + /* do it for non-capture codecs only */ + if (codec->type != AZF_CODEC_CAPTURE) snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS) | DMA_RUN_SOMETHING1 | @@ -1014,20 +1046,19 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, DMA_SOMETHING_ELSE ); - spin_unlock_irqrestore(&chip->reg_lock, flags); + spin_unlock_irqrestore(codec->lock, flags); snd_azf3328_dbgcallleave(); } static inline void -snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, - enum snd_azf3328_codec_type codec_type +snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328_codec_data *codec ) { /* choose lowest frequency for low power consumption. * While this will cause louder noise due to rather coarse frequency, * it should never matter since output should always * get disabled properly when idle anyway. */ - snd_azf3328_codec_setfmt(chip, codec_type, AZF_FREQ_4000, 8, 1); + snd_azf3328_codec_setfmt(codec, AZF_FREQ_4000, 8, 1); } static void @@ -1101,69 +1132,87 @@ snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, /* ...and adjust clock, too * (reduce noise and power consumption) */ if (!enable) - snd_azf3328_codec_setfmt_lowpower( - chip, - codec_type - ); + snd_azf3328_codec_setfmt_lowpower(codec); codec->running = enable; } } static void -snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip, - enum snd_azf3328_codec_type codec_type, +snd_azf3328_codec_setdmaa(struct snd_azf3328_codec_data *codec, unsigned long addr, - unsigned int count, - unsigned int size + unsigned int period_bytes, + unsigned int buffer_bytes ) { - const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; snd_azf3328_dbgcallenter(); + WARN_ONCE(period_bytes & 1, "odd period length!?\n"); + WARN_ONCE(buffer_bytes != 2 * period_bytes, + "missed our input expectations! %u vs. %u\n", + buffer_bytes, period_bytes); if (!codec->running) { /* AZF3328 uses a two buffer pointer DMA transfer approach */ - unsigned long flags, addr_area2; + unsigned long flags; /* width 32bit (prevent overflow): */ - u32 count_areas, lengths; + u32 area_length; + struct codec_setup_io { + u32 dma_start_1; + u32 dma_start_2; + u32 dma_lengths; + } __attribute__((packed)) setup_io; + + area_length = buffer_bytes/2; + + setup_io.dma_start_1 = addr; + setup_io.dma_start_2 = addr+area_length; - count_areas = size/2; - addr_area2 = addr+count_areas; - snd_azf3328_dbgcodec("setdma: buffers %08lx[%u] / %08lx[%u]\n", - addr, count_areas, addr_area2, count_areas); + snd_azf3328_dbgcodec( + "setdma: buffers %08x[%u] / %08x[%u], %u, %u\n", + setup_io.dma_start_1, area_length, + setup_io.dma_start_2, area_length, + period_bytes, buffer_bytes); - count_areas--; /* max. index */ + /* Hmm, are we really supposed to decrement this by 1?? + Most definitely certainly not: configuring full length does + work properly (i.e. likely better), and BTW we + violated possibly differing frame sizes with this... + + area_length--; |* max. index *| + */ /* build combined I/O buffer length word */ - lengths = (count_areas << 16) | (count_areas); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_1, addr); - snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_2, - addr_area2); - snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_LENGTHS, - lengths); - spin_unlock_irqrestore(&chip->reg_lock, flags); + setup_io.dma_lengths = (area_length << 16) | (area_length); + + spin_lock_irqsave(codec->lock, flags); + snd_azf3328_codec_outl_multi( + codec, IDX_IO_CODEC_DMA_START_1, &setup_io, 3 + ); + spin_unlock_irqrestore(codec->lock, flags); } snd_azf3328_dbgcallleave(); } static int -snd_azf3328_codec_prepare(struct snd_pcm_substream *substream) +snd_azf3328_pcm_prepare(struct snd_pcm_substream *substream) { -#if 0 - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_azf3328_codec_data *codec = runtime->private_data; +#if 0 unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); #endif snd_azf3328_dbgcallenter(); + + codec->dma_base = runtime->dma_addr; + #if 0 - snd_azf3328_codec_setfmt(chip, AZF_CODEC_..., + snd_azf3328_codec_setfmt(codec, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_codec_setdmaa(chip, AZF_CODEC_..., + snd_azf3328_codec_setdmaa(codec, runtime->dma_addr, count, size); #endif snd_azf3328_dbgcallleave(); @@ -1171,24 +1220,23 @@ snd_azf3328_codec_prepare(struct snd_pcm_substream *substream) } static int -snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, - struct snd_pcm_substream *substream, int cmd) +snd_azf3328_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_azf3328_codec_data *codec = runtime->private_data; int result = 0; u16 flags1; bool previously_muted = 0; - bool is_playback_codec = (AZF_CODEC_PLAYBACK == codec_type); + bool is_main_mixer_playback_codec = (AZF_CODEC_PLAYBACK == codec->type); - snd_azf3328_dbgcalls("snd_azf3328_codec_trigger cmd %d\n", cmd); + snd_azf3328_dbgcalls("snd_azf3328_pcm_trigger cmd %d\n", cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_START: snd_azf3328_dbgcodec("START %s\n", codec->name); - if (is_playback_codec) { + if (is_main_mixer_playback_codec) { /* mute WaveOut (avoid clicking during setup) */ previously_muted = snd_azf3328_mixer_set_mute( @@ -1196,12 +1244,12 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, ); } - snd_azf3328_codec_setfmt(chip, codec_type, + snd_azf3328_codec_setfmt(codec, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); /* first, remember current value: */ flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS); @@ -1211,14 +1259,14 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(codec, IDX_IO_CODEC_IRQTYPE, 0xffff); - spin_unlock(&chip->reg_lock); + spin_unlock(codec->lock); - snd_azf3328_codec_setdmaa(chip, codec_type, runtime->dma_addr, + snd_azf3328_codec_setdmaa(codec, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream) ); - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); #ifdef WIN9X /* FIXME: enable playback/recording??? */ flags1 |= DMA_RUN_SOMETHING1 | DMA_RUN_SOMETHING2; @@ -1242,10 +1290,10 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, DMA_EPILOGUE_SOMETHING | DMA_SOMETHING_ELSE); #endif - spin_unlock(&chip->reg_lock); - snd_azf3328_ctrl_codec_activity(chip, codec_type, 1); + spin_unlock(codec->lock); + snd_azf3328_ctrl_codec_activity(chip, codec->type, 1); - if (is_playback_codec) { + if (is_main_mixer_playback_codec) { /* now unmute WaveOut */ if (!previously_muted) snd_azf3328_mixer_set_mute( @@ -1258,19 +1306,19 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgcodec("RESUME %s\n", codec->name); /* resume codec if we were active */ - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); if (codec->running) snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, snd_azf3328_codec_inw( codec, IDX_IO_CODEC_DMA_FLAGS ) | DMA_RESUME ); - spin_unlock(&chip->reg_lock); + spin_unlock(codec->lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgcodec("STOP %s\n", codec->name); - if (is_playback_codec) { + if (is_main_mixer_playback_codec) { /* mute WaveOut (avoid clicking during setup) */ previously_muted = snd_azf3328_mixer_set_mute( @@ -1278,7 +1326,7 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, ); } - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); /* first, remember current value: */ flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS); @@ -1293,10 +1341,10 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, flags1 &= ~DMA_RUN_SOMETHING1; snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); - spin_unlock(&chip->reg_lock); - snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); + spin_unlock(codec->lock); + snd_azf3328_ctrl_codec_activity(chip, codec->type, 0); - if (is_playback_codec) { + if (is_main_mixer_playback_codec) { /* now unmute WaveOut */ if (!previously_muted) snd_azf3328_mixer_set_mute( @@ -1330,67 +1378,29 @@ snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, return result; } -static int -snd_azf3328_codec_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_azf3328_codec_trigger(AZF_CODEC_PLAYBACK, substream, cmd); -} - -static int -snd_azf3328_codec_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_azf3328_codec_trigger(AZF_CODEC_CAPTURE, substream, cmd); -} - -static int -snd_azf3328_codec_i2s_out_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_azf3328_codec_trigger(AZF_CODEC_I2S_OUT, substream, cmd); -} - static snd_pcm_uframes_t -snd_azf3328_codec_pointer(struct snd_pcm_substream *substream, - enum snd_azf3328_codec_type codec_type +snd_azf3328_pcm_pointer(struct snd_pcm_substream *substream ) { - const struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; - unsigned long bufptr, result; + const struct snd_azf3328_codec_data *codec = + substream->runtime->private_data; + unsigned long result; snd_pcm_uframes_t frmres; -#ifdef QUERY_HARDWARE - bufptr = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_START_1); -#else - bufptr = substream->runtime->dma_addr; -#endif result = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_CURRPOS); /* calculate offset */ - result -= bufptr; +#ifdef QUERY_HARDWARE + result -= snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_START_1); +#else + result -= codec->dma_base; +#endif frmres = bytes_to_frames( substream->runtime, result); - snd_azf3328_dbgcodec("%s @ 0x%8lx, frames %8ld\n", - codec->name, result, frmres); + snd_azf3328_dbgcodec("%08li %s @ 0x%8lx, frames %8ld\n", + jiffies, codec->name, result, frmres); return frmres; } -static snd_pcm_uframes_t -snd_azf3328_codec_playback_pointer(struct snd_pcm_substream *substream) -{ - return snd_azf3328_codec_pointer(substream, AZF_CODEC_PLAYBACK); -} - -static snd_pcm_uframes_t -snd_azf3328_codec_capture_pointer(struct snd_pcm_substream *substream) -{ - return snd_azf3328_codec_pointer(substream, AZF_CODEC_CAPTURE); -} - -static snd_pcm_uframes_t -snd_azf3328_codec_i2s_out_pointer(struct snd_pcm_substream *substream) -{ - return snd_azf3328_codec_pointer(substream, AZF_CODEC_I2S_OUT); -} - /******************************************************************/ #ifdef SUPPORT_GAMEPORT @@ -1532,7 +1542,7 @@ snd_azf3328_gameport_cooked_read(struct gameport *gameport, } } - /* trigger next axes sampling, to be evaluated the next time we + /* trigger next sampling of axes, to be evaluated the next time we * enter this function */ /* for some very, very strange reason we cannot enable @@ -1624,29 +1634,29 @@ snd_azf3328_irq_log_unknown_type(u8 which) } static inline void -snd_azf3328_codec_interrupt(struct snd_azf3328 *chip, u8 status) +snd_azf3328_pcm_interrupt(const struct snd_azf3328_codec_data *first_codec, + u8 status +) { u8 which; enum snd_azf3328_codec_type codec_type; - const struct snd_azf3328_codec_data *codec; + const struct snd_azf3328_codec_data *codec = first_codec; for (codec_type = AZF_CODEC_PLAYBACK; codec_type <= AZF_CODEC_I2S_OUT; - ++codec_type) { + ++codec_type, ++codec) { /* skip codec if there's no interrupt for it */ if (!(status & (1 << codec_type))) continue; - codec = &chip->codecs[codec_type]; - - spin_lock(&chip->reg_lock); + spin_lock(codec->lock); which = snd_azf3328_codec_inb(codec, IDX_IO_CODEC_IRQTYPE); /* ack all IRQ types immediately */ snd_azf3328_codec_outb(codec, IDX_IO_CODEC_IRQTYPE, which); - spin_unlock(&chip->reg_lock); + spin_unlock(codec->lock); - if ((chip->pcm[codec_type]) && (codec->substream)) { + if (codec->substream) { snd_pcm_period_elapsed(codec->substream); snd_azf3328_dbgcodec("%s period done (#%x), @ %x\n", codec->name, @@ -1701,7 +1711,7 @@ snd_azf3328_interrupt(int irq, void *dev_id) } if (status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_I2S_OUT)) - snd_azf3328_codec_interrupt(chip, status); + snd_azf3328_pcm_interrupt(chip->codecs, status); if (status & IRQ_GAMEPORT) snd_azf3328_gameport_interrupt(chip); @@ -1789,101 +1799,85 @@ snd_azf3328_pcm_open(struct snd_pcm_substream *substream, { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; snd_azf3328_dbgcallenter(); - chip->codecs[codec_type].substream = substream; + codec->substream = substream; /* same parameters for all our codecs - at least we think so... */ runtime->hw = snd_azf3328_hardware; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); + runtime->private_data = codec; snd_azf3328_dbgcallleave(); return 0; } static int -snd_azf3328_playback_open(struct snd_pcm_substream *substream) +snd_azf3328_pcm_playback_open(struct snd_pcm_substream *substream) { return snd_azf3328_pcm_open(substream, AZF_CODEC_PLAYBACK); } static int -snd_azf3328_capture_open(struct snd_pcm_substream *substream) +snd_azf3328_pcm_capture_open(struct snd_pcm_substream *substream) { return snd_azf3328_pcm_open(substream, AZF_CODEC_CAPTURE); } static int -snd_azf3328_i2s_out_open(struct snd_pcm_substream *substream) +snd_azf3328_pcm_i2s_out_open(struct snd_pcm_substream *substream) { return snd_azf3328_pcm_open(substream, AZF_CODEC_I2S_OUT); } static int -snd_azf3328_pcm_close(struct snd_pcm_substream *substream, - enum snd_azf3328_codec_type codec_type +snd_azf3328_pcm_close(struct snd_pcm_substream *substream ) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + struct snd_azf3328_codec_data *codec = + substream->runtime->private_data; snd_azf3328_dbgcallenter(); - chip->codecs[codec_type].substream = NULL; + codec->substream = NULL; snd_azf3328_dbgcallleave(); return 0; } -static int -snd_azf3328_playback_close(struct snd_pcm_substream *substream) -{ - return snd_azf3328_pcm_close(substream, AZF_CODEC_PLAYBACK); -} - -static int -snd_azf3328_capture_close(struct snd_pcm_substream *substream) -{ - return snd_azf3328_pcm_close(substream, AZF_CODEC_CAPTURE); -} - -static int -snd_azf3328_i2s_out_close(struct snd_pcm_substream *substream) -{ - return snd_azf3328_pcm_close(substream, AZF_CODEC_I2S_OUT); -} - /******************************************************************/ static struct snd_pcm_ops snd_azf3328_playback_ops = { - .open = snd_azf3328_playback_open, - .close = snd_azf3328_playback_close, + .open = snd_azf3328_pcm_playback_open, + .close = snd_azf3328_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_codec_prepare, - .trigger = snd_azf3328_codec_playback_trigger, - .pointer = snd_azf3328_codec_playback_pointer + .prepare = snd_azf3328_pcm_prepare, + .trigger = snd_azf3328_pcm_trigger, + .pointer = snd_azf3328_pcm_pointer }; static struct snd_pcm_ops snd_azf3328_capture_ops = { - .open = snd_azf3328_capture_open, - .close = snd_azf3328_capture_close, + .open = snd_azf3328_pcm_capture_open, + .close = snd_azf3328_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_codec_prepare, - .trigger = snd_azf3328_codec_capture_trigger, - .pointer = snd_azf3328_codec_capture_pointer + .prepare = snd_azf3328_pcm_prepare, + .trigger = snd_azf3328_pcm_trigger, + .pointer = snd_azf3328_pcm_pointer }; static struct snd_pcm_ops snd_azf3328_i2s_out_ops = { - .open = snd_azf3328_i2s_out_open, - .close = snd_azf3328_i2s_out_close, + .open = snd_azf3328_pcm_i2s_out_open, + .close = snd_azf3328_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_codec_prepare, - .trigger = snd_azf3328_codec_i2s_out_trigger, - .pointer = snd_azf3328_codec_i2s_out_pointer + .prepare = snd_azf3328_pcm_prepare, + .trigger = snd_azf3328_pcm_trigger, + .pointer = snd_azf3328_pcm_pointer }; static int __devinit @@ -1966,7 +1960,7 @@ snd_azf3328_timer_start(struct snd_timer *timer) snd_azf3328_dbgtimer("delay was too low (%d)!\n", delay); delay = 49; /* minimum time is 49 ticks */ } - snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay); + snd_azf3328_dbgtimer("setting timer countdown value %d\n", delay); delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE; spin_lock_irqsave(&chip->reg_lock, flags); snd_azf3328_ctrl_outl(chip, IDX_IO_TIMER_VALUE, delay); @@ -2180,6 +2174,7 @@ snd_azf3328_create(struct snd_card *card, }; u8 dma_init; enum snd_azf3328_codec_type codec_type; + struct snd_azf3328_codec_data *codec_setup; *rchip = NULL; @@ -2217,15 +2212,23 @@ snd_azf3328_create(struct snd_card *card, chip->opl3_io = pci_resource_start(pci, 3); chip->mixer_io = pci_resource_start(pci, 4); - chip->codecs[AZF_CODEC_PLAYBACK].io_base = - chip->ctrl_io + AZF_IO_OFFS_CODEC_PLAYBACK; - chip->codecs[AZF_CODEC_PLAYBACK].name = "PLAYBACK"; - chip->codecs[AZF_CODEC_CAPTURE].io_base = - chip->ctrl_io + AZF_IO_OFFS_CODEC_CAPTURE; - chip->codecs[AZF_CODEC_CAPTURE].name = "CAPTURE"; - chip->codecs[AZF_CODEC_I2S_OUT].io_base = - chip->ctrl_io + AZF_IO_OFFS_CODEC_I2S_OUT; - chip->codecs[AZF_CODEC_I2S_OUT].name = "I2S_OUT"; + codec_setup = &chip->codecs[AZF_CODEC_PLAYBACK]; + codec_setup->io_base = chip->ctrl_io + AZF_IO_OFFS_CODEC_PLAYBACK; + codec_setup->lock = &chip->reg_lock; + codec_setup->type = AZF_CODEC_PLAYBACK; + codec_setup->name = "PLAYBACK"; + + codec_setup = &chip->codecs[AZF_CODEC_CAPTURE]; + codec_setup->io_base = chip->ctrl_io + AZF_IO_OFFS_CODEC_CAPTURE; + codec_setup->lock = &chip->reg_lock; + codec_setup->type = AZF_CODEC_CAPTURE; + codec_setup->name = "CAPTURE"; + + codec_setup = &chip->codecs[AZF_CODEC_I2S_OUT]; + codec_setup->io_base = chip->ctrl_io + AZF_IO_OFFS_CODEC_I2S_OUT; + codec_setup->lock = &chip->reg_lock; + codec_setup->type = AZF_CODEC_I2S_OUT; + codec_setup->name = "I2S_OUT"; if (request_irq(pci->irq, snd_azf3328_interrupt, IRQF_SHARED, card->shortname, chip)) { @@ -2257,15 +2260,15 @@ snd_azf3328_create(struct snd_card *card, struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; - /* shutdown codecs to save power */ + /* shutdown codecs to reduce power / noise */ /* have ...ctrl_codec_activity() act properly */ codec->running = 1; snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); - spin_lock_irq(&chip->reg_lock); + spin_lock_irq(codec->lock); snd_azf3328_codec_outb(codec, IDX_IO_CODEC_DMA_FLAGS, dma_init); - spin_unlock_irq(&chip->reg_lock); + spin_unlock_irq(codec->lock); } snd_card_set_dev(card, &pci->dev); @@ -2419,6 +2422,7 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + /* same pcm object for playback/capture */ snd_pcm_suspend_all(chip->pcm[AZF_CODEC_PLAYBACK]); snd_pcm_suspend_all(chip->pcm[AZF_CODEC_I2S_OUT]); diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 37e1b5df5ab8..2958a05b5293 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -637,15 +637,9 @@ static struct snd_kcontrol_new snd_bt87x_capture_boost = { static int snd_bt87x_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *info) { - static char *texts[3] = {"TV Tuner", "FM", "Mic/Line"}; + static const char *const texts[3] = {"TV Tuner", "FM", "Mic/Line"}; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item > 2) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, texts[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 3, texts); } static int snd_bt87x_capture_source_get(struct snd_kcontrol *kcontrol, diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index f19c11077255..fc53b9bca26d 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -188,7 +188,7 @@ #define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */ /* PTR[5:0], Default: 0x0 */ #define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */ -#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */ +#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */ /* DMA[31:0], Default: 0x0 */ #define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */ /* SIZE[31:16], Default: 0x0 */ diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index d2d12c08f937..01b49388fafd 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1082,7 +1082,7 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ca0106_pcm *epcm = runtime->private_data; snd_pcm_uframes_t ptr, ptr1, ptr2 = 0; - int channel = channel=epcm->channel_id; + int channel = epcm->channel_id; if (!epcm->running) return 0; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 329968edca9b..b5bb036ef73c 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2507,14 +2507,12 @@ static int snd_cmipci_line_in_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct cmipci *cm = snd_kcontrol_chip(kcontrol); - static char *texts[3] = { "Line-In", "Rear Output", "Bass Output" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = cm->chip_version >= 39 ? 3 : 2; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + static const char *const texts[3] = { + "Line-In", "Rear Output", "Bass Output" + }; + + return snd_ctl_enum_info(uinfo, 1, + cm->chip_version >= 39 ? 3 : 2, texts); } static inline unsigned int get_line_in_mode(struct cmipci *cm) @@ -2564,14 +2562,9 @@ static int snd_cmipci_line_in_mode_put(struct snd_kcontrol *kcontrol, static int snd_cmipci_mic_in_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { "Mic-In", "Center/LFE Output" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + static const char *const texts[2] = { "Mic-In", "Center/LFE Output" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_cmipci_mic_in_mode_get(struct snd_kcontrol *kcontrol, diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c index a3301cc4ab82..185b00088320 100644 --- a/sound/pci/cs5535audio/cs5535audio_pm.c +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -90,12 +90,7 @@ int snd_cs5535audio_resume(struct pci_dev *pci) int i; pci_set_power_state(pci, PCI_D0); - if (pci_restore_state(pci) < 0) { - printk(KERN_ERR "cs5535audio: pci_restore_state failed, " - "disabling device\n"); - snd_card_disconnect(card); - return -EIO; - } + pci_restore_state(pci); if (pci_enable_device(pci) < 0) { printk(KERN_ERR "cs5535audio: pci_enable_device failed, " "disabling device\n"); diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index df47f738098d..0c701e4ec8a5 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -114,7 +114,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard."); */ #define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */ #define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */ -#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */ +#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */ #define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size */ #define PLAYBACK_POINTER 0x06 /* Playback period pointer. Sample currently in DAC */ #define PLAYBACK_UNKNOWN1 0x07 diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h index 153214940336..00f4817533b1 100644 --- a/sound/pci/emu10k1/p16v.h +++ b/sound/pci/emu10k1/p16v.h @@ -96,7 +96,7 @@ #define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */ #define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */ #define PLAYBACK_UNKNOWN3 0x03 /* Not used */ -#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */ +#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */ #define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */ #define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */ #define PLAYBACK_FIFO_END_ADDRESS 0x07 /* Playback FIFO end address */ diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 23a58f0d6cb9..7c17f45d876d 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -220,7 +220,7 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define RINGB_EN_2CODEC 0x0020 #define RINGB_SING_BIT_DUAL 0x0040 -/* ****Port Adresses**** */ +/* ****Port Addresses**** */ /* Write & Read */ #define ESM_INDEX 0x02 diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 644e3f14f8ca..ae5c5d5e4b7c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1919,6 +1919,16 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); +static int find_empty_mixer_ctl_idx(struct hda_codec *codec, const char *name) +{ + int idx; + for (idx = 0; idx < 16; idx++) { /* 16 ctlrs should be large enough */ + if (!_snd_hda_find_mixer_ctl(codec, name, idx)) + return idx; + } + return -EBUSY; +} + /** * snd_hda_ctl_add - Add a control element and assign to the codec * @codec: HD-audio codec @@ -2124,10 +2134,10 @@ int snd_hda_codec_reset(struct hda_codec *codec) * This function returns zero if successful or a negative error code. */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char **slaves) + unsigned int *tlv, const char * const *slaves) { struct snd_kcontrol *kctl; - const char **s; + const char * const *s; int err; for (s = slaves; *s && !snd_hda_find_mixer_ctl(codec, *s); s++) @@ -2654,8 +2664,6 @@ static struct snd_kcontrol_new dig_mixes[] = { { } /* end */ }; -#define SPDIF_MAX_IDX 4 /* 4 instances should be enough to probe */ - /** * snd_hda_create_spdif_out_ctls - create Output SPDIF-related controls * @codec: the HDA codec @@ -2673,12 +2681,8 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) struct snd_kcontrol_new *dig_mix; int idx; - for (idx = 0; idx < SPDIF_MAX_IDX; idx++) { - if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Playback Switch", - idx)) - break; - } - if (idx >= SPDIF_MAX_IDX) { + idx = find_empty_mixer_ctl_idx(codec, "IEC958 Playback Switch"); + if (idx < 0) { printk(KERN_ERR "hda_codec: too many IEC958 outputs\n"); return -EBUSY; } @@ -2829,12 +2833,8 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) struct snd_kcontrol_new *dig_mix; int idx; - for (idx = 0; idx < SPDIF_MAX_IDX; idx++) { - if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Capture Switch", - idx)) - break; - } - if (idx >= SPDIF_MAX_IDX) { + idx = find_empty_mixer_ctl_idx(codec, "IEC958 Capture Switch"); + if (idx < 0) { printk(KERN_ERR "hda_codec: too many IEC958 inputs\n"); return -EBUSY; } @@ -3689,7 +3689,7 @@ EXPORT_SYMBOL_HDA(snd_hda_build_pcms); * If no entries are matching, the function returns a negative value. */ int snd_hda_check_board_config(struct hda_codec *codec, - int num_configs, const char **models, + int num_configs, const char * const *models, const struct snd_pci_quirk *tbl) { if (codec->modelname && models) { @@ -3753,7 +3753,7 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_config); * If no entries are matching, the function returns a negative value. */ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, - int num_configs, const char **models, + int num_configs, const char * const *models, const struct snd_pci_quirk *tbl) { const struct snd_pci_quirk *q; @@ -3808,21 +3808,32 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) for (; knew->name; knew++) { struct snd_kcontrol *kctl; + int addr = 0, idx = 0; if (knew->iface == -1) /* skip this codec private value */ continue; - kctl = snd_ctl_new1(knew, codec); - if (!kctl) - return -ENOMEM; - err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) { - if (!codec->addr) - return err; + for (;;) { kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; - kctl->id.device = codec->addr; + if (addr > 0) + kctl->id.device = addr; + if (idx > 0) + kctl->id.index = idx; err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) + if (!err) + break; + /* try first with another device index corresponding to + * the codec addr; if it still fails (or it's the + * primary codec), then try another control index + */ + if (!addr && codec->addr) + addr = codec->addr; + else if (!idx && !knew->index) { + idx = find_empty_mixer_ctl_idx(codec, + knew->name); + if (idx <= 0) + return err; + } else return err; } } @@ -4560,6 +4571,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, } memset(cfg->hp_pins + cfg->hp_outs, 0, sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs)); + if (!cfg->hp_outs) + cfg->line_out_type = AUTO_PIN_HP_OUT; + } /* sort by sequence */ @@ -4676,7 +4690,7 @@ const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, int check_location) { unsigned int def_conf; - static const char *mic_names[] = { + static const char * const mic_names[] = { "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic", }; int attr; diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index cb0c23a6b473..4a663471dadc 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -189,6 +189,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->channels = GRAB_BITS(buf, 0, 0, 3); a->channels++; + a->sample_bits = 0; + a->max_bitrate = 0; + a->format = GRAB_BITS(buf, 0, 3, 4); switch (a->format) { case AUDIO_CODING_TYPE_REF_STREAM_HEADER: @@ -198,7 +201,6 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, case AUDIO_CODING_TYPE_LPCM: val = GRAB_BITS(buf, 2, 0, 3); - a->sample_bits = 0; for (i = 0; i < 3; i++) if (val & (1 << i)) a->sample_bits |= cea_sample_sizes[i + 1]; @@ -598,24 +600,19 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, { int i; - pcm->rates = 0; - pcm->formats = 0; - pcm->maxbps = 0; - pcm->channels_min = -1; - pcm->channels_max = 0; + /* assume basic audio support (the basic audio flag is not in ELD; + * however, all audio capable sinks are required to support basic + * audio) */ + pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; + pcm->formats = SNDRV_PCM_FMTBIT_S16_LE; + pcm->maxbps = 16; + pcm->channels_max = 2; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; - if (a->channels < pcm->channels_min) - pcm->channels_min = a->channels; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { - if (a->sample_bits & AC_SUPPCM_BITS_16) { - pcm->formats |= SNDRV_PCM_FMTBIT_S16_LE; - if (pcm->maxbps < 16) - pcm->maxbps = 16; - } if (a->sample_bits & AC_SUPPCM_BITS_20) { pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE; if (pcm->maxbps < 20) @@ -635,7 +632,6 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, /* restrict the parameters by the values the codec provides */ pcm->rates &= codec_pars->rates; pcm->formats &= codec_pars->formats; - pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min); pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max); pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index fb0582f8d725..a63c54d9d767 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -762,7 +762,8 @@ static int check_existing_control(struct hda_codec *codec, const char *type, con /* * build output mixer controls */ -static int create_output_mixers(struct hda_codec *codec, const char **names) +static int create_output_mixers(struct hda_codec *codec, + const char * const *names) { struct hda_gspec *spec = codec->spec; int i, err; @@ -780,8 +781,8 @@ static int create_output_mixers(struct hda_codec *codec, const char **names) static int build_output_controls(struct hda_codec *codec) { struct hda_gspec *spec = codec->spec; - static const char *types_speaker[] = { "Speaker", "Headphone" }; - static const char *types_line[] = { "Front", "Headphone" }; + static const char * const types_speaker[] = { "Speaker", "Headphone" }; + static const char * const types_line[] = { "Front", "Headphone" }; switch (spec->pcm_vol_nodes) { case 1: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21aa9b0e28f6..2e91a991eb15 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1235,7 +1235,8 @@ static int azx_setup_periods(struct azx *chip, pos_adj = 0; } else { ofs = setup_bdle(substream, azx_dev, - &bdl, ofs, pos_adj, 1); + &bdl, ofs, pos_adj, + !substream->runtime->no_period_wakeup); if (ofs < 0) goto error; } @@ -1247,7 +1248,8 @@ static int azx_setup_periods(struct azx *chip, period_bytes - pos_adj, 0); else ofs = setup_bdle(substream, azx_dev, &bdl, ofs, - period_bytes, 1); + period_bytes, + !substream->runtime->no_period_wakeup); if (ofs < 0) goto error; } @@ -1515,7 +1517,8 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* No full-resume yet implemented */ /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START), + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, @@ -2296,9 +2299,11 @@ static int azx_dev_free(struct snd_device *device) */ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1028, 0x0470, "Dell Inspiron 1120", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), @@ -2804,6 +2809,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { #endif /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, + /* VMware HDAudio */ + { PCI_DEVICE(0x15ad, 0x1977), .driver_data = AZX_DRIVER_GENERIC }, /* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */ { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 46bbefe2e4a9..3ab5e7a303db 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -140,7 +140,7 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); int snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char **slaves); + unsigned int *tlv, const char * const *slaves); int snd_hda_codec_reset(struct hda_codec *codec); /* amp value bits */ @@ -341,10 +341,10 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen); * Misc */ int snd_hda_check_board_config(struct hda_codec *codec, int num_configs, - const char **modelnames, + const char * const *modelnames, const struct snd_pci_quirk *pci_list); int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, - int num_configs, const char **models, + int num_configs, const char * const *models, const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index f025200f2a62..bfe74c2fb079 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -418,7 +418,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer, static const char *get_pwr_state(u32 state) { - static const char *buf[4] = { + static const char * const buf[4] = { "D0", "D1", "D2", "D3" }; if (state < 4) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index f7ff3f7ccb8e..8dabab798689 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -46,6 +46,9 @@ struct ad198x_spec { unsigned int cur_eapd; unsigned int need_dac_fix; + hda_nid_t *alt_dac_nid; + struct hda_pcm_stream *stream_analog_alt_playback; + /* capture */ unsigned int num_adc_nids; hda_nid_t *adc_nids; @@ -81,8 +84,8 @@ struct ad198x_spec { #endif /* for virtual master */ hda_nid_t vmaster_nid; - const char **slave_vols; - const char **slave_sws; + const char * const *slave_vols; + const char * const *slave_sws; }; /* @@ -130,7 +133,7 @@ static int ad198x_init(struct hda_codec *codec) return 0; } -static const char *ad_slave_vols[] = { +static const char * const ad_slave_vols[] = { "Front Playback Volume", "Surround Playback Volume", "Center Playback Volume", @@ -143,7 +146,7 @@ static const char *ad_slave_vols[] = { NULL }; -static const char *ad_slave_sws[] = { +static const char * const ad_slave_sws[] = { "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch", @@ -156,6 +159,25 @@ static const char *ad_slave_sws[] = { NULL }; +static const char * const ad1988_6stack_fp_slave_vols[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + "Side Playback Volume", + "IEC958 Playback Volume", + NULL +}; + +static const char * const ad1988_6stack_fp_slave_sws[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + "Side Playback Switch", + "IEC958 Playback Switch", + NULL +}; static void ad198x_free_kctls(struct hda_codec *codec); #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -309,6 +331,38 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } +static int ad198x_alt_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ad198x_spec *spec = codec->spec; + snd_hda_codec_setup_stream(codec, spec->alt_dac_nid[0], stream_tag, + 0, format); + return 0; +} + +static int ad198x_alt_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ad198x_spec *spec = codec->spec; + snd_hda_codec_cleanup_stream(codec, spec->alt_dac_nid[0]); + return 0; +} + +static struct hda_pcm_stream ad198x_pcm_analog_alt_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in ad198x_build_pcms */ + .ops = { + .prepare = ad198x_alt_playback_pcm_prepare, + .cleanup = ad198x_alt_playback_pcm_cleanup + }, +}; + /* * Digital out */ @@ -446,6 +500,17 @@ static int ad198x_build_pcms(struct hda_codec *codec) } } + if (spec->alt_dac_nid && spec->stream_analog_alt_playback) { + codec->num_pcms++; + info = spec->pcm_rec + 2; + info->name = "AD198x Headphone"; + info->pcm_type = HDA_PCM_TYPE_AUDIO; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *spec->stream_analog_alt_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->alt_dac_nid[0]; + } + return 0; } @@ -666,7 +731,7 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), @@ -729,7 +794,7 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT), /* HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */ @@ -775,7 +840,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), { @@ -1069,7 +1134,7 @@ enum { AD1986A_MODELS }; -static const char *ad1986a_models[AD1986A_MODELS] = { +static const char * const ad1986a_models[AD1986A_MODELS] = { [AD1986A_6STACK] = "6stack", [AD1986A_3STACK] = "3stack", [AD1986A_LAPTOP] = "laptop", @@ -1358,7 +1423,7 @@ static struct snd_kcontrol_new ad1983_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), { @@ -1515,8 +1580,8 @@ static struct snd_kcontrol_new ad1981_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 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("Front Mic Boost", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), { @@ -1726,8 +1791,8 @@ static struct snd_kcontrol_new ad1981_hp_mixers[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), #endif - HDA_CODEC_VOLUME("Mic Boost", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost", 0x18, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x18, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), { @@ -1774,7 +1839,7 @@ static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = { 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("Mic Boost Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), { @@ -1813,7 +1878,7 @@ enum { AD1981_MODELS }; -static const char *ad1981_models[AD1981_MODELS] = { +static const char * const ad1981_models[AD1981_MODELS] = { [AD1981_HP] = "hp", [AD1981_THINKPAD] = "thinkpad", [AD1981_BASIC] = "basic", @@ -2015,6 +2080,7 @@ static int patch_ad1981(struct hda_codec *codec) enum { AD1988_6STACK, AD1988_6STACK_DIG, + AD1988_6STACK_DIG_FP, AD1988_3STACK, AD1988_3STACK_DIG, AD1988_LAPTOP, @@ -2047,6 +2113,10 @@ static hda_nid_t ad1988_6stack_dac_nids_rev2[4] = { 0x04, 0x05, 0x0a, 0x06 }; +static hda_nid_t ad1988_alt_dac_nid[1] = { + 0x03 +}; + static hda_nid_t ad1988_3stack_dac_nids_rev2[3] = { 0x04, 0x0a, 0x06 }; @@ -2160,8 +2230,37 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), + + { } /* end */ +}; + +static struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + + HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), + HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), + + HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -2203,8 +2302,8 @@ static struct snd_kcontrol_new ad1988_3stack_mixers2[] = { HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -2232,7 +2331,7 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2445,6 +2544,68 @@ static struct hda_verb ad1988_6stack_init_verbs[] = { { } }; +static struct hda_verb ad1988_6stack_fp_init_verbs[] = { + /* Front, Surround, CLFE, side DAC; unmute as default */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Headphone; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-A front headphon path */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-D line-out path */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Port-F surround path */ + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Port-G CLFE path */ + {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Port-H side path */ + {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* Mono out path */ + {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ + /* Port-B front mic-in path */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Port-C line-in path */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Port-E mic-in path */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Analog CD Input */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ + + { } +}; + static struct hda_verb ad1988_capture_init_verbs[] = { /* mute analog mix */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -2792,7 +2953,9 @@ static int ad1988_auto_create_multi_out_ctls(struct ad198x_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; + static const char * const chname[4] = { + "Front", "Surround", NULL /*CLFE*/, "Side" + }; hda_nid_t nid; int i, err; @@ -2902,7 +3065,7 @@ static int new_analog_input(struct ad198x_spec *spec, hda_nid_t pin, idx = ad1988_pin_idx(pin); bnid = ad1988_boost_nids[idx]; if (bnid) { - sprintf(name, "%s Boost", ctlname); + sprintf(name, "%s Boost Volume", ctlname); return add_control(spec, AD_CTL_WIDGET_VOL, name, HDA_COMPOSE_AMP_VAL(bnid, 3, idx, HDA_OUTPUT)); @@ -3074,13 +3237,13 @@ static int ad1988_auto_init(struct hda_codec *codec) return 0; } - /* */ -static const char *ad1988_models[AD1988_MODEL_LAST] = { +static const char * const ad1988_models[AD1988_MODEL_LAST] = { [AD1988_6STACK] = "6stack", [AD1988_6STACK_DIG] = "6stack-dig", + [AD1988_6STACK_DIG_FP] = "6stack-dig-fp", [AD1988_3STACK] = "3stack", [AD1988_3STACK_DIG] = "3stack-dig", [AD1988_LAPTOP] = "laptop", @@ -3140,6 +3303,7 @@ static int patch_ad1988(struct hda_codec *codec) switch (board_config) { case AD1988_6STACK: case AD1988_6STACK_DIG: + case AD1988_6STACK_DIG_FP: spec->multiout.max_channels = 8; spec->multiout.num_dacs = 4; if (is_rev2(codec)) @@ -3152,10 +3316,22 @@ static int patch_ad1988(struct hda_codec *codec) spec->mixers[0] = ad1988_6stack_mixers1_rev2; else spec->mixers[0] = ad1988_6stack_mixers1; - spec->mixers[1] = ad1988_6stack_mixers2; + if (board_config == AD1988_6STACK_DIG_FP) { + spec->mixers[1] = ad1988_6stack_fp_mixers; + spec->slave_vols = ad1988_6stack_fp_slave_vols; + spec->slave_sws = ad1988_6stack_fp_slave_sws; + spec->alt_dac_nid = ad1988_alt_dac_nid; + spec->stream_analog_alt_playback = + &ad198x_pcm_analog_alt_playback; + } else + spec->mixers[1] = ad1988_6stack_mixers2; spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1988_6stack_init_verbs; - if (board_config == AD1988_6STACK_DIG) { + if (board_config == AD1988_6STACK_DIG_FP) + spec->init_verbs[0] = ad1988_6stack_fp_init_verbs; + else + spec->init_verbs[0] = ad1988_6stack_init_verbs; + if ((board_config == AD1988_6STACK_DIG) || + (board_config == AD1988_6STACK_DIG_FP)) { spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; spec->dig_in_nid = AD1988_SPDIF_IN; } @@ -3300,8 +3476,8 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -3399,7 +3575,7 @@ static struct hda_amp_list ad1884_loopbacks[] = { }; #endif -static const char *ad1884_slave_vols[] = { +static const char * const ad1884_slave_vols[] = { "PCM Playback Volume", "Mic Playback Volume", "Mono Playback Volume", @@ -3499,9 +3675,9 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -3560,8 +3736,8 @@ static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Line-In Boost", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line-In Boost Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -3637,7 +3813,7 @@ enum { AD1984_MODELS }; -static const char *ad1984_models[AD1984_MODELS] = { +static const char * const ad1984_models[AD1984_MODELS] = { [AD1984_BASIC] = "basic", [AD1984_THINKPAD] = "thinkpad", [AD1984_DELL_DESKTOP] = "dell_desktop", @@ -3745,9 +3921,9 @@ static struct snd_kcontrol_new ad1884a_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -3888,9 +4064,9 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), { } /* end */ @@ -4126,8 +4302,8 @@ static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), { @@ -4255,8 +4431,8 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT), { } /* end */ }; @@ -4308,7 +4484,7 @@ enum { AD1884A_MODELS }; -static const char *ad1884a_models[AD1884A_MODELS] = { +static const char * const ad1884a_models[AD1884A_MODELS] = { [AD1884A_DESKTOP] = "desktop", [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", @@ -4494,9 +4670,9 @@ static struct snd_kcontrol_new ad1882_base_mixers[] = { HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line-In Boost", 0x3a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line-In Boost Volume", 0x3a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -4547,7 +4723,7 @@ static struct snd_kcontrol_new ad1882a_loopback_mixers[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Digital Mic Boost Volume", 0x1f, 0x0, HDA_INPUT), { } /* end */ }; @@ -4696,7 +4872,7 @@ enum { AD1882_MODELS }; -static const char *ad1882_models[AD1986A_MODELS] = { +static const char * const ad1882_models[AD1986A_MODELS] = { [AD1882_3STACK] = "3stack", [AD1882_6STACK] = "6stack", }; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 18af38ebf757..a07b031090d8 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -490,7 +490,7 @@ static int parse_digital_input(struct hda_codec *codec) * create mixer controls */ -static const char *dir_sfx[2] = { "Playback", "Capture" }; +static const char * const dir_sfx[2] = { "Playback", "Capture" }; static int add_mute(struct hda_codec *codec, const char *name, int index, unsigned int pval, int dir, struct snd_kcontrol **kctlp) @@ -1156,7 +1156,7 @@ static int cs_parse_auto_config(struct hda_codec *codec) return 0; } -static const char *cs420x_models[CS420X_MODELS] = { +static const char * const cs420x_models[CS420X_MODELS] = { [CS420X_MBP53] = "mbp53", [CS420X_MBP55] = "mbp55", [CS420X_IMAC27] = "imac27", diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index ff60908f4554..1f8bbcd0f802 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -608,7 +608,7 @@ static void cmi9880_free(struct hda_codec *codec) /* */ -static const char *cmi9880_models[CMI_MODELS] = { +static const char * const cmi9880_models[CMI_MODELS] = { [CMI_MINIMAL] = "minimal", [CMI_MIN_FP] = "min_fp", [CMI_FULL] = "full", diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 846d1ead47fd..9bb030a469cd 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -537,13 +537,13 @@ static struct snd_kcontrol_new cxt_beep_mixer[] = { }; #endif -static const char *slave_vols[] = { +static const char * const slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", NULL }; -static const char *slave_sws[] = { +static const char * const slave_sws[] = { "Headphone Playback Switch", "Speaker Playback Switch", NULL @@ -869,16 +869,16 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, } static struct snd_kcontrol_new cxt5045_mixers[] = { - HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Ext Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x17, 0x2, HDA_INPUT), HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -910,16 +910,16 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { }; static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { - HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Ext Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x17, 0x1, HDA_INPUT), HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -947,7 +947,7 @@ static struct hda_verb cxt5045_init_verbs[] = { {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Record selector: Int mic */ + /* Record selector: Internal mic */ {0x1a, AC_VERB_SET_CONNECT_SEL,0x1}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, @@ -960,7 +960,7 @@ static struct hda_verb cxt5045_init_verbs[] = { }; static struct hda_verb cxt5045_benq_init_verbs[] = { - /* Int Mic, Mic */ + /* Internal Mic, Mic */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, /* Line In,HP, Amp */ @@ -973,7 +973,7 @@ static struct hda_verb cxt5045_benq_init_verbs[] = { {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Record selector: Int mic */ + /* Record selector: Internal mic */ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, @@ -1134,7 +1134,7 @@ enum { CXT5045_MODELS }; -static const char *cxt5045_models[CXT5045_MODELS] = { +static const char * const cxt5045_models[CXT5045_MODELS] = { [CXT5045_LAPTOP_HPSENSE] = "laptop-hpsense", [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense", [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense", @@ -1376,7 +1376,7 @@ static void cxt5047_hp_unsol_event(struct hda_codec *codec, static struct snd_kcontrol_new cxt5047_base_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), @@ -1579,7 +1579,7 @@ enum { CXT5047_MODELS }; -static const char *cxt5047_models[CXT5047_MODELS] = { +static const char * const cxt5047_models[CXT5047_MODELS] = { [CXT5047_LAPTOP] = "laptop", [CXT5047_LAPTOP_HP] = "laptop-hp", [CXT5047_LAPTOP_EAPD] = "laptop-eapd", @@ -1796,8 +1796,8 @@ static struct snd_kcontrol_new cxt5051_playback_mixers[] = { static struct snd_kcontrol_new cxt5051_capture_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), {} @@ -1806,8 +1806,8 @@ static struct snd_kcontrol_new cxt5051_capture_mixers[] = { static struct snd_kcontrol_new cxt5051_hp_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("External Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("External Mic Switch", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x15, 0x00, HDA_INPUT), {} }; @@ -1826,8 +1826,8 @@ static struct snd_kcontrol_new cxt5051_f700_mixers[] = { static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), {} }; @@ -1847,7 +1847,7 @@ static struct hda_verb cxt5051_init_verbs[] = { {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Int mic */ + /* Record selector: Internal mic */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, @@ -1874,7 +1874,7 @@ static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Int mic */ + /* Record selector: Internal mic */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, /* SPDIF route: PCM */ @@ -1904,7 +1904,7 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Int mic */ + /* Record selector: Internal mic */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, @@ -1932,7 +1932,7 @@ static struct hda_verb cxt5051_f700_init_verbs[] = { {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Record selector: Int mic */ + /* Record selector: Internal mic */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, /* SPDIF route: PCM */ @@ -1995,7 +1995,7 @@ enum { CXT5051_MODELS }; -static const char *cxt5051_models[CXT5051_MODELS] = { +static const char *const cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", @@ -2111,25 +2111,35 @@ static struct hda_channel_mode cxt5066_modes[1] = { { 2, NULL }, }; +#define HP_PRESENT_PORT_A (1 << 0) +#define HP_PRESENT_PORT_D (1 << 1) +#define hp_port_a_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_A) +#define hp_port_d_present(spec) ((spec)->hp_present & HP_PRESENT_PORT_D) + static void cxt5066_update_speaker(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; unsigned int pinctl; - snd_printdd("CXT5066: update speaker, hp_present=%d\n", - spec->hp_present); + snd_printdd("CXT5066: update speaker, hp_present=%d, cur_eapd=%d\n", + spec->hp_present, spec->cur_eapd); /* Port A (HP) */ - pinctl = ((spec->hp_present & 1) && spec->cur_eapd) ? PIN_HP : 0; + pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0; snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); /* Port D (HP/LO) */ - pinctl = ((spec->hp_present & 2) && spec->cur_eapd) - ? spec->port_d_mode : 0; - /* Mute if Port A is connected on Thinkpad */ - if (spec->thinkpad && (spec->hp_present & 1)) - pinctl = 0; + pinctl = spec->cur_eapd ? spec->port_d_mode : 0; + if (spec->dell_automute || spec->thinkpad) { + /* Mute if Port A is connected */ + if (hp_port_a_present(spec)) + pinctl = 0; + } else { + /* Thinkpad/Dell doesn't give pin-D status */ + if (!hp_port_d_present(spec)) + pinctl = 0; + } snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); @@ -2137,14 +2147,6 @@ static void cxt5066_update_speaker(struct hda_codec *codec) pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); - - if (spec->dell_automute) { - /* DELL AIO Port Rule: PortA > PortD > IntSpk */ - pinctl = (!(spec->hp_present & 1) && spec->cur_eapd) - ? PIN_OUT : 0; - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); - } } /* turn on/off EAPD (+ mute HP) as a master switch */ @@ -2378,8 +2380,8 @@ static void cxt5066_hp_automute(struct hda_codec *codec) /* Port D */ portD = snd_hda_jack_detect(codec, 0x1c); - spec->hp_present = !!(portA); - spec->hp_present |= portD ? 2 : 0; + spec->hp_present = portA ? HP_PRESENT_PORT_A : 0; + spec->hp_present |= portD ? HP_PRESENT_PORT_D : 0; snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n", portA, portD, spec->hp_present); cxt5066_update_speaker(codec); @@ -2727,7 +2729,7 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { static struct snd_kcontrol_new cxt5066_vostro_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Int Mic Boost Capture Enum", + .name = "Internal Mic Boost Capture Enum", .info = cxt5066_mic_boost_mux_enum_info, .get = cxt5066_mic_boost_mux_enum_get, .put = cxt5066_mic_boost_mux_enum_put, @@ -2953,7 +2955,7 @@ static struct hda_verb cxt5066_init_verbs_ideapad[] = { {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* internal microphone */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable int mic */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */ /* EAPD */ {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ @@ -3008,7 +3010,7 @@ static struct hda_verb cxt5066_init_verbs_thinkpad[] = { {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* internal microphone */ - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable int mic */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable internal mic */ /* EAPD */ {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ @@ -3082,7 +3084,7 @@ enum { CXT5066_MODELS }; -static const char *cxt5066_models[CXT5066_MODELS] = { +static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP] = "laptop", [CXT5066_DELL_LAPTOP] = "dell-laptop", [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", @@ -3095,8 +3097,8 @@ static const char *cxt5066_models[CXT5066_MODELS] = { static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", - CXT5066_DELL_LAPTOP), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), @@ -3108,15 +3110,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x21c8, "Thinkpad Edge 11", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} }; @@ -3421,6 +3417,9 @@ static void cx_auto_hp_automute(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, present ? 0 : PIN_OUT); } + for (i = 0; !present && i < cfg->line_outs; i++) + if (snd_hda_jack_detect(codec, cfg->line_out_pins[i])) + present = 1; for (i = 0; i < cfg->speaker_outs; i++) { snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, @@ -3747,7 +3746,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; int i, err; int num_line = 0, num_hp = 0, num_spk = 0; - static const char *texts[3] = { "Front", "Surround", "CLFE" }; + static const char * const texts[3] = { "Front", "Surround", "CLFE" }; if (spec->dac_info_filled == 1) return cx_auto_add_pb_volume(codec, spec->dac_info[0].dac, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d3e49aa5b9ec..2d5b83fa8d24 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -31,10 +31,15 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/moduleparam.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +static bool static_hdmi_pcm; +module_param(static_hdmi_pcm, bool, 0644); +MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); + /* * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device * could support two independent pipes, each of them can be connected to one or @@ -812,6 +817,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld; struct hda_pcm_stream *codec_pars; + struct snd_pcm_runtime *runtime = substream->runtime; unsigned int idx; for (idx = 0; idx < spec->num_cvts; idx++) @@ -827,19 +833,26 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, *codec_pars = *hinfo; eld = &spec->sink_eld[idx]; - if (eld->sad_count > 0) { + if (!static_hdmi_pcm && eld->eld_valid && eld->sad_count > 0) { hdmi_eld_update_pcm_info(eld, hinfo, codec_pars); if (hinfo->channels_min > hinfo->channels_max || !hinfo->rates || !hinfo->formats) return -ENODEV; } else { /* fallback to the codec default */ - hinfo->channels_min = codec_pars->channels_min; hinfo->channels_max = codec_pars->channels_max; hinfo->rates = codec_pars->rates; hinfo->formats = codec_pars->formats; hinfo->maxbps = codec_pars->maxbps; } + /* store the updated parameters */ + runtime->hw.channels_min = hinfo->channels_min; + runtime->hw.channels_max = hinfo->channels_max; + runtime->hw.formats = hinfo->formats; + runtime->hw.rates = hinfo->rates; + + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, 2); return 0; } @@ -905,23 +918,28 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) spec->pin[spec->num_pins] = pin_nid; spec->num_pins++; - /* - * It is assumed that converter nodes come first in the node list and - * hence have been registered and usable now. - */ return hdmi_read_pin_conn(codec, pin_nid); } static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) { + int i, found_pin = 0; struct hdmi_spec *spec = codec->spec; - if (spec->num_cvts >= MAX_HDMI_CVTS) { - snd_printk(KERN_WARNING - "HDMI: no space for converter %d\n", nid); - return -E2BIG; + for (i = 0; i < spec->num_pins; i++) + if (nid == spec->pin_cvt[i]) { + found_pin = 1; + break; + } + + if (!found_pin) { + snd_printdd("HDMI: Skipping node %d (no connection)\n", nid); + return -EINVAL; } + if (snd_BUG_ON(spec->num_cvts >= MAX_HDMI_CVTS)) + return -E2BIG; + spec->cvt[spec->num_cvts] = nid; spec->num_cvts++; @@ -932,6 +950,8 @@ static int hdmi_parse_codec(struct hda_codec *codec) { hda_nid_t nid; int i, nodes; + int num_tmp_cvts = 0; + hda_nid_t tmp_cvt[MAX_HDMI_CVTS]; nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); if (!nid || nodes < 0) { @@ -942,6 +962,7 @@ static int hdmi_parse_codec(struct hda_codec *codec) for (i = 0; i < nodes; i++, nid++) { unsigned int caps; unsigned int type; + unsigned int config; caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); type = get_wcaps_type(caps); @@ -951,17 +972,32 @@ static int hdmi_parse_codec(struct hda_codec *codec) switch (type) { case AC_WID_AUD_OUT: - hdmi_add_cvt(codec, nid); + if (num_tmp_cvts >= MAX_HDMI_CVTS) { + snd_printk(KERN_WARNING + "HDMI: no space for converter %d\n", nid); + continue; + } + tmp_cvt[num_tmp_cvts] = nid; + num_tmp_cvts++; break; case AC_WID_PIN: caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) continue; + + config = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + continue; + hdmi_add_pin(codec, nid); break; } } + for (i = 0; i < num_tmp_cvts; i++) + hdmi_add_cvt(codec, tmp_cvt[i]); + /* * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event * can be lost and presence sense verb will become inaccurate if the @@ -1166,11 +1202,56 @@ static int nvhdmi_7x_init(struct hda_codec *codec) return 0; } +static unsigned int channels_2_6_8[] = { + 2, 6, 8 +}; + +static unsigned int channels_2_8[] = { + 2, 8 +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_2_6_8_channels = { + .count = ARRAY_SIZE(channels_2_6_8), + .list = channels_2_6_8, + .mask = 0, +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_2_8_channels = { + .count = ARRAY_SIZE(channels_2_8), + .list = channels_2_8, + .mask = 0, +}; + static int simple_playback_pcm_open(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; + struct snd_pcm_hw_constraint_list *hw_constraints_channels = NULL; + + switch (codec->preset->id) { + case 0x10de0002: + case 0x10de0003: + case 0x10de0005: + case 0x10de0006: + hw_constraints_channels = &hw_constraints_2_8_channels; + break; + case 0x10de0007: + hw_constraints_channels = &hw_constraints_2_6_8_channels; + break; + default: + break; + } + + if (hw_constraints_channels != NULL) { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + hw_constraints_channels); + } else { + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, 2); + } + return snd_hda_multi_out_dig_open(codec, &spec->multiout); } @@ -1533,7 +1614,7 @@ static struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, -{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi }, +{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_generic_hdmi }, { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_generic_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_generic_hdmi }, { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_generic_hdmi }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8fddc9d08726..269dbff70b92 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -231,7 +231,6 @@ enum { ALC888_ACER_ASPIRE_8930G, ALC888_ACER_ASPIRE_7730G, ALC883_MEDION, - ALC883_MEDION_MD2, ALC883_MEDION_WIM2160, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, @@ -304,6 +303,8 @@ struct alc_customize_define { unsigned int fixup:1; /* Means that this sku is set by driver, not read from hw */ }; +struct alc_fixup; + struct alc_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -405,6 +406,11 @@ struct alc_spec { /* for PLL fix */ hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; + + /* fix-up list */ + int fixup_id; + const struct alc_fixup *fixup_list; + const char *fixup_name; }; /* @@ -1678,48 +1684,137 @@ struct alc_pincfg { u32 val; }; +struct alc_model_fixup { + const int id; + const char *name; +}; + struct alc_fixup { - unsigned int sku; - const struct alc_pincfg *pins; - const struct hda_verb *verbs; + int type; + bool chained; + int chain_id; + union { + unsigned int sku; + const struct alc_pincfg *pins; + const struct hda_verb *verbs; + void (*func)(struct hda_codec *codec, + const struct alc_fixup *fix, + int action); + } v; }; -static void alc_pick_fixup(struct hda_codec *codec, - const struct snd_pci_quirk *quirk, - const struct alc_fixup *fix, - int pre_init) +enum { + ALC_FIXUP_INVALID, + ALC_FIXUP_SKU, + ALC_FIXUP_PINS, + ALC_FIXUP_VERBS, + ALC_FIXUP_FUNC, +}; + +enum { + ALC_FIXUP_ACT_PRE_PROBE, + ALC_FIXUP_ACT_PROBE, + ALC_FIXUP_ACT_INIT, +}; + +static void alc_apply_fixup(struct hda_codec *codec, int action) { - const struct alc_pincfg *cfg; - struct alc_spec *spec; + struct alc_spec *spec = codec->spec; + int id = spec->fixup_id; + const char *modelname = spec->fixup_name; + int depth = 0; - quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (!quirk) + if (!spec->fixup_list) return; - fix += quirk->value; - cfg = fix->pins; - if (pre_init && fix->sku) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - snd_printdd(KERN_INFO "hda_codec: %s: Apply sku override for %s\n", - codec->chip_name, quirk->name); -#endif - spec = codec->spec; - spec->cdefine.sku_cfg = fix->sku; - spec->cdefine.fixup = 1; + + while (id >= 0) { + const struct alc_fixup *fix = spec->fixup_list + id; + const struct alc_pincfg *cfg; + + switch (fix->type) { + case ALC_FIXUP_SKU: + if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku) + break;; + snd_printdd(KERN_INFO "hda_codec: %s: " + "Apply sku override for %s\n", + codec->chip_name, modelname); + spec->cdefine.sku_cfg = fix->v.sku; + spec->cdefine.fixup = 1; + break; + case ALC_FIXUP_PINS: + cfg = fix->v.pins; + if (action != ALC_FIXUP_ACT_PRE_PROBE || !cfg) + break; + snd_printdd(KERN_INFO "hda_codec: %s: " + "Apply pincfg for %s\n", + codec->chip_name, modelname); + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, + cfg->val); + break; + case ALC_FIXUP_VERBS: + if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs) + break; + snd_printdd(KERN_INFO "hda_codec: %s: " + "Apply fix-verbs for %s\n", + codec->chip_name, modelname); + add_verb(codec->spec, fix->v.verbs); + break; + case ALC_FIXUP_FUNC: + if (!fix->v.func) + break; + snd_printdd(KERN_INFO "hda_codec: %s: " + "Apply fix-func for %s\n", + codec->chip_name, modelname); + fix->v.func(codec, fix, action); + break; + default: + snd_printk(KERN_ERR "hda_codec: %s: " + "Invalid fixup type %d\n", + codec->chip_name, fix->type); + break; + } + if (!fix[id].chained) + break; + if (++depth > 10) + break; + id = fix[id].chain_id; } - if (pre_init && cfg) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n", - codec->chip_name, quirk->name); -#endif - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); +} + +static void alc_pick_fixup(struct hda_codec *codec, + const struct alc_model_fixup *models, + const struct snd_pci_quirk *quirk, + const struct alc_fixup *fixlist) +{ + struct alc_spec *spec = codec->spec; + int id = -1; + const char *name = NULL; + + if (codec->modelname && models) { + while (models->name) { + if (!strcmp(codec->modelname, models->name)) { + id = models->id; + name = models->name; + break; + } + models++; + } } - if (!pre_init && fix->verbs) { + if (id < 0) { + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); + if (quirk) { + id = quirk->value; #ifdef CONFIG_SND_DEBUG_VERBOSE - snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n", - codec->chip_name, quirk->name); + name = quirk->name; #endif - add_verb(codec->spec, fix->verbs); + } + } + + spec->fixup_id = id; + if (id >= 0) { + spec->fixup_list = fixlist; + spec->fixup_name = name; } } @@ -1981,6 +2076,7 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; @@ -2120,17 +2216,17 @@ static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { { .num_items = 5, .items = { - { "Ext Mic", 0x0 }, + { "Mic", 0x0 }, { "Line In", 0x2 }, { "CD", 0x4 }, { "Input Mix", 0xa }, - { "Int Mic", 0xb }, + { "Internal Mic", 0xb }, }, }, { .num_items = 4, .items = { - { "Ext Mic", 0x0 }, + { "Mic", 0x0 }, { "Line In", 0x2 }, { "CD", 0x4 }, { "Input Mix", 0xa }, @@ -2187,7 +2283,7 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -2205,7 +2301,7 @@ static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -2796,10 +2892,10 @@ static struct snd_kcontrol_new alc880_fujitsu_mixer[] = { HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -2820,7 +2916,7 @@ static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = { /* * slave controls for virtual master */ -static const char *alc_slave_vols[] = { +static const char * const alc_slave_vols[] = { "Front Playback Volume", "Surround Playback Volume", "Center Playback Volume", @@ -2834,7 +2930,7 @@ static const char *alc_slave_vols[] = { NULL, }; -static const char *alc_slave_sws[] = { +static const char * const alc_slave_sws[] = { "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch", @@ -3307,7 +3403,7 @@ static struct hda_verb alc880_beep_init_verbs[] = { }; /* auto-toggle front mic */ -static void alc880_uniwill_mic_automute(struct hda_codec *codec) +static void alc88x_simple_mic_automute(struct hda_codec *codec) { unsigned int present; unsigned char bits; @@ -3329,7 +3425,7 @@ static void alc880_uniwill_setup(struct hda_codec *codec) static void alc880_uniwill_init_hook(struct hda_codec *codec) { alc_automute_amp(codec); - alc880_uniwill_mic_automute(codec); + alc88x_simple_mic_automute(codec); } static void alc880_uniwill_unsol_event(struct hda_codec *codec, @@ -3340,7 +3436,7 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, */ switch (res >> 28) { case ALC880_MIC_EVENT: - alc880_uniwill_mic_automute(codec); + alc88x_simple_mic_automute(codec); break; default: alc_automute_amp_unsol_event(codec, res); @@ -3815,6 +3911,8 @@ static int alc_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); + alc_apply_fixup(codec, ALC_FIXUP_ACT_INIT); + hda_call_check_power_status(codec, 0x01); return 0; } @@ -4513,7 +4611,7 @@ static struct hda_verb alc880_test_init_verbs[] = { /* */ -static const char *alc880_models[ALC880_MODEL_LAST] = { +static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_3ST] = "3stack", [ALC880_TCL_S700] = "tcl", [ALC880_3ST_DIG] = "3stack-digout", @@ -4595,6 +4693,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG), + SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG), SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700), @@ -5022,13 +5121,33 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, return 0; } +static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg, + bool can_be_master) +{ + if (!cfg->hp_outs && !cfg->speaker_outs && can_be_master) + return "Master"; + + switch (cfg->line_out_type) { + case AUTO_PIN_SPEAKER_OUT: + return "Speaker"; + case AUTO_PIN_HP_OUT: + return "Headphone"; + default: + if (cfg->line_outs == 1) + return "PCM"; + break; + } + return NULL; +} + /* add playback controls from the parsed DAC table */ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - static const char *chname[4] = { + static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; + const char *pfx = alc_get_line_out_pfx(cfg, false); hda_nid_t nid; int i, err; @@ -5036,7 +5155,7 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, if (!spec->multiout.dac_nids[i]) continue; nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i])); - if (i == 2) { + if (!pfx && i == 2) { /* Center/LFE */ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, "Center", @@ -5063,18 +5182,17 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, if (err < 0) return err; } else { - const char *pfx; - if (cfg->line_outs == 1 && - cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) - pfx = "Speaker"; - else - pfx = chname[i]; - err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, + const char *name = pfx; + if (!name) + name = chname[i]; + err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + name, i, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, + err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + name, i, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); if (err < 0) @@ -5154,7 +5272,8 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx, type, type_idx = 0; + int i, err, idx, type_idx = 0; + const char *prev_label = NULL; for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t pin; @@ -5164,12 +5283,13 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec, if (!alc_is_input_pin(codec, pin)) continue; - type = cfg->inputs[i].type; - if (i > 0 && type == cfg->inputs[i - 1].type) + label = hda_get_autocfg_input_label(codec, cfg, i); + if (prev_label && !strcmp(label, prev_label)) type_idx++; else type_idx = 0; - label = hda_get_autocfg_input_label(codec, cfg, i); + prev_label = label; + if (mixer) { idx = get_connection_index(codec, mixer, pin); if (idx >= 0) { @@ -7022,7 +7142,8 @@ enum { static const struct alc_fixup alc260_fixups[] = { [PINFIX_HP_DC5750] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x11, 0x90130110 }, /* speaker */ { } } @@ -7037,7 +7158,7 @@ static struct snd_pci_quirk alc260_fixup_tbl[] = { /* * ALC260 configurations */ -static const char *alc260_models[ALC260_MODEL_LAST] = { +static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_HP] = "hp", [ALC260_HP_3013] = "hp-3013", @@ -7233,8 +7354,10 @@ static int patch_alc260(struct hda_codec *codec) board_config = ALC260_AUTO; } - if (board_config == ALC260_AUTO) - alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 1); + if (board_config == ALC260_AUTO) { + alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC260_AUTO) { /* automatic parse from the BIOS config */ @@ -7282,8 +7405,7 @@ static int patch_alc260(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); - if (board_config == ALC260_AUTO) - alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); spec->vmaster_nid = 0x08; @@ -7405,7 +7527,7 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, - { "Int Mic", 0x1 }, + { "Internal Mic", 0x1 }, { "Line", 0x2 }, { "CD", 0x4 }, }, @@ -7415,7 +7537,7 @@ static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { .num_items = 2, .items = { { "Mic", 0x0 }, - { "Int Mic", 0x1 }, + { "Internal Mic", 0x1 }, }, }; @@ -7850,10 +7972,10 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -7877,8 +7999,8 @@ static struct snd_kcontrol_new alc885_mbp3_mixer[] = { HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT), { } /* end */ }; @@ -7895,8 +8017,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x19, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT), { } /* end */ }; @@ -7911,7 +8033,7 @@ static struct snd_kcontrol_new alc885_macmini3_mixer[] = { HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT), { } /* end */ }; @@ -7930,7 +8052,7 @@ static struct snd_kcontrol_new alc882_w2jc_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -7945,10 +8067,10 @@ static struct snd_kcontrol_new alc882_targa_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), { } /* end */ }; @@ -7968,7 +8090,7 @@ static struct snd_kcontrol_new alc882_asus_a7j_mixer[] = { HDA_CODEC_MUTE("Mobile Line Playback Switch", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), { } /* end */ }; @@ -7981,7 +8103,7 @@ static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -8762,10 +8884,10 @@ static struct snd_kcontrol_new alc883_mitac_mixer[] = { HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -8776,11 +8898,11 @@ static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -8790,11 +8912,11 @@ static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -8807,10 +8929,10 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -8830,10 +8952,10 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -8854,10 +8976,10 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -8878,10 +9000,10 @@ static struct snd_kcontrol_new alc885_8ch_intel_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x1b, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x1b, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -8901,10 +9023,10 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -8925,7 +9047,7 @@ static struct snd_kcontrol_new alc883_targa_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -8938,20 +9060,20 @@ static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -8962,7 +9084,7 @@ static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -8975,21 +9097,8 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -9036,7 +9145,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -9049,7 +9158,7 @@ static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -9071,10 +9180,10 @@ static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -9095,8 +9204,8 @@ static struct snd_kcontrol_new alc889A_mb31_mixer[] = { HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT), /* Boost mixers */ - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT), /* Input mixers */ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), @@ -9110,7 +9219,7 @@ static struct snd_kcontrol_new alc883_vaiott_mixer[] = { HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -9140,7 +9249,7 @@ static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), { } /* end */ }; @@ -9181,16 +9290,6 @@ static void alc883_mitac_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x17; } -/* auto-toggle front mic */ -/* -static void alc883_mitac_mic_automute(struct hda_codec *codec) -{ - unsigned char bits = snd_hda_jack_detect(codec, 0x18) ? HDA_AMP_MUTE : 0; - - snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); -} -*/ - static struct hda_verb alc883_mitac_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -9434,18 +9533,8 @@ static void alc883_lenovo_ms7195_unsol_event(struct hda_codec *codec, alc888_lenovo_ms7195_rca_automute(codec); } -static struct hda_verb alc883_medion_md2_verbs[] = { - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - { } /* end */ -}; - /* toggle speaker-output according to the hp-jack state */ -static void alc883_medion_md2_setup(struct hda_codec *codec) +static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9457,15 +9546,6 @@ static void alc883_medion_md2_setup(struct hda_codec *codec) #define alc883_targa_init_hook alc882_targa_init_hook #define alc883_targa_unsol_event alc882_targa_unsol_event -static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x18); - snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - static void alc883_clevo_m720_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9477,7 +9557,7 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec) static void alc883_clevo_m720_init_hook(struct hda_codec *codec) { alc_automute_amp(codec); - alc883_clevo_m720_mic_automute(codec); + alc88x_simple_mic_automute(codec); } static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, @@ -9485,7 +9565,7 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, { switch (res >> 26) { case ALC880_MIC_EVENT: - alc883_clevo_m720_mic_automute(codec); + alc88x_simple_mic_automute(codec); break; default: alc_automute_amp_unsol_event(codec, res); @@ -9701,7 +9781,7 @@ static hda_nid_t alc1200_slave_dig_outs[] = { /* * configuration and preset */ -static const char *alc882_models[ALC882_MODEL_LAST] = { +static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC882_3ST_DIG] = "3stack-dig", [ALC882_6ST_DIG] = "6stack-dig", [ALC882_ARIMA] = "arima", @@ -9730,7 +9810,6 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", [ALC883_MEDION] = "medion", - [ALC883_MEDION_MD2] = "medion-md2", [ALC883_MEDION_WIM2160] = "medion-wim2160", [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC883_LENOVO_101E_2ch] = "lenovo-101e", @@ -10378,19 +10457,6 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, - [ALC883_MEDION_MD2] = { - .mixers = { alc883_medion_md2_mixer}, - .init_verbs = { alc883_init_verbs, alc883_medion_md2_verbs}, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), - .channel_mode = alc883_3ST_2ch_modes, - .input_mux = &alc883_capture_source, - .unsol_event = alc_automute_amp_unsol_event, - .setup = alc883_medion_md2_setup, - .init_hook = alc_automute_amp, - }, [ALC883_MEDION_WIM2160] = { .mixers = { alc883_medion_wim2160_mixer }, .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs }, @@ -10467,7 +10533,7 @@ static struct alc_config_preset alc882_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_lenovo_nb0763_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .setup = alc883_medion_md2_setup, + .setup = alc883_lenovo_nb0763_setup, .init_hook = alc_automute_amp, }, [ALC888_LENOVO_MS7195_DIG] = { @@ -10666,7 +10732,8 @@ enum { static const struct alc_fixup alc882_fixups[] = { [PINFIX_ABIT_AW9D_MAX] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x15, 0x01080104 }, /* side */ { 0x16, 0x01011012 }, /* rear */ { 0x17, 0x01016011 }, /* clfe */ @@ -10674,13 +10741,15 @@ static const struct alc_fixup alc882_fixups[] = { } }, [PINFIX_PB_M5210] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, {} } }, [PINFIX_ACER_ASPIRE_7736] = { - .sku = ALC_FIXUP_SKU_IGNORE, + .type = ALC_FIXUP_SKU, + .v.sku = ALC_FIXUP_SKU_IGNORE, }, }; @@ -10830,17 +10899,29 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int i, err; + int type_idx = 0; hda_nid_t nid; + const char *prev_label = NULL; for (i = 0; i < cfg->num_inputs; i++) { if (cfg->inputs[i].type > AUTO_PIN_MIC) break; nid = cfg->inputs[i].pin; if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { - char label[32]; - snprintf(label, sizeof(label), "%s Boost", - hda_get_autocfg_input_label(codec, cfg, i)); - err = add_control(spec, ALC_CTL_WIDGET_VOL, label, 0, + const char *label; + char boost_label[32]; + + label = hda_get_autocfg_input_label(codec, cfg, i); + if (prev_label && !strcmp(label, prev_label)) + type_idx++; + else + type_idx = 0; + prev_label = label; + + snprintf(boost_label, sizeof(boost_label), + "%s Boost Volume", label); + err = add_control(spec, ALC_CTL_WIDGET_VOL, + boost_label, type_idx, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); if (err < 0) return err; @@ -10849,6 +10930,9 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) return 0; } +static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg); + /* almost identical with ALC880 parser... */ static int alc882_parse_auto_config(struct hda_codec *codec) { @@ -10866,7 +10950,10 @@ static int alc882_parse_auto_config(struct hda_codec *codec) err = alc880_auto_fill_dac_nids(spec, &spec->autocfg); if (err < 0) return err; - err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (codec->vendor_id == 0x10ec0887) + err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg); + else + err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0], @@ -10954,8 +11041,10 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC882_AUTO; } - if (board_config == ALC882_AUTO) - alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1); + if (board_config == ALC882_AUTO) { + alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } alc_auto_parse_customize_define(codec); @@ -11031,8 +11120,7 @@ static int patch_alc882(struct hda_codec *codec) if (has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - if (board_config == ALC882_AUTO) - alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); spec->vmaster_nid = 0x0c; @@ -11082,10 +11170,10 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), @@ -11186,10 +11274,10 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), @@ -11211,7 +11299,7 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x1a, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x1a, 0, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), @@ -11222,7 +11310,7 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Rear Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Rear Mic Boost Volume", 0x18, 0, HDA_INPUT), { } /* end */ }; @@ -11242,7 +11330,7 @@ static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), { } /* end */ }; @@ -11349,10 +11437,10 @@ static struct snd_kcontrol_new alc262_hippo_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -11366,10 +11454,10 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), { } /* end */ }; @@ -11437,10 +11525,10 @@ static struct snd_kcontrol_new alc262_tyan_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), { } /* end */ }; @@ -11624,7 +11712,7 @@ static struct snd_kcontrol_new alc262_nec_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -11679,7 +11767,7 @@ static struct hda_input_mux alc262_fujitsu_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, - { "Int Mic", 0x1 }, + { "Internal Mic", 0x1 }, { "CD", 0x4 }, }, }; @@ -11831,12 +11919,12 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -11867,12 +11955,12 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -11881,10 +11969,10 @@ static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), { } /* end */ }; @@ -11910,8 +11998,8 @@ static struct snd_kcontrol_new alc262_ultra_mixer[] = { HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Mic Boost", 0x15, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Mic Boost Volume", 0x15, 0, HDA_INPUT), { } /* end */ }; @@ -12081,13 +12169,8 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, spec->multiout.dac_nids = spec->private_dac_nids; spec->multiout.dac_nids[0] = 2; - if (!cfg->speaker_pins[0] && !cfg->hp_pins[0]) - pfx = "Master"; - else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) - pfx = "Speaker"; - else if (cfg->line_out_type == AUTO_PIN_HP_OUT) - pfx = "Headphone"; - else + pfx = alc_get_line_out_pfx(cfg, true); + if (!pfx) pfx = "Front"; for (i = 0; i < 2; i++) { err = alc262_add_out_sw_ctl(spec, cfg->line_out_pins[i], pfx, i); @@ -12427,19 +12510,14 @@ enum { static const struct alc_fixup alc262_fixups[] = { [PINFIX_FSC_H270] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x14, 0x99130110 }, /* speaker */ { 0x15, 0x0221142f }, /* front HP */ { 0x1b, 0x0121141f }, /* rear HP */ { } } }, - [PINFIX_PB_M5210] = { - .verbs = (const struct hda_verb[]) { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, - {} - } - }, }; static struct snd_pci_quirk alc262_fixup_tbl[] = { @@ -12529,7 +12607,7 @@ static void alc262_auto_init(struct hda_codec *codec) /* * configuration and preset */ -static const char *alc262_models[ALC262_MODEL_LAST] = { +static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_BASIC] = "basic", [ALC262_HIPPO] = "hippo", [ALC262_HIPPO_1] = "hippo_1", @@ -12870,8 +12948,10 @@ static int patch_alc262(struct hda_codec *codec) board_config = ALC262_AUTO; } - if (board_config == ALC262_AUTO) - alc_pick_fixup(codec, alc262_fixup_tbl, alc262_fixups, 1); + if (board_config == ALC262_AUTO) { + alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC262_AUTO) { /* automatic parse from the BIOS config */ @@ -12941,8 +13021,7 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->no_analog && has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - if (board_config == ALC262_AUTO) - alc_pick_fixup(codec, alc262_fixup_tbl, alc262_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); spec->vmaster_nid = 0x0c; @@ -12988,9 +13067,9 @@ static struct snd_kcontrol_new alc268_base_mixer[] = { HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT), { } }; @@ -12999,9 +13078,9 @@ static struct snd_kcontrol_new alc268_toshiba_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT), { } }; @@ -13105,9 +13184,9 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { .put = alc268_acer_master_sw_put, .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), }, - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT), { } }; @@ -13123,8 +13202,8 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { .put = alc268_acer_master_sw_put, .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), }, - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT), { } }; @@ -13216,8 +13295,8 @@ static struct snd_kcontrol_new alc268_dell_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), { } }; @@ -13250,8 +13329,8 @@ static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT), - HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), { } }; @@ -13716,7 +13795,7 @@ static void alc268_auto_init(struct hda_codec *codec) /* * configuration and preset */ -static const char *alc268_models[ALC268_MODEL_LAST] = { +static const char * const alc268_models[ALC268_MODEL_LAST] = { [ALC267_QUANTA_IL1] = "quanta-il1", [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", @@ -14074,10 +14153,10 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), { } /* end */ @@ -14097,10 +14176,10 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { }, HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), { } }; @@ -14118,13 +14197,13 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { }, HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x1b, 0, HDA_INPUT), { } }; @@ -14154,30 +14233,30 @@ static struct snd_kcontrol_new alc269_asus_mixer[] = { static struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("IntMic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), { } /* end */ }; static struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), { } /* end */ }; static struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("IntMic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), { } /* end */ }; static struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), { } /* end */ }; @@ -14796,31 +14875,91 @@ static int alc269_resume(struct hda_codec *codec) } #endif /* SND_HDA_NEEDS_RESUME */ +static void alc269_fixup_hweq(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + int coef; + + if (action != ALC_FIXUP_ACT_INIT) + return; + coef = alc_read_coef_idx(codec, 0x1e); + alc_write_coef_idx(codec, 0x1e, coef | 0x80); +} + enum { ALC269_FIXUP_SONY_VAIO, + ALC275_FIXUP_SONY_VAIO_GPIO2, ALC269_FIXUP_DELL_M101Z, + ALC269_FIXUP_SKU_IGNORE, + ALC269_FIXUP_ASUS_G73JW, + ALC269_FIXUP_LENOVO_EAPD, + ALC275_FIXUP_SONY_HWEQ, }; static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_SONY_VAIO] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, {} } }, + [ALC275_FIXUP_SONY_VAIO_GPIO2] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x01, AC_VERB_SET_GPIO_MASK, 0x04}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_SONY_VAIO + }, [ALC269_FIXUP_DELL_M101Z] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { /* Enables internal speaker */ {0x20, AC_VERB_SET_COEF_INDEX, 13}, {0x20, AC_VERB_SET_PROC_COEF, 0x4040}, {} } }, + [ALC269_FIXUP_SKU_IGNORE] = { + .type = ALC_FIXUP_SKU, + .v.sku = ALC_FIXUP_SKU_IGNORE, + }, + [ALC269_FIXUP_ASUS_G73JW] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x17, 0x99130111 }, /* subwoofer */ + { } + } + }, + [ALC269_FIXUP_LENOVO_EAPD] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0}, + {} + } + }, + [ALC275_FIXUP_SONY_HWEQ] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_hweq, + .chained = true, + .chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2 + } }; static struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), + SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), + SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), + SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), {} }; @@ -14828,7 +14967,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = { /* * configuration and preset */ -static const char *alc269_models[ALC269_MODEL_LAST] = { +static const char * const alc269_models[ALC269_MODEL_LAST] = { [ALC269_BASIC] = "basic", [ALC269_QUANTA_FL1] = "quanta", [ALC269_AMIC] = "laptop-amic", @@ -15070,28 +15209,29 @@ static int patch_alc269(struct hda_codec *codec) alc_auto_parse_customize_define(codec); - coef = alc_read_coef_idx(codec, 0); - if ((coef & 0x00f0) == 0x0010) { - if (codec->bus->pci->subsystem_vendor == 0x1025 && - spec->cdefine.platform_type == 1) { - alc_codec_rename(codec, "ALC271X"); - spec->codec_variant = ALC269_TYPE_ALC271X; - } else if ((coef & 0xf000) == 0x1000) { - spec->codec_variant = ALC269_TYPE_ALC270; - } else if ((coef & 0xf000) == 0x2000) { - alc_codec_rename(codec, "ALC259"); - spec->codec_variant = ALC269_TYPE_ALC259; - } else if ((coef & 0xf000) == 0x3000) { - alc_codec_rename(codec, "ALC258"); - spec->codec_variant = ALC269_TYPE_ALC258; - } else { - alc_codec_rename(codec, "ALC269VB"); - spec->codec_variant = ALC269_TYPE_ALC269VB; - } - } else - alc_fix_pll_init(codec, 0x20, 0x04, 15); - - alc269_fill_coef(codec); + if (codec->vendor_id == 0x10ec0269) { + coef = alc_read_coef_idx(codec, 0); + if ((coef & 0x00f0) == 0x0010) { + if (codec->bus->pci->subsystem_vendor == 0x1025 && + spec->cdefine.platform_type == 1) { + alc_codec_rename(codec, "ALC271X"); + spec->codec_variant = ALC269_TYPE_ALC271X; + } else if ((coef & 0xf000) == 0x1000) { + spec->codec_variant = ALC269_TYPE_ALC270; + } else if ((coef & 0xf000) == 0x2000) { + alc_codec_rename(codec, "ALC259"); + spec->codec_variant = ALC269_TYPE_ALC259; + } else if ((coef & 0xf000) == 0x3000) { + alc_codec_rename(codec, "ALC258"); + spec->codec_variant = ALC269_TYPE_ALC258; + } else { + alc_codec_rename(codec, "ALC269VB"); + spec->codec_variant = ALC269_TYPE_ALC269VB; + } + } else + alc_fix_pll_init(codec, 0x20, 0x04, 15); + alc269_fill_coef(codec); + } board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, alc269_models, @@ -15103,8 +15243,10 @@ static int patch_alc269(struct hda_codec *codec) board_config = ALC269_AUTO; } - if (board_config == ALC269_AUTO) - alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1); + if (board_config == ALC269_AUTO) { + alc_pick_fixup(codec, NULL, alc269_fixup_tbl, alc269_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC269_AUTO) { /* automatic parse from the BIOS config */ @@ -15165,8 +15307,7 @@ static int patch_alc269(struct hda_codec *codec) if (has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); - if (board_config == ALC269_AUTO) - alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); spec->vmaster_nid = 0x02; @@ -15854,41 +15995,33 @@ static int alc861_auto_fill_dac_nids(struct hda_codec *codec, return 0; } -static int alc861_create_out_sw(struct hda_codec *codec, const char *pfx, - hda_nid_t nid, unsigned int chs) +static int __alc861_create_out_sw(struct hda_codec *codec, const char *pfx, + hda_nid_t nid, int idx, unsigned int chs) { - return add_pb_sw_ctrl(codec->spec, ALC_CTL_WIDGET_MUTE, pfx, + return __add_pb_sw_ctrl(codec->spec, ALC_CTL_WIDGET_MUTE, pfx, idx, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); } +#define alc861_create_out_sw(codec, pfx, nid, chs) \ + __alc861_create_out_sw(codec, pfx, nid, 0, chs) + /* add playback controls from the parsed DAC table */ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { struct alc_spec *spec = codec->spec; - static const char *chname[4] = { + static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; + const char *pfx = alc_get_line_out_pfx(cfg, true); hda_nid_t nid; int i, err; - if (cfg->line_outs == 1) { - const char *pfx = NULL; - if (!cfg->hp_outs) - pfx = "Master"; - else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) - pfx = "Speaker"; - if (pfx) { - nid = spec->multiout.dac_nids[0]; - return alc861_create_out_sw(codec, pfx, nid, 3); - } - } - for (i = 0; i < cfg->line_outs; i++) { nid = spec->multiout.dac_nids[i]; if (!nid) continue; - if (i == 2) { + if (!pfx && i == 2) { /* Center/LFE */ err = alc861_create_out_sw(codec, "Center", nid, 1); if (err < 0) @@ -15897,7 +16030,10 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec, if (err < 0) return err; } else { - err = alc861_create_out_sw(codec, chname[i], nid, 3); + const char *name = pfx; + if (!name) + name = chname[i]; + err = __alc861_create_out_sw(codec, name, nid, i, 3); if (err < 0) return err; } @@ -16080,7 +16216,7 @@ static struct hda_amp_list alc861_loopbacks[] = { /* * configuration and preset */ -static const char *alc861_models[ALC861_MODEL_LAST] = { +static const char * const alc861_models[ALC861_MODEL_LAST] = { [ALC861_3ST] = "3stack", [ALC660_3ST] = "3stack-660", [ALC861_3ST_DIG] = "3stack-dig", @@ -16230,7 +16366,8 @@ enum { static const struct alc_fixup alc861_fixups[] = { [PINFIX_FSC_AMILO_PI1505] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x0b, 0x0221101f }, /* HP */ { 0x0f, 0x90170310 }, /* speaker */ { } @@ -16265,8 +16402,10 @@ static int patch_alc861(struct hda_codec *codec) board_config = ALC861_AUTO; } - if (board_config == ALC861_AUTO) - alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1); + if (board_config == ALC861_AUTO) { + alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC861_AUTO) { /* automatic parse from the BIOS config */ @@ -16303,8 +16442,7 @@ static int patch_alc861(struct hda_codec *codec) spec->vmaster_nid = 0x03; - if (board_config == ALC861_AUTO) - alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) { @@ -16369,8 +16507,8 @@ static struct hda_input_mux alc861vd_capture_source = { static struct hda_input_mux alc861vd_dallas_capture_source = { .num_items = 2, .items = { - { "Ext Mic", 0x0 }, - { "Int Mic", 0x1 }, + { "Mic", 0x0 }, + { "Internal Mic", 0x1 }, }, }; @@ -16449,11 +16587,11 @@ static struct snd_kcontrol_new alc861vd_6st_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), @@ -16472,11 +16610,11 @@ static struct snd_kcontrol_new alc861vd_3st_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), @@ -16496,11 +16634,11 @@ static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), @@ -16511,19 +16649,19 @@ static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = { }; /* Pin assignment: Speaker=0x14, HP = 0x15, - * Ext Mic=0x18, Int Mic = 0x19, CD = 0x1c, PC Beep = 0x1d + * Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d */ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -16688,18 +16826,6 @@ static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {} }; -static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x18); - bits = present ? HDA_AMP_MUTE : 0; - - snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); -} - static void alc861vd_lenovo_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -16710,7 +16836,7 @@ static void alc861vd_lenovo_setup(struct hda_codec *codec) static void alc861vd_lenovo_init_hook(struct hda_codec *codec) { alc_automute_amp(codec); - alc861vd_lenovo_mic_automute(codec); + alc88x_simple_mic_automute(codec); } static void alc861vd_lenovo_unsol_event(struct hda_codec *codec, @@ -16718,7 +16844,7 @@ static void alc861vd_lenovo_unsol_event(struct hda_codec *codec, { switch (res >> 26) { case ALC880_MIC_EVENT: - alc861vd_lenovo_mic_automute(codec); + alc88x_simple_mic_automute(codec); break; default: alc_automute_amp_unsol_event(codec, res); @@ -16793,7 +16919,7 @@ static void alc861vd_dallas_setup(struct hda_codec *codec) /* * configuration and preset */ -static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { +static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = { [ALC660VD_3ST] = "3stack-660", [ALC660VD_3ST_DIG] = "3stack-660-digout", [ALC660VD_ASUS_V1S] = "asus-v1s", @@ -17008,12 +17134,15 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) #define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c) /* add playback controls from the parsed DAC table */ -/* Based on ALC880 version. But ALC861VD has separate, +/* Based on ALC880 version. But ALC861VD and ALC887 have separate, * different NIDs for mute/unmute switch and volume control */ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - static const char *chname[4] = {"Front", "Surround", "CLFE", "Side"}; + static const char * const chname[4] = { + "Front", "Surround", "CLFE", "Side" + }; + const char *pfx = alc_get_line_out_pfx(cfg, true); hda_nid_t nid_v, nid_s; int i, err; @@ -17027,7 +17156,7 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, alc880_dac_to_idx( spec->multiout.dac_nids[i])); - if (i == 2) { + if (!pfx && i == 2) { /* Center/LFE */ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, "Center", @@ -17054,24 +17183,17 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, if (err < 0) return err; } else { - const char *pfx; - if (cfg->line_outs == 1 && - cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { - if (!cfg->hp_pins) - pfx = "Speaker"; - else - pfx = "PCM"; - } else - pfx = chname[i]; - err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, + const char *name = pfx; + if (!name) + name = chname[i]; + err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + name, i, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - if (cfg->line_outs == 1 && - cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) - pfx = "Speaker"; - err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, + err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + name, i, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); if (err < 0) @@ -17204,7 +17326,8 @@ enum { /* reset GPIO1 */ static const struct alc_fixup alc861vd_fixups[] = { [ALC660VD_FIX_ASUS_GPIO1] = { - .verbs = (const struct hda_verb[]) { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, @@ -17239,8 +17362,10 @@ static int patch_alc861vd(struct hda_codec *codec) board_config = ALC861VD_AUTO; } - if (board_config == ALC861VD_AUTO) - alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1); + if (board_config == ALC861VD_AUTO) { + alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + } if (board_config == ALC861VD_AUTO) { /* automatic parse from the BIOS config */ @@ -17288,8 +17413,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->vmaster_nid = 0x02; - if (board_config == ALC861VD_AUTO) - alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -17535,13 +17659,13 @@ static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, - HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("e-Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("i-Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -17685,8 +17809,8 @@ static struct snd_kcontrol_new alc663_g71v_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -17697,8 +17821,8 @@ static struct snd_kcontrol_new alc663_g50v_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), { } /* end */ @@ -18531,13 +18655,13 @@ static struct snd_kcontrol_new alc662_ecs_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, - HDA_CODEC_VOLUME("e-Mic/LineIn Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("e-Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("e-Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic/LineIn Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("i-Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), { } /* end */ }; @@ -18548,13 +18672,13 @@ static struct snd_kcontrol_new alc272_nc10_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), { } /* end */ }; @@ -18572,7 +18696,7 @@ static struct snd_kcontrol_new alc272_nc10_mixer[] = { /* * configuration and preset */ -static const char *alc662_models[ALC662_MODEL_LAST] = { +static const char * const alc662_models[ALC662_MODEL_LAST] = { [ALC662_3ST_2ch_DIG] = "3stack-dig", [ALC662_3ST_6ch_DIG] = "3stack-6ch-dig", [ALC662_3ST_6ch] = "3stack-6ch", @@ -19059,20 +19183,24 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec, return 0; } -static inline int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx, - hda_nid_t nid, unsigned int chs) +static inline int __alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx, + hda_nid_t nid, int idx, unsigned int chs) { - return add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, + return __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, idx, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); } -static inline int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx, - hda_nid_t nid, unsigned int chs) +static inline int __alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx, + hda_nid_t nid, int idx, unsigned int chs) { - return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, + return __add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, idx, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT)); } +#define alc662_add_vol_ctl(spec, pfx, nid, chs) \ + __alc662_add_vol_ctl(spec, pfx, nid, 0, chs) +#define alc662_add_sw_ctl(spec, pfx, nid, chs) \ + __alc662_add_sw_ctl(spec, pfx, nid, 0, chs) #define alc662_add_stereo_vol(spec, pfx, nid) \ alc662_add_vol_ctl(spec, pfx, nid, 3) #define alc662_add_stereo_sw(spec, pfx, nid) \ @@ -19083,9 +19211,10 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { struct alc_spec *spec = codec->spec; - static const char *chname[4] = { + static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; + const char *pfx = alc_get_line_out_pfx(cfg, true); hda_nid_t nid, mix; int i, err; @@ -19096,7 +19225,7 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid); if (!mix) continue; - if (i == 2) { + if (!pfx && i == 2) { /* Center/LFE */ err = alc662_add_vol_ctl(spec, "Center", nid, 1); if (err < 0) @@ -19111,22 +19240,13 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, if (err < 0) return err; } else { - const char *pfx; - if (cfg->line_outs == 1 && - cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { - if (cfg->hp_outs) - pfx = "Speaker"; - else - pfx = "PCM"; - } else - pfx = chname[i]; - err = alc662_add_vol_ctl(spec, pfx, nid, 3); + const char *name = pfx; + if (!name) + name = chname[i]; + err = __alc662_add_vol_ctl(spec, name, nid, i, 3); if (err < 0) return err; - if (cfg->line_outs == 1 && - cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) - pfx = "Speaker"; - err = alc662_add_sw_ctl(spec, pfx, mix, 3); + err = __alc662_add_sw_ctl(spec, name, mix, i, 3); if (err < 0) return err; } @@ -19323,24 +19443,45 @@ static void alc662_auto_init(struct hda_codec *codec) alc_inithook(codec); } +static void alc272_fixup_mario(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action != ALC_FIXUP_ACT_PROBE) + return; + if (snd_hda_override_amp_caps(codec, 0x2, HDA_OUTPUT, + (0x3b << AC_AMPCAP_OFFSET_SHIFT) | + (0x3b << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x03 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT))) + printk(KERN_WARNING + "hda_codec: failed to override amp caps for NID 0x2\n"); +} + enum { ALC662_FIXUP_ASPIRE, ALC662_FIXUP_IDEAPAD, + ALC272_FIXUP_MARIO, }; static const struct alc_fixup alc662_fixups[] = { [ALC662_FIXUP_ASPIRE] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x15, 0x99130112 }, /* subwoofer */ { } } }, [ALC662_FIXUP_IDEAPAD] = { - .pins = (const struct alc_pincfg[]) { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { { 0x17, 0x99130112 }, /* subwoofer */ { } } }, + [ALC272_FIXUP_MARIO] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc272_fixup_mario, + } }; static struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -19351,6 +19492,10 @@ static struct snd_pci_quirk alc662_fixup_tbl[] = { {} }; +static const struct alc_model_fixup alc662_fixup_models[] = { + {.id = ALC272_FIXUP_MARIO, .name = "mario"}, + {} +}; static int patch_alc662(struct hda_codec *codec) @@ -19389,7 +19534,9 @@ static int patch_alc662(struct hda_codec *codec) } if (board_config == ALC662_AUTO) { - alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 1); + alc_pick_fixup(codec, alc662_fixup_models, + alc662_fixup_tbl, alc662_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) { @@ -19447,11 +19594,11 @@ static int patch_alc662(struct hda_codec *codec) } spec->vmaster_nid = 0x02; + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + codec->patch_ops = alc_patch_ops; - if (board_config == ALC662_AUTO) { + if (board_config == ALC662_AUTO) spec->init_hook = alc662_auto_init; - alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 0); - } alc_init_jacks(codec); @@ -19577,9 +19724,9 @@ static struct snd_kcontrol_new alc680_base_mixer[] = { HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Int Mic Boost", 0x12, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x12, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost Volume", 0x19, 0, HDA_INPUT), { } }; @@ -19839,7 +19986,7 @@ static void alc680_auto_init(struct hda_codec *codec) /* * configuration and preset */ -static const char *alc680_models[ALC680_MODEL_LAST] = { +static const char * const alc680_models[ALC680_MODEL_LAST] = { [ALC680_BASE] = "base", [ALC680_AUTO] = "auto", }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index efa4225f5fd6..9ea48b425d0b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -266,7 +266,7 @@ struct sigmatel_spec { struct sigmatel_mic_route int_mic; struct sigmatel_mic_route dock_mic; - const char **spdif_labels; + const char * const *spdif_labels; hda_nid_t dig_in_nid; hda_nid_t mono_nid; @@ -389,6 +389,9 @@ static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = { 0x11, 0x20, 0 }; +#define STAC92HD88XXX_NUM_DMICS STAC92HD83XXX_NUM_DMICS +#define stac92hd88xxx_dmic_nids stac92hd83xxx_dmic_nids + #define STAC92HD87B_NUM_DMICS 1 static hda_nid_t stac92hd87b_dmic_nids[STAC92HD87B_NUM_DMICS + 1] = { 0x11, 0 @@ -521,7 +524,7 @@ static unsigned long stac927x_capsws[] = { HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), }; -static const char *stac927x_spdif_labels[5] = { +static const char * const stac927x_spdif_labels[5] = { "Digital Playback", "ADAT", "Analog Mux 1", "Analog Mux 2", "Analog Mux 3" }; @@ -1059,7 +1062,7 @@ static struct snd_kcontrol_new stac_smux_mixer = { .put = stac92xx_smux_enum_put, }; -static const char *slave_vols[] = { +static const char * const slave_vols[] = { "Front Playback Volume", "Surround Playback Volume", "Center Playback Volume", @@ -1070,7 +1073,7 @@ static const char *slave_vols[] = { NULL }; -static const char *slave_sws[] = { +static const char * const slave_sws[] = { "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch", @@ -1351,7 +1354,7 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_9200_PANASONIC] = ref9200_pin_configs, }; -static const char *stac9200_models[STAC_9200_MODELS] = { +static const char * const stac9200_models[STAC_9200_MODELS] = { [STAC_AUTO] = "auto", [STAC_REF] = "ref", [STAC_9200_OQO] = "oqo", @@ -1497,7 +1500,7 @@ static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = { [STAC_M6] = stac925xM6_pin_configs, }; -static const char *stac925x_models[STAC_925x_MODELS] = { +static const char * const stac925x_models[STAC_925x_MODELS] = { [STAC_925x_AUTO] = "auto", [STAC_REF] = "ref", [STAC_M1] = "m1", @@ -1571,7 +1574,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_INTEL] = intel_dg45id_pin_configs, }; -static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { +static const char * const stac92hd73xx_models[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_AUTO] = "auto", [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", @@ -1657,7 +1660,7 @@ static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs, }; -static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { +static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_AUTO] = "auto", [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", @@ -1719,7 +1722,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_HP_DV4_1222NR] = NULL, }; -static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { +static const char * const stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_AUTO] = "auto", [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", @@ -1912,7 +1915,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_922X_DELL_M82] = dell_922x_m82_pin_configs, }; -static const char *stac922x_models[STAC_922X_MODELS] = { +static const char * const stac922x_models[STAC_922X_MODELS] = { [STAC_922X_AUTO] = "auto", [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", @@ -2074,7 +2077,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { [STAC_927X_VOLKNOB] = NULL, }; -static const char *stac927x_models[STAC_927X_MODELS] = { +static const char * const stac927x_models[STAC_927X_MODELS] = { [STAC_927X_AUTO] = "auto", [STAC_D965_REF_NO_JD] = "ref-no-jd", [STAC_D965_REF] = "ref", @@ -2177,7 +2180,7 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { [STAC_9205_EAPD] = NULL, }; -static const char *stac9205_models[STAC_9205_MODELS] = { +static const char * const stac9205_models[STAC_9205_MODELS] = { [STAC_9205_AUTO] = "auto", [STAC_9205_REF] = "ref", [STAC_9205_DELL_M42] = "dell-m42", @@ -3120,7 +3123,7 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, int type) { struct sigmatel_spec *spec = codec->spec; - static const char *chname[4] = { + static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; hda_nid_t nid; @@ -3253,7 +3256,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, } /* labels for mono mux outputs */ -static const char *stac92xx_mono_labels[4] = { +static const char * const stac92xx_mono_labels[4] = { "DAC0", "DAC1", "Mixer", "DAC2" }; @@ -3377,7 +3380,7 @@ static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) return 0; }; -static const char *stac92xx_spdif_labels[3] = { +static const char * const stac92xx_spdif_labels[3] = { "Digital Playback", "Analog Mux 1", "Analog Mux 2", }; @@ -3385,7 +3388,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *spdif_mux = &spec->private_smux; - const char **labels = spec->spdif_labels; + const char * const *labels = spec->spdif_labels; int i, num_cons; hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; @@ -3406,7 +3409,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) } /* labels for dmic mux inputs */ -static const char *stac92xx_dmic_labels[5] = { +static const char * const stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", "Digital Mic 3", "Digital Mic 4" }; @@ -3481,6 +3484,8 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, label = hda_get_input_pin_label(codec, nid, 1); snd_hda_add_imux_item(dimux, label, index, &type_idx); + if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) + snd_hda_add_imux_item(imux, label, index, &type_idx); err = create_elem_capture_vol(codec, nid, label, type_idx, HDA_INPUT); @@ -3492,9 +3497,6 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, if (err < 0) return err; } - - if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) - snd_hda_add_imux_item(imux, label, index, NULL); } return 0; @@ -3592,7 +3594,7 @@ static int stac_check_auto_mic(struct hda_codec *codec) if (check_mic_pin(codec, spec->dmic_nids[i], &fixed, &ext, &dock)) return 0; - if (!fixed && !ext && !dock) + if (!fixed || (!ext && !dock)) return 0; /* no input to switch */ if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP)) return 0; /* no unsol support */ @@ -5331,7 +5333,7 @@ again: return 0; } -static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) +static int hp_bnb2011_with_dock(struct hda_codec *codec) { if (codec->vendor_id != 0x111d7605 && codec->vendor_id != 0x111d76d1) @@ -5346,10 +5348,6 @@ static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) case 0x103c161d: case 0x103c161e: case 0x103c161f: - case 0x103c1620: - case 0x103c1621: - case 0x103c1622: - case 0x103c1623: case 0x103c162a: case 0x103c162b: @@ -5358,41 +5356,9 @@ static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) case 0x103c1631: case 0x103c1633: - + case 0x103c1634: case 0x103c1635: - case 0x103c164f: - - case 0x103c1676: - case 0x103c1677: - case 0x103c1678: - case 0x103c1679: - case 0x103c167a: - case 0x103c167b: - case 0x103c167c: - case 0x103c167d: - case 0x103c167e: - case 0x103c167f: - case 0x103c1680: - case 0x103c1681: - case 0x103c1682: - case 0x103c1683: - case 0x103c1684: - case 0x103c1685: - case 0x103c1686: - case 0x103c1687: - case 0x103c1688: - case 0x103c1689: - case 0x103c168a: - case 0x103c168b: - case 0x103c168c: - case 0x103c168d: - case 0x103c168e: - case 0x103c168f: - case 0x103c1690: - case 0x103c1691: - case 0x103c1692: - case 0x103c3587: case 0x103c3588: case 0x103c3589: @@ -5400,9 +5366,9 @@ static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) case 0x103c3667: case 0x103c3668: - /* set BTL amp level to 13.43dB for louder speaker output */ - return snd_hda_codec_write_cache(codec, codec->afg, 0, - 0x7F4, 0x14); + case 0x103c3669: + + return 1; } return 0; } @@ -5418,12 +5384,17 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + if (hp_bnb2011_with_dock(codec)) { + snd_hda_codec_set_pincfg(codec, 0xa, 0x2101201f); + snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e); + } + /* reset pin power-down; Windows may leave these bits after reboot */ snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7EC, 0); snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7ED, 0); codec->no_trigger_sense = 1; codec->spec = spec; - spec->linear_tone_beep = 1; + spec->linear_tone_beep = 0; codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->digbeep_nid = 0x21; spec->dmic_nids = stac92hd83xxx_dmic_nids; @@ -5463,15 +5434,21 @@ again: spec->num_dmics = stac92xx_connected_ports(codec, stac92hd87b_dmic_nids, STAC92HD87B_NUM_DMICS); - /* Fall through */ + spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids); + spec->pin_nids = stac92hd88xxx_pin_nids; + spec->mono_nid = 0; + spec->num_pwrs = 0; + break; case 0x111d7666: case 0x111d7667: case 0x111d7668: case 0x111d7669: + spec->num_dmics = stac92xx_connected_ports(codec, + stac92hd88xxx_dmic_nids, + STAC92HD88XXX_NUM_DMICS); spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids); spec->pin_nids = stac92hd88xxx_pin_nids; spec->mono_nid = 0; - spec->digbeep_nid = 0; spec->num_pwrs = 0; break; case 0x111d7604: @@ -5538,8 +5515,6 @@ again: AC_VERB_SET_CONNECT_SEL, num_dacs); } - stac92hd83xxx_set_system_btl_amp(codec); - codec->proc_widget_hook = stac92hd_proc_hook; return 0; @@ -6262,7 +6237,7 @@ static unsigned int stac9872_vaio_pin_configs[9] = { 0x90a7013e }; -static const char *stac9872_models[STAC_9872_MODELS] = { +static const char * const stac9872_models[STAC_9872_MODELS] = { [STAC_9872_AUTO] = "auto", [STAC_9872_VAIO] = "vaio", }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 7f4852a478a1..a76c3260d941 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2281,7 +2281,9 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b}; int i, err; @@ -2370,7 +2372,7 @@ static void create_hp_imux(struct via_spec *spec) { int i; struct hda_input_mux *imux = &spec->private_imux[1]; - static const char *texts[] = { "OFF", "ON", NULL}; + static const char * const texts[] = { "OFF", "ON", NULL}; /* for hp mode select */ for (i = 0; texts[i]; i++) @@ -2890,7 +2892,9 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29}; int i, err; @@ -3433,7 +3437,9 @@ static int vt1708B_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid_vols[] = {0x16, 0x18, 0x26, 0x27}; hda_nid_t nid, nid_vol = 0; int i, err; @@ -3861,7 +3867,9 @@ static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid_vols[] = {0x10, 0x11, 0x24, 0x25}; hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x26, 0x27}; hda_nid_t nid, nid_vol, nid_mute; @@ -4304,7 +4312,7 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { int err, i; struct hda_input_mux *imux; - static const char *texts[] = { "ON", "OFF", NULL}; + static const char * const texts[] = { "ON", "OFF", NULL}; if (!pin) return 0; spec->multiout.hp_nid = 0x1D; @@ -4615,7 +4623,9 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + static const char * const chname[4] = { + "Front", "Surround", "C/LFE", "Side" + }; hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb}; hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27}; hda_nid_t nid, nid_vol, nid_mute = 0; @@ -5064,7 +5074,9 @@ static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { char name[32]; - static const char *chname[3] = { "Front", "Surround", "C/LFE" }; + static const char * const chname[3] = { + "Front", "Surround", "C/LFE" + }; hda_nid_t nid_vols[] = {0x10, 0x11, 0x25}; hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27}; hda_nid_t nid, nid_vol, nid_mute; diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 712c1710f9a2..7b62de089fee 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -96,6 +96,11 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice) tmp |= ICE1712_DELTA_AP_CCLK | ICE1712_DELTA_AP_CS_CODEC; tmp &= ~ICE1712_DELTA_AP_CS_DIGITAL; break; + case ICE1712_SUBDEVICE_DELTA66E: + tmp |= ICE1712_DELTA_66E_CCLK | ICE1712_DELTA_66E_CS_CHIP_A | + ICE1712_DELTA_66E_CS_CHIP_B; + tmp &= ~ICE1712_DELTA_66E_CS_CS8427; + break; case ICE1712_SUBDEVICE_VX442: tmp |= ICE1712_VX442_CCLK | ICE1712_VX442_CODEC_CHIP_A | ICE1712_VX442_CODEC_CHIP_B; tmp &= ~ICE1712_VX442_CS_DIGITAL; @@ -119,6 +124,9 @@ static void ap_cs8427_codec_deassert(struct snd_ice1712 *ice, unsigned char tmp) case ICE1712_SUBDEVICE_DELTA410: tmp |= ICE1712_DELTA_AP_CS_DIGITAL; break; + case ICE1712_SUBDEVICE_DELTA66E: + tmp |= ICE1712_DELTA_66E_CS_CS8427; + break; case ICE1712_SUBDEVICE_VX442: tmp |= ICE1712_VX442_CS_DIGITAL; break; @@ -276,6 +284,20 @@ static void delta1010lt_ak4524_lock(struct snd_akm4xxx *ak, int chip) } /* + * AK4524 on Delta66 rev E to choose the chip address + */ +static void delta66e_ak4524_lock(struct snd_akm4xxx *ak, int chip) +{ + struct snd_ak4xxx_private *priv = (void *)ak->private_value[0]; + struct snd_ice1712 *ice = ak->private_data[0]; + + snd_ice1712_save_gpio_status(ice); + priv->cs_mask = + priv->cs_addr = chip == 0 ? ICE1712_DELTA_66E_CS_CHIP_A : + ICE1712_DELTA_66E_CS_CHIP_B; +} + +/* * AK4528 on VX442 to choose the chip mask */ static void vx442_ak4524_lock(struct snd_akm4xxx *ak, int chip) @@ -487,6 +509,29 @@ static struct snd_ak4xxx_private akm_delta1010lt_priv __devinitdata = { .mask_flags = 0, }; +static struct snd_akm4xxx akm_delta66e __devinitdata = { + .type = SND_AK4524, + .num_adcs = 4, + .num_dacs = 4, + .ops = { + .lock = delta66e_ak4524_lock, + .set_rate_val = delta_ak4524_set_rate_val + } +}; + +static struct snd_ak4xxx_private akm_delta66e_priv __devinitdata = { + .caddr = 2, + .cif = 0, /* the default level of the CIF pin from AK4524 */ + .data_mask = ICE1712_DELTA_66E_DOUT, + .clk_mask = ICE1712_DELTA_66E_CCLK, + .cs_mask = 0, + .cs_addr = 0, /* set later */ + .cs_none = 0, + .add_flags = 0, + .mask_flags = 0, +}; + + static struct snd_akm4xxx akm_delta44 __devinitdata = { .type = SND_AK4524, .num_adcs = 4, @@ -644,9 +689,11 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_init(ak, &akm_delta44, &akm_delta44_priv, ice); break; case ICE1712_SUBDEVICE_VX442: - case ICE1712_SUBDEVICE_DELTA66E: err = snd_ice1712_akm4xxx_init(ak, &akm_vx442, &akm_vx442_priv, ice); break; + case ICE1712_SUBDEVICE_DELTA66E: + err = snd_ice1712_akm4xxx_init(ak, &akm_delta66e, &akm_delta66e_priv, ice); + break; default: snd_BUG(); return -EINVAL; diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index 1a0ac6cd6501..11a9c3a76507 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -144,6 +144,17 @@ extern struct snd_ice1712_card_info snd_ice1712_delta_cards[]; #define ICE1712_DELTA_1010LT_CS_NONE 0x50 /* nothing */ #define ICE1712_DELTA_1010LT_WORDCLOCK 0x80 /* sample clock source: 0 = Word Clock Input, 1 = S/PDIF Input ??? */ +/* M-Audio Delta 66 rev. E definitions. + * Newer revisions of Delta 66 have CS8427 over SPI for + * S/PDIF transceiver instead of CS8404/CS8414. */ +/* 0x01 = DFS */ +#define ICE1712_DELTA_66E_CCLK 0x02 /* SPI clock */ +#define ICE1712_DELTA_66E_DIN 0x04 /* data input */ +#define ICE1712_DELTA_66E_DOUT 0x08 /* data output */ +#define ICE1712_DELTA_66E_CS_CS8427 0x10 /* chip select, low = CS8427 */ +#define ICE1712_DELTA_66E_CS_CHIP_A 0x20 /* AK4524 #0 */ +#define ICE1712_DELTA_66E_CS_CHIP_B 0x40 /* AK4524 #1 */ + /* Digigram VX442 definitions */ #define ICE1712_VX442_CCLK 0x02 /* SPI clock */ #define ICE1712_VX442_DIN 0x04 /* data input */ diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index acd8f15f7bff..0f8726551fde 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -1,10 +1,8 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o -snd-hifier-objs := hifier.o -snd-oxygen-objs := oxygen.o +snd-oxygen-objs := oxygen.o xonar_dg.o snd-virtuoso-objs := virtuoso.o xonar_lib.o \ xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.o obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o -obj-$(CONFIG_SND_HIFIER) += snd-hifier.o obj-$(CONFIG_SND_OXYGEN) += snd-oxygen.o obj-$(CONFIG_SND_VIRTUOSO) += snd-virtuoso.o diff --git a/sound/pci/oxygen/cs4245.h b/sound/pci/oxygen/cs4245.h new file mode 100644 index 000000000000..5e0197e07dd1 --- /dev/null +++ b/sound/pci/oxygen/cs4245.h @@ -0,0 +1,107 @@ +#define CS4245_CHIP_ID 0x01 +#define CS4245_POWER_CTRL 0x02 +#define CS4245_DAC_CTRL_1 0x03 +#define CS4245_ADC_CTRL 0x04 +#define CS4245_MCLK_FREQ 0x05 +#define CS4245_SIGNAL_SEL 0x06 +#define CS4245_PGA_B_CTRL 0x07 +#define CS4245_PGA_A_CTRL 0x08 +#define CS4245_ANALOG_IN 0x09 +#define CS4245_DAC_A_CTRL 0x0a +#define CS4245_DAC_B_CTRL 0x0b +#define CS4245_DAC_CTRL_2 0x0c +#define CS4245_INT_STATUS 0x0d +#define CS4245_INT_MASK 0x0e +#define CS4245_INT_MODE_MSB 0x0f +#define CS4245_INT_MODE_LSB 0x10 + +/* Chip ID */ +#define CS4245_CHIP_PART_MASK 0xf0 +#define CS4245_CHIP_REV_MASK 0x0f + +/* Power Control */ +#define CS4245_FREEZE 0x80 +#define CS4245_PDN_MIC 0x08 +#define CS4245_PDN_ADC 0x04 +#define CS4245_PDN_DAC 0x02 +#define CS4245_PDN 0x01 + +/* DAC Control */ +#define CS4245_DAC_FM_MASK 0xc0 +#define CS4245_DAC_FM_SINGLE 0x00 +#define CS4245_DAC_FM_DOUBLE 0x40 +#define CS4245_DAC_FM_QUAD 0x80 +#define CS4245_DAC_DIF_MASK 0x30 +#define CS4245_DAC_DIF_LJUST 0x00 +#define CS4245_DAC_DIF_I2S 0x10 +#define CS4245_DAC_DIF_RJUST_16 0x20 +#define CS4245_DAC_DIF_RJUST_24 0x30 +#define CS4245_RESERVED_1 0x08 +#define CS4245_MUTE_DAC 0x04 +#define CS4245_DEEMPH 0x02 +#define CS4245_DAC_MASTER 0x01 + +/* ADC Control */ +#define CS4245_ADC_FM_MASK 0xc0 +#define CS4245_ADC_FM_SINGLE 0x00 +#define CS4245_ADC_FM_DOUBLE 0x40 +#define CS4245_ADC_FM_QUAD 0x80 +#define CS4245_ADC_DIF_MASK 0x10 +#define CS4245_ADC_DIF_LJUST 0x00 +#define CS4245_ADC_DIF_I2S 0x10 +#define CS4245_MUTE_ADC 0x04 +#define CS4245_HPF_FREEZE 0x02 +#define CS4245_ADC_MASTER 0x01 + +/* MCLK Frequency */ +#define CS4245_MCLK1_MASK 0x70 +#define CS4245_MCLK1_SHIFT 4 +#define CS4245_MCLK2_MASK 0x07 +#define CS4245_MCLK2_SHIFT 0 +#define CS4245_MCLK_1 0 +#define CS4245_MCLK_1_5 1 +#define CS4245_MCLK_2 2 +#define CS4245_MCLK_3 3 +#define CS4245_MCLK_4 4 + +/* Signal Selection */ +#define CS4245_A_OUT_SEL_MASK 0x60 +#define CS4245_A_OUT_SEL_HIZ 0x00 +#define CS4245_A_OUT_SEL_DAC 0x20 +#define CS4245_A_OUT_SEL_PGA 0x40 +#define CS4245_LOOP 0x02 +#define CS4245_ASYNCH 0x01 + +/* Channel B/A PGA Control */ +#define CS4245_PGA_GAIN_MASK 0x3f + +/* ADC Input Control */ +#define CS4245_PGA_SOFT 0x10 +#define CS4245_PGA_ZERO 0x08 +#define CS4245_SEL_MASK 0x07 +#define CS4245_SEL_MIC 0x00 +#define CS4245_SEL_INPUT_1 0x01 +#define CS4245_SEL_INPUT_2 0x02 +#define CS4245_SEL_INPUT_3 0x03 +#define CS4245_SEL_INPUT_4 0x04 +#define CS4245_SEL_INPUT_5 0x05 +#define CS4245_SEL_INPUT_6 0x06 + +/* DAC Channel A/B Volume Control */ +#define CS4245_VOL_MASK 0xff + +/* DAC Control 2 */ +#define CS4245_DAC_SOFT 0x80 +#define CS4245_DAC_ZERO 0x40 +#define CS4245_INVERT_DAC 0x20 +#define CS4245_INT_ACTIVE_HIGH 0x01 + +/* Interrupt Status/Mask/Mode */ +#define CS4245_ADC_CLK_ERR 0x08 +#define CS4245_DAC_CLK_ERR 0x04 +#define CS4245_ADC_OVFL 0x02 +#define CS4245_ADC_UNDRFL 0x01 + + +#define CS4245_SPI_ADDRESS (0x9e << 16) +#define CS4245_SPI_WRITE (0 << 16) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c deleted file mode 100644 index 5a87d683691f..000000000000 --- a/sound/pci/oxygen/hifier.c +++ /dev/null @@ -1,239 +0,0 @@ -/* - * C-Media CMI8788 driver for the MediaTek/TempoTec HiFier Fantasia - * - * Copyright (c) Clemens Ladisch <clemens@ladisch.de> - * - * - * This driver is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License, version 2. - * - * 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 driver; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -/* - * CMI8788: - * - * SPI 0 -> AK4396 - */ - -#include <linux/delay.h> -#include <linux/pci.h> -#include <sound/control.h> -#include <sound/core.h> -#include <sound/initval.h> -#include <sound/pcm.h> -#include <sound/tlv.h> -#include "oxygen.h" -#include "ak4396.h" - -MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); -MODULE_DESCRIPTION("TempoTec HiFier driver"); -MODULE_LICENSE("GPL v2"); - -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, "card index"); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string"); -module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "enable card"); - -static DEFINE_PCI_DEVICE_TABLE(hifier_ids) = { - { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, - { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, - { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, - { } -}; -MODULE_DEVICE_TABLE(pci, hifier_ids); - -struct hifier_data { - u8 ak4396_regs[5]; -}; - -static void ak4396_write(struct oxygen *chip, u8 reg, u8 value) -{ - struct hifier_data *data = chip->model_data; - - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | - OXYGEN_SPI_DATA_LENGTH_2 | - OXYGEN_SPI_CLOCK_160 | - (0 << OXYGEN_SPI_CODEC_SHIFT) | - OXYGEN_SPI_CEN_LATCH_CLOCK_HI, - AK4396_WRITE | (reg << 8) | value); - data->ak4396_regs[reg] = value; -} - -static void ak4396_write_cached(struct oxygen *chip, u8 reg, u8 value) -{ - struct hifier_data *data = chip->model_data; - - if (value != data->ak4396_regs[reg]) - ak4396_write(chip, reg, value); -} - -static void hifier_registers_init(struct oxygen *chip) -{ - struct hifier_data *data = chip->model_data; - - ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); - ak4396_write(chip, AK4396_CONTROL_2, - data->ak4396_regs[AK4396_CONTROL_2]); - ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); - ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); -} - -static void hifier_init(struct oxygen *chip) -{ - struct hifier_data *data = chip->model_data; - - data->ak4396_regs[AK4396_CONTROL_2] = - AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; - hifier_registers_init(chip); - - snd_component_add(chip->card, "AK4396"); - snd_component_add(chip->card, "CS5340"); -} - -static void hifier_cleanup(struct oxygen *chip) -{ -} - -static void hifier_resume(struct oxygen *chip) -{ - hifier_registers_init(chip); -} - -static void set_ak4396_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - struct hifier_data *data = chip->model_data; - u8 value; - - value = data->ak4396_regs[AK4396_CONTROL_2] & ~AK4396_DFS_MASK; - if (params_rate(params) <= 54000) - value |= AK4396_DFS_NORMAL; - else if (params_rate(params) <= 108000) - value |= AK4396_DFS_DOUBLE; - else - value |= AK4396_DFS_QUAD; - - msleep(1); /* wait for the new MCLK to become stable */ - - if (value != data->ak4396_regs[AK4396_CONTROL_2]) { - ak4396_write(chip, AK4396_CONTROL_1, - AK4396_DIF_24_MSB); - ak4396_write(chip, AK4396_CONTROL_2, value); - ak4396_write(chip, AK4396_CONTROL_1, - AK4396_DIF_24_MSB | AK4396_RSTN); - } -} - -static void update_ak4396_volume(struct oxygen *chip) -{ - ak4396_write_cached(chip, AK4396_LCH_ATT, chip->dac_volume[0]); - ak4396_write_cached(chip, AK4396_RCH_ATT, chip->dac_volume[1]); -} - -static void update_ak4396_mute(struct oxygen *chip) -{ - struct hifier_data *data = chip->model_data; - u8 value; - - value = data->ak4396_regs[AK4396_CONTROL_2] & ~AK4396_SMUTE; - if (chip->dac_mute) - value |= AK4396_SMUTE; - ak4396_write_cached(chip, AK4396_CONTROL_2, value); -} - -static void set_cs5340_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ -} - -static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); - -static const struct oxygen_model model_hifier = { - .shortname = "C-Media CMI8787", - .longname = "C-Media Oxygen HD Audio", - .chip = "CMI8788", - .init = hifier_init, - .cleanup = hifier_cleanup, - .resume = hifier_resume, - .get_i2s_mclk = oxygen_default_i2s_mclk, - .set_dac_params = set_ak4396_params, - .set_adc_params = set_cs5340_params, - .update_dac_volume = update_ak4396_volume, - .update_dac_mute = update_ak4396_mute, - .dac_tlv = ak4396_db_scale, - .model_data_size = sizeof(struct hifier_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_1, - .dac_channels = 2, - .dac_volume_min = 0, - .dac_volume_max = 255, - .function_flags = OXYGEN_FUNCTION_SPI, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - -static int __devinit get_hifier_model(struct oxygen *chip, - const struct pci_device_id *id) -{ - chip->model = model_hifier; - return 0; -} - -static int __devinit hifier_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) -{ - static int dev; - int err; - - if (dev >= SNDRV_CARDS) - return -ENODEV; - if (!enable[dev]) { - ++dev; - return -ENOENT; - } - err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, - hifier_ids, get_hifier_model); - if (err >= 0) - ++dev; - return err; -} - -static struct pci_driver hifier_driver = { - .name = "CMI8787HiFier", - .id_table = hifier_ids, - .probe = hifier_probe, - .remove = __devexit_p(oxygen_pci_remove), -#ifdef CONFIG_PM - .suspend = oxygen_pci_suspend, - .resume = oxygen_pci_resume, -#endif -}; - -static int __init alsa_card_hifier_init(void) -{ - return pci_register_driver(&hifier_driver); -} - -static void __exit alsa_card_hifier_exit(void) -{ - pci_unregister_driver(&hifier_driver); -} - -module_init(alsa_card_hifier_init) -module_exit(alsa_card_hifier_exit) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 98a8eb3c92f7..d7e8ddd9a67b 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -20,19 +20,32 @@ /* * CMI8788: * - * SPI 0 -> 1st AK4396 (front) - * SPI 1 -> 2nd AK4396 (surround) - * SPI 2 -> 3rd AK4396 (center/LFE) - * SPI 3 -> WM8785 - * SPI 4 -> 4th AK4396 (back) + * SPI 0 -> 1st AK4396 (front) + * SPI 1 -> 2nd AK4396 (surround) + * SPI 2 -> 3rd AK4396 (center/LFE) + * SPI 3 -> WM8785 + * SPI 4 -> 4th AK4396 (back) * - * GPIO 0 -> DFS0 of AK5385 - * GPIO 1 -> DFS1 of AK5385 - * GPIO 8 -> enable headphone amplifier on HT-Omega models + * GPIO 0 -> DFS0 of AK5385 + * GPIO 1 -> DFS1 of AK5385 + * + * X-Meridian models: + * GPIO 4 -> enable extension S/PDIF input + * GPIO 6 -> enable on-board S/PDIF input + * + * Claro models: + * GPIO 6 -> S/PDIF from optical (0) or coaxial (1) input + * GPIO 8 -> enable headphone amplifier * * CM9780: * - * GPO 0 -> route line-in (0) or AC97 output (1) to ADC input + * LINE_OUT -> input of ADC + * + * AUX_IN <- aux + * CD_IN <- CD + * MIC_IN <- mic + * + * GPO 0 -> route line-in (0) or AC97 output (1) to ADC input */ #include <linux/delay.h> @@ -41,18 +54,22 @@ #include <sound/ac97_codec.h> #include <sound/control.h> #include <sound/core.h> +#include <sound/info.h> #include <sound/initval.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/tlv.h> #include "oxygen.h" +#include "xonar_dg.h" #include "ak4396.h" #include "wm8785.h" MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_DESCRIPTION("C-Media CMI8788 driver"); MODULE_LICENSE("GPL v2"); -MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8788}}"); +MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8786}" + ",{C-Media,CMI8787}" + ",{C-Media,CMI8788}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; @@ -66,24 +83,46 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); enum { - MODEL_CMEDIA_REF, /* C-Media's reference design */ - MODEL_MERIDIAN, /* AuzenTech X-Meridian */ - MODEL_CLARO, /* HT-Omega Claro */ - MODEL_CLARO_HALO, /* HT-Omega Claro halo */ + MODEL_CMEDIA_REF, + MODEL_MERIDIAN, + MODEL_MERIDIAN_2G, + MODEL_CLARO, + MODEL_CLARO_HALO, + MODEL_FANTASIA, + MODEL_SERENADE, + MODEL_2CH_OUTPUT, + MODEL_HG2PCI, + MODEL_XONAR_DG, }; static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { + /* C-Media's reference design */ { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF }, + { OXYGEN_PCI_SUBID(0x10b0, 0x0217), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0218), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0219), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x0001), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x0010), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF }, - { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, + /* Asus Xonar DG */ + { OXYGEN_PCI_SUBID(0x1043, 0x8467), .driver_data = MODEL_XONAR_DG }, + /* PCI 2.0 HD Audio */ + { OXYGEN_PCI_SUBID(0x13f6, 0x8782), .driver_data = MODEL_2CH_OUTPUT }, + /* Kuroutoshikou CMI8787-HG2PCI */ + { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_HG2PCI }, + /* TempoTec HiFier Fantasia */ + { OXYGEN_PCI_SUBID(0x14c3, 0x1710), .driver_data = MODEL_FANTASIA }, + /* TempoTec HiFier Serenade */ + { OXYGEN_PCI_SUBID(0x14c3, 0x1711), .driver_data = MODEL_SERENADE }, + /* AuzenTech X-Meridian */ { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, + /* AuzenTech X-Meridian 2G */ + { OXYGEN_PCI_SUBID(0x5431, 0x017a), .driver_data = MODEL_MERIDIAN_2G }, + /* HT-Omega Claro */ { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CLARO }, + /* HT-Omega Claro halo */ { OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_CLARO_HALO }, { } }; @@ -95,9 +134,15 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_AK5385_DFS_DOUBLE 0x0001 #define GPIO_AK5385_DFS_QUAD 0x0002 +#define GPIO_MERIDIAN_DIG_MASK 0x0050 +#define GPIO_MERIDIAN_DIG_EXT 0x0010 +#define GPIO_MERIDIAN_DIG_BOARD 0x0040 + +#define GPIO_CLARO_DIG_COAX 0x0040 #define GPIO_CLARO_HP 0x0100 struct generic_data { + unsigned int dacs; u8 ak4396_regs[4][5]; u16 wm8785_regs[3]; }; @@ -148,7 +193,7 @@ static void ak4396_registers_init(struct oxygen *chip) struct generic_data *data = chip->model_data; unsigned int i; - for (i = 0; i < 4; ++i) { + for (i = 0; i < data->dacs; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); ak4396_write(chip, i, AK4396_CONTROL_2, @@ -166,6 +211,7 @@ static void ak4396_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; + data->dacs = chip->model.dac_channels_pcm / 2; data->ak4396_regs[0][AK4396_CONTROL_2] = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; ak4396_registers_init(chip); @@ -207,6 +253,10 @@ static void generic_init(struct oxygen *chip) static void meridian_init(struct oxygen *chip) { + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_MERIDIAN_DIG_MASK); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + GPIO_MERIDIAN_DIG_BOARD, GPIO_MERIDIAN_DIG_MASK); ak4396_init(chip); ak5385_init(chip); } @@ -220,6 +270,8 @@ static void claro_enable_hp(struct oxygen *chip) static void claro_init(struct oxygen *chip) { + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CLARO_DIG_COAX); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_DIG_COAX); ak4396_init(chip); wm8785_init(chip); claro_enable_hp(chip); @@ -227,11 +279,24 @@ static void claro_init(struct oxygen *chip) static void claro_halo_init(struct oxygen *chip) { + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CLARO_DIG_COAX); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_DIG_COAX); ak4396_init(chip); ak5385_init(chip); claro_enable_hp(chip); } +static void fantasia_init(struct oxygen *chip) +{ + ak4396_init(chip); + snd_component_add(chip->card, "CS5340"); +} + +static void stereo_output_init(struct oxygen *chip) +{ + ak4396_init(chip); +} + static void generic_cleanup(struct oxygen *chip) { } @@ -268,6 +333,11 @@ static void claro_resume(struct oxygen *chip) claro_enable_hp(chip); } +static void stereo_resume(struct oxygen *chip) +{ + ak4396_registers_init(chip); +} + static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -286,7 +356,7 @@ static void set_ak4396_params(struct oxygen *chip, msleep(1); /* wait for the new MCLK to become stable */ if (value != data->ak4396_regs[0][AK4396_CONTROL_2]) { - for (i = 0; i < 4; ++i) { + for (i = 0; i < data->dacs; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB); ak4396_write(chip, i, AK4396_CONTROL_2, value); @@ -298,9 +368,10 @@ static void set_ak4396_params(struct oxygen *chip, static void update_ak4396_volume(struct oxygen *chip) { + struct generic_data *data = chip->model_data; unsigned int i; - for (i = 0; i < 4; ++i) { + for (i = 0; i < data->dacs; ++i) { ak4396_write_cached(chip, i, AK4396_LCH_ATT, chip->dac_volume[i * 2]); ak4396_write_cached(chip, i, AK4396_RCH_ATT, @@ -317,7 +388,7 @@ static void update_ak4396_mute(struct oxygen *chip) value = data->ak4396_regs[0][AK4396_CONTROL_2] & ~AK4396_SMUTE; if (chip->dac_mute) value |= AK4396_SMUTE; - for (i = 0; i < 4; ++i) + for (i = 0; i < data->dacs; ++i) ak4396_write_cached(chip, i, AK4396_CONTROL_2, value); } @@ -356,6 +427,10 @@ static void set_ak5385_params(struct oxygen *chip, value, GPIO_AK5385_DFS_MASK); } +static void set_no_params(struct oxygen *chip, struct snd_pcm_hw_params *params) +{ +} + static int rolloff_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -363,13 +438,7 @@ static int rolloff_info(struct snd_kcontrol *ctl, "Sharp Roll-off", "Slow Roll-off" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int rolloff_get(struct snd_kcontrol *ctl, @@ -400,7 +469,7 @@ static int rolloff_put(struct snd_kcontrol *ctl, reg &= ~AK4396_SLOW; changed = reg != data->ak4396_regs[0][AK4396_CONTROL_2]; if (changed) { - for (i = 0; i < 4; ++i) + for (i = 0; i < data->dacs; ++i) ak4396_write(chip, i, AK4396_CONTROL_2, reg); } mutex_unlock(&chip->mutex); @@ -421,13 +490,7 @@ static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) "None", "High-pass Filter" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) @@ -466,6 +529,100 @@ static const struct snd_kcontrol_new hpf_control = { .put = hpf_put, }; +static int meridian_dig_source_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { "On-board", "Extension" }; + + return snd_ctl_enum_info(info, 1, 2, names); +} + +static int claro_dig_source_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { "Optical", "Coaxial" }; + + return snd_ctl_enum_info(info, 1, 2, names); +} + +static int meridian_dig_source_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + value->value.enumerated.item[0] = + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & + GPIO_MERIDIAN_DIG_EXT); + return 0; +} + +static int claro_dig_source_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + value->value.enumerated.item[0] = + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & + GPIO_CLARO_DIG_COAX); + return 0; +} + +static int meridian_dig_source_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 old_reg, new_reg; + int changed; + + mutex_lock(&chip->mutex); + old_reg = oxygen_read16(chip, OXYGEN_GPIO_DATA); + new_reg = old_reg & ~GPIO_MERIDIAN_DIG_MASK; + if (value->value.enumerated.item[0] == 0) + new_reg |= GPIO_MERIDIAN_DIG_BOARD; + else + new_reg |= GPIO_MERIDIAN_DIG_EXT; + changed = new_reg != old_reg; + if (changed) + oxygen_write16(chip, OXYGEN_GPIO_DATA, new_reg); + mutex_unlock(&chip->mutex); + return changed; +} + +static int claro_dig_source_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 old_reg, new_reg; + int changed; + + mutex_lock(&chip->mutex); + old_reg = oxygen_read16(chip, OXYGEN_GPIO_DATA); + new_reg = old_reg & ~GPIO_CLARO_DIG_COAX; + if (value->value.enumerated.item[0]) + new_reg |= GPIO_CLARO_DIG_COAX; + changed = new_reg != old_reg; + if (changed) + oxygen_write16(chip, OXYGEN_GPIO_DATA, new_reg); + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new meridian_dig_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Source Capture Enum", + .info = meridian_dig_source_info, + .get = meridian_dig_source_get, + .put = meridian_dig_source_put, +}; + +static const struct snd_kcontrol_new claro_dig_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Source Capture Enum", + .info = claro_dig_source_info, + .get = claro_dig_source_get, + .put = claro_dig_source_put, +}; + static int generic_mixer_init(struct oxygen *chip) { return snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); @@ -484,6 +641,81 @@ static int generic_wm8785_mixer_init(struct oxygen *chip) return 0; } +static int meridian_mixer_init(struct oxygen *chip) +{ + int err; + + err = generic_mixer_init(chip); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&meridian_dig_source_control, chip)); + if (err < 0) + return err; + return 0; +} + +static int claro_mixer_init(struct oxygen *chip) +{ + int err; + + err = generic_wm8785_mixer_init(chip); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&claro_dig_source_control, chip)); + if (err < 0) + return err; + return 0; +} + +static int claro_halo_mixer_init(struct oxygen *chip) +{ + int err; + + err = generic_mixer_init(chip); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&claro_dig_source_control, chip)); + if (err < 0) + return err; + return 0; +} + +static void dump_ak4396_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct generic_data *data = chip->model_data; + unsigned int dac, i; + + for (dac = 0; dac < data->dacs; ++dac) { + snd_iprintf(buffer, "\nAK4396 %u:", dac + 1); + for (i = 0; i < 5; ++i) + snd_iprintf(buffer, " %02x", data->ak4396_regs[dac][i]); + } + snd_iprintf(buffer, "\n"); +} + +static void dump_wm8785_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct generic_data *data = chip->model_data; + unsigned int i; + + snd_iprintf(buffer, "\nWM8785:"); + for (i = 0; i < 3; ++i) + snd_iprintf(buffer, " %03x", data->wm8785_regs[i]); + snd_iprintf(buffer, "\n"); +} + +static void dump_oxygen_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + dump_ak4396_registers(chip, buffer); + dump_wm8785_registers(chip, buffer); +} + static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static const struct oxygen_model model_generic = { @@ -494,11 +726,11 @@ static const struct oxygen_model model_generic = { .mixer_init = generic_wm8785_mixer_init, .cleanup = generic_cleanup, .resume = generic_resume, - .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, .update_dac_mute = update_ak4396_mute, + .dump_registers = dump_oxygen_registers, .dac_tlv = ak4396_db_scale, .model_data_size = sizeof(struct generic_data), .device_config = PLAYBACK_0_TO_I2S | @@ -508,11 +740,14 @@ static const struct oxygen_model model_generic = { CAPTURE_1_FROM_SPDIF | CAPTURE_2_FROM_AC97_1 | AC97_CD_INPUT, - .dac_channels = 8, + .dac_channels_pcm = 8, + .dac_channels_mixer = 8, .dac_volume_min = 0, .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_mclks = OXYGEN_MCLKS(256, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -520,42 +755,87 @@ static const struct oxygen_model model_generic = { static int __devinit get_oxygen_model(struct oxygen *chip, const struct pci_device_id *id) { + static const char *const names[] = { + [MODEL_MERIDIAN] = "AuzenTech X-Meridian", + [MODEL_MERIDIAN_2G] = "AuzenTech X-Meridian 2G", + [MODEL_CLARO] = "HT-Omega Claro", + [MODEL_CLARO_HALO] = "HT-Omega Claro halo", + [MODEL_FANTASIA] = "TempoTec HiFier Fantasia", + [MODEL_SERENADE] = "TempoTec HiFier Serenade", + [MODEL_HG2PCI] = "CMI8787-HG2PCI", + }; + chip->model = model_generic; switch (id->driver_data) { case MODEL_MERIDIAN: + case MODEL_MERIDIAN_2G: chip->model.init = meridian_init; - chip->model.mixer_init = generic_mixer_init; + chip->model.mixer_init = meridian_mixer_init; chip->model.resume = meridian_resume; chip->model.set_adc_params = set_ak5385_params; + chip->model.dump_registers = dump_ak4396_registers; chip->model.device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF; + if (id->driver_data == MODEL_MERIDIAN) + chip->model.device_config |= AC97_CD_INPUT; break; case MODEL_CLARO: chip->model.init = claro_init; + chip->model.mixer_init = claro_mixer_init; chip->model.cleanup = claro_cleanup; chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; break; case MODEL_CLARO_HALO: chip->model.init = claro_halo_init; - chip->model.mixer_init = generic_mixer_init; + chip->model.mixer_init = claro_halo_mixer_init; chip->model.cleanup = claro_cleanup; chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; chip->model.set_adc_params = set_ak5385_params; + chip->model.dump_registers = dump_ak4396_registers; chip->model.device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF; break; + case MODEL_FANTASIA: + case MODEL_SERENADE: + case MODEL_2CH_OUTPUT: + case MODEL_HG2PCI: + chip->model.shortname = "C-Media CMI8787"; + chip->model.chip = "CMI8787"; + if (id->driver_data == MODEL_FANTASIA) + chip->model.init = fantasia_init; + else + chip->model.init = stereo_output_init; + chip->model.resume = stereo_resume; + chip->model.mixer_init = generic_mixer_init; + chip->model.set_adc_params = set_no_params; + chip->model.dump_registers = dump_ak4396_registers; + chip->model.device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF; + if (id->driver_data == MODEL_FANTASIA) { + chip->model.device_config |= CAPTURE_0_FROM_I2S_1; + chip->model.adc_mclks = OXYGEN_MCLKS(256, 128, 128); + } + chip->model.dac_channels_pcm = 2; + chip->model.dac_channels_mixer = 2; + break; + case MODEL_XONAR_DG: + chip->model = model_xonar_dg; + break; } if (id->driver_data == MODEL_MERIDIAN || + id->driver_data == MODEL_MERIDIAN_2G || id->driver_data == MODEL_CLARO_HALO) { chip->model.misc_flags = OXYGEN_MISC_MIDI; chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; } + if (id->driver_data < ARRAY_SIZE(names) && names[id->driver_data]) + chip->model.shortname = names[id->driver_data]; return 0; } diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 7d5222caa0a9..c2ae63d17cd2 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -16,6 +16,10 @@ #define PCM_AC97 5 #define PCM_COUNT 6 +#define OXYGEN_MCLKS(f_single, f_double, f_quad) ((MCLK_##f_single << 0) | \ + (MCLK_##f_double << 2) | \ + (MCLK_##f_quad << 4)) + #define OXYGEN_IO_SIZE 0x100 #define OXYGEN_EEPROM_ID 0x434d /* "CM" */ @@ -35,6 +39,7 @@ #define MIDI_OUTPUT 0x0800 #define MIDI_INPUT 0x1000 #define AC97_CD_INPUT 0x2000 +#define AC97_FMIC_SWITCH 0x4000 enum { CONTROL_SPDIF_PCM, @@ -65,6 +70,7 @@ struct snd_pcm_hardware; struct snd_pcm_hw_params; struct snd_kcontrol_new; struct snd_rawmidi; +struct snd_info_buffer; struct oxygen; struct oxygen_model { @@ -79,8 +85,6 @@ struct oxygen_model { void (*resume)(struct oxygen *chip); void (*pcm_hardware_filter)(unsigned int channel, struct snd_pcm_hardware *hardware); - unsigned int (*get_i2s_mclk)(struct oxygen *chip, unsigned int channel, - struct snd_pcm_hw_params *hw_params); void (*set_dac_params)(struct oxygen *chip, struct snd_pcm_hw_params *params); void (*set_adc_params)(struct oxygen *chip, @@ -92,15 +96,19 @@ struct oxygen_model { void (*uart_input)(struct oxygen *chip); void (*ac97_switch)(struct oxygen *chip, unsigned int reg, unsigned int mute); + void (*dump_registers)(struct oxygen *chip, + struct snd_info_buffer *buffer); const unsigned int *dac_tlv; - unsigned long private_data; size_t model_data_size; unsigned int device_config; - u8 dac_channels; + u8 dac_channels_pcm; + u8 dac_channels_mixer; u8 dac_volume_min; u8 dac_volume_max; u8 misc_flags; u8 function_flags; + u8 dac_mclks; + u8 adc_mclks; u16 dac_i2s_format; u16 adc_i2s_format; }; @@ -121,7 +129,6 @@ struct oxygen { u8 pcm_running; u8 dac_routing; u8 spdif_playback_enable; - u8 revision; u8 has_ac97_0; u8 has_ac97_1; u32 spdif_bits; @@ -167,8 +174,6 @@ void oxygen_update_spdif_source(struct oxygen *chip); /* oxygen_pcm.c */ int oxygen_pcm_init(struct oxygen *chip); -unsigned int oxygen_default_i2s_mclk(struct oxygen *chip, unsigned int channel, - struct snd_pcm_hw_params *hw_params); /* oxygen_io.c */ diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 09b2b2a36df5..f5164b1e1c80 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -197,11 +197,11 @@ void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; - /* should not need more than 7.68 us (24 * 320 ns) */ + /* should not need more than 30.72 us (24 * 1.28 us) */ count = 10; while ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & OXYGEN_SPI_BUSY) && count > 0) { - udelay(1); + udelay(4); --count; } diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 969605fbcb7f..70b739816fcc 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -202,7 +202,13 @@ static void oxygen_proc_read(struct snd_info_entry *entry, struct oxygen *chip = entry->private_data; int i, j; - snd_iprintf(buffer, "CMI8788\n\n"); + switch (oxygen_read8(chip, OXYGEN_REVISION) & OXYGEN_PACKAGE_ID_MASK) { + case OXYGEN_PACKAGE_ID_8786: i = '6'; break; + case OXYGEN_PACKAGE_ID_8787: i = '7'; break; + case OXYGEN_PACKAGE_ID_8788: i = '8'; break; + default: i = '?'; break; + } + snd_iprintf(buffer, "CMI878%c:\n", i); for (i = 0; i < OXYGEN_IO_SIZE; i += 0x10) { snd_iprintf(buffer, "%02x:", i); for (j = 0; j < 0x10; ++j) @@ -212,7 +218,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry, if (mutex_lock_interruptible(&chip->mutex) < 0) return; if (chip->has_ac97_0) { - snd_iprintf(buffer, "\nAC97\n"); + snd_iprintf(buffer, "\nAC97:\n"); for (i = 0; i < 0x80; i += 0x10) { snd_iprintf(buffer, "%02x:", i); for (j = 0; j < 0x10; j += 2) @@ -222,7 +228,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry, } } if (chip->has_ac97_1) { - snd_iprintf(buffer, "\nAC97 2\n"); + snd_iprintf(buffer, "\nAC97 2:\n"); for (i = 0; i < 0x80; i += 0x10) { snd_iprintf(buffer, "%02x:", i); for (j = 0; j < 0x10; j += 2) @@ -232,13 +238,15 @@ static void oxygen_proc_read(struct snd_info_entry *entry, } } mutex_unlock(&chip->mutex); + if (chip->model.dump_registers) + chip->model.dump_registers(chip, buffer); } static void oxygen_proc_init(struct oxygen *chip) { struct snd_info_entry *entry; - if (!snd_card_proc_new(chip->card, "cmi8788", &entry)) + if (!snd_card_proc_new(chip->card, "oxygen", &entry)) snd_info_set_text_ops(entry, chip, oxygen_proc_read); } #else @@ -262,7 +270,7 @@ oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[]) */ subdevice = oxygen_read_eeprom(chip, 2); /* use default ID if EEPROM is missing */ - if (subdevice == 0xffff) + if (subdevice == 0xffff && oxygen_read_eeprom(chip, 1) == 0xffff) subdevice = 0x8788; /* * We use only the subsystem device ID for searching because it is @@ -364,12 +372,7 @@ static void oxygen_init(struct oxygen *chip) (IEC958_AES1_CON_PCM_CODER << OXYGEN_SPDIF_CATEGORY_SHIFT); chip->spdif_pcm_bits = chip->spdif_bits; - if (oxygen_read8(chip, OXYGEN_REVISION) & OXYGEN_REVISION_2) - chip->revision = 2; - else - chip->revision = 1; - - if (chip->revision == 1) + if (!(oxygen_read8(chip, OXYGEN_REVISION) & OXYGEN_REVISION_2)) oxygen_set_bits8(chip, OXYGEN_MISC, OXYGEN_MISC_PCI_MEM_W_1_CLOCK); @@ -406,28 +409,40 @@ static void oxygen_init(struct oxygen *chip) (OXYGEN_FORMAT_16 << OXYGEN_MULTICH_FORMAT_SHIFT)); oxygen_write8(chip, OXYGEN_REC_CHANNELS, OXYGEN_REC_CHANNELS_2_2_2); oxygen_write16(chip, OXYGEN_I2S_MULTICH_FORMAT, - OXYGEN_RATE_48000 | chip->model.dac_i2s_format | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + OXYGEN_RATE_48000 | + chip->model.dac_i2s_format | + OXYGEN_I2S_MCLK(chip->model.dac_mclks) | + OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | + OXYGEN_I2S_BCLK_64); if (chip->model.device_config & CAPTURE_0_FROM_I2S_1) oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, - OXYGEN_RATE_48000 | chip->model.adc_i2s_format | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + OXYGEN_RATE_48000 | + chip->model.adc_i2s_format | + OXYGEN_I2S_MCLK(chip->model.adc_mclks) | + OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | + OXYGEN_I2S_BCLK_64); else oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, - OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); + OXYGEN_I2S_MASTER | + OXYGEN_I2S_MUTE_MCLK); if (chip->model.device_config & (CAPTURE_0_FROM_I2S_2 | CAPTURE_2_FROM_I2S_2)) oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, - OXYGEN_RATE_48000 | chip->model.adc_i2s_format | - OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + OXYGEN_RATE_48000 | + chip->model.adc_i2s_format | + OXYGEN_I2S_MCLK(chip->model.adc_mclks) | + OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | + OXYGEN_I2S_BCLK_64); else oxygen_write16(chip, OXYGEN_I2S_B_FORMAT, - OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); + OXYGEN_I2S_MASTER | + OXYGEN_I2S_MUTE_MCLK); oxygen_write16(chip, OXYGEN_I2S_C_FORMAT, - OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK); + OXYGEN_I2S_MASTER | + OXYGEN_I2S_MUTE_MCLK); oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_OUT_ENABLE | OXYGEN_SPDIF_LOOPBACK); @@ -649,8 +664,8 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, strcpy(card->driver, chip->model.chip); strcpy(card->shortname, chip->model.shortname); - sprintf(card->longname, "%s (rev %u) at %#lx, irq %i", - chip->model.longname, chip->revision, chip->addr, chip->irq); + sprintf(card->longname, "%s at %#lx, irq %i", + chip->model.longname, chip->addr, chip->irq); strcpy(card->mixername, chip->model.chip); snd_component_add(card, chip->model.chip); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 2849b36f5f7e..9bff14d5895d 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -31,7 +31,7 @@ static int dac_volume_info(struct snd_kcontrol *ctl, struct oxygen *chip = ctl->private_data; info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = chip->model.dac_channels; + info->count = chip->model.dac_channels_mixer; info->value.integer.min = chip->model.dac_volume_min; info->value.integer.max = chip->model.dac_volume_max; return 0; @@ -44,7 +44,7 @@ static int dac_volume_get(struct snd_kcontrol *ctl, unsigned int i; mutex_lock(&chip->mutex); - for (i = 0; i < chip->model.dac_channels; ++i) + for (i = 0; i < chip->model.dac_channels_mixer; ++i) value->value.integer.value[i] = chip->dac_volume[i]; mutex_unlock(&chip->mutex); return 0; @@ -59,7 +59,7 @@ static int dac_volume_put(struct snd_kcontrol *ctl, changed = 0; mutex_lock(&chip->mutex); - for (i = 0; i < chip->model.dac_channels; ++i) + for (i = 0; i < chip->model.dac_channels_mixer; ++i) if (value->value.integer.value[i] != chip->dac_volume[i]) { chip->dac_volume[i] = value->value.integer.value[i]; changed = 1; @@ -97,6 +97,16 @@ static int dac_mute_put(struct snd_kcontrol *ctl, return changed; } +static unsigned int upmix_item_count(struct oxygen *chip) +{ + if (chip->model.dac_channels_pcm < 8) + return 2; + else if (chip->model.update_center_lfe_mix) + return 5; + else + return 3; +} + static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { static const char *const names[5] = { @@ -107,15 +117,9 @@ static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) "Front+Surround+Center/LFE+Back", }; struct oxygen *chip = ctl->private_data; - unsigned int count = chip->model.update_center_lfe_mix ? 5 : 3; + unsigned int count = upmix_item_count(chip); - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = count; - if (info->value.enumerated.item >= count) - info->value.enumerated.item = count - 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, count, names); } static int upmix_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) @@ -188,7 +192,7 @@ void oxygen_update_dac_routing(struct oxygen *chip) static int upmix_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; - unsigned int count = chip->model.update_center_lfe_mix ? 5 : 3; + unsigned int count = upmix_item_count(chip); int changed; if (value->value.enumerated.item[0] >= count) @@ -430,30 +434,31 @@ static int spdif_input_default_get(struct snd_kcontrol *ctl, return 0; } -static int spdif_loopback_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +static int spdif_bit_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; + u32 bit = ctl->private_value; value->value.integer.value[0] = - !!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) - & OXYGEN_SPDIF_LOOPBACK); + !!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL) & bit); return 0; } -static int spdif_loopback_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +static int spdif_bit_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; + u32 bit = ctl->private_value; u32 oldreg, newreg; int changed; spin_lock_irq(&chip->reg_lock); oldreg = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL); if (value->value.integer.value[0]) - newreg = oldreg | OXYGEN_SPDIF_LOOPBACK; + newreg = oldreg | bit; else - newreg = oldreg & ~OXYGEN_SPDIF_LOOPBACK; + newreg = oldreg & ~bit; changed = newreg != oldreg; if (changed) oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, newreg); @@ -644,6 +649,46 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, return change; } +static int mic_fmic_source_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[] = { "Mic Jack", "Front Panel" }; + + return snd_ctl_enum_info(info, 1, 2, names); +} + +static int mic_fmic_source_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + + mutex_lock(&chip->mutex); + value->value.enumerated.item[0] = + !!(oxygen_read_ac97(chip, 0, CM9780_JACK) & CM9780_FMIC2MIC); + mutex_unlock(&chip->mutex); + return 0; +} + +static int mic_fmic_source_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 oldreg, newreg; + int change; + + mutex_lock(&chip->mutex); + oldreg = oxygen_read_ac97(chip, 0, CM9780_JACK); + if (value->value.enumerated.item[0]) + newreg = oldreg | CM9780_FMIC2MIC; + else + newreg = oldreg & ~CM9780_FMIC2MIC; + change = newreg != oldreg; + if (change) + oxygen_write_ac97(chip, 0, CM9780_JACK, newreg); + mutex_unlock(&chip->mutex); + return change; +} + static int ac97_fp_rec_volume_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -791,8 +836,17 @@ static const struct snd_kcontrol_new spdif_input_controls[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("Loopback ", NONE, SWITCH), .info = snd_ctl_boolean_mono_info, - .get = spdif_loopback_get, - .put = spdif_loopback_put, + .get = spdif_bit_switch_get, + .put = spdif_bit_switch_put, + .private_value = OXYGEN_SPDIF_LOOPBACK, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("Validity Check ",CAPTURE,SWITCH), + .info = snd_ctl_boolean_mono_info, + .get = spdif_bit_switch_get, + .put = spdif_bit_switch_put, + .private_value = OXYGEN_SPDIF_SPDVALID, }, }; @@ -908,6 +962,13 @@ static const struct snd_kcontrol_new ac97_controls[] = { AC97_VOLUME("Mic Capture Volume", 0, AC97_MIC, 0), AC97_SWITCH("Mic Capture Switch", 0, AC97_MIC, 15, 1), AC97_SWITCH("Mic Boost (+20dB)", 0, AC97_MIC, 6, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Source Capture Enum", + .info = mic_fmic_source_info, + .get = mic_fmic_source_get, + .put = mic_fmic_source_put, + }, AC97_SWITCH("Line Capture Switch", 0, AC97_LINE, 15, 1), AC97_VOLUME("CD Capture Volume", 0, AC97_CD, 1), AC97_SWITCH("CD Capture Switch", 0, AC97_CD, 15, 1), @@ -970,7 +1031,10 @@ static int add_controls(struct oxygen *chip, continue; } if (!strcmp(template.name, "Stereo Upmixing") && - chip->model.dac_channels == 2) + chip->model.dac_channels_pcm == 2) + continue; + if (!strcmp(template.name, "Mic Source Capture Enum") && + !(chip->model.device_config & AC97_FMIC_SWITCH)) continue; if (!strncmp(template.name, "CD Capture ", 11) && !(chip->model.device_config & AC97_CD_INPUT)) diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 814667442eb0..d5533e34ece9 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -39,7 +39,8 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_32000 | @@ -65,7 +66,8 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_32000 | @@ -91,7 +93,8 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_SYNC_START, + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, @@ -140,7 +143,7 @@ static int oxygen_open(struct snd_pcm_substream *substream, runtime->hw.rate_min = 44100; break; case PCM_MULTICH: - runtime->hw.channels_max = chip->model.dac_channels; + runtime->hw.channels_max = chip->model.dac_channels_pcm; break; } if (chip->model.pcm_hardware_filter) @@ -271,17 +274,6 @@ static unsigned int oxygen_rate(struct snd_pcm_hw_params *hw_params) } } -unsigned int oxygen_default_i2s_mclk(struct oxygen *chip, - unsigned int channel, - struct snd_pcm_hw_params *hw_params) -{ - if (params_rate(hw_params) <= 96000) - return OXYGEN_I2S_MCLK_256; - else - return OXYGEN_I2S_MCLK_128; -} -EXPORT_SYMBOL(oxygen_default_i2s_mclk); - static unsigned int oxygen_i2s_bits(struct snd_pcm_hw_params *hw_params) { if (params_format(hw_params) == SNDRV_PCM_FORMAT_S32_LE) @@ -341,6 +333,26 @@ static int oxygen_hw_params(struct snd_pcm_substream *substream, return 0; } +static u16 get_mclk(struct oxygen *chip, unsigned int channel, + struct snd_pcm_hw_params *params) +{ + unsigned int mclks, shift; + + if (channel == PCM_MULTICH) + mclks = chip->model.dac_mclks; + else + mclks = chip->model.adc_mclks; + + if (params_rate(params) <= 48000) + shift = 0; + else if (params_rate(params) <= 96000) + shift = 2; + else + shift = 4; + + return OXYGEN_I2S_MCLK(mclks >> shift); +} + static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -357,8 +369,8 @@ static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, OXYGEN_REC_FORMAT_A_MASK); oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, oxygen_rate(hw_params) | - chip->model.get_i2s_mclk(chip, PCM_A, hw_params) | chip->model.adc_i2s_format | + get_mclk(chip, PCM_A, hw_params) | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | @@ -393,9 +405,8 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, if (!is_ac97) oxygen_write16_masked(chip, OXYGEN_I2S_B_FORMAT, oxygen_rate(hw_params) | - chip->model.get_i2s_mclk(chip, PCM_B, - hw_params) | chip->model.adc_i2s_format | + get_mclk(chip, PCM_B, hw_params) | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | @@ -476,8 +487,7 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, oxygen_rate(hw_params) | chip->model.dac_i2s_format | - chip->model.get_i2s_mclk(chip, PCM_MULTICH, - hw_params) | + get_mclk(chip, PCM_MULTICH, hw_params) | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | @@ -530,7 +540,10 @@ static int oxygen_prepare(struct snd_pcm_substream *substream) oxygen_set_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); oxygen_clear_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); - chip->interrupt_mask |= channel_mask; + if (substream->runtime->no_period_wakeup) + chip->interrupt_mask &= ~channel_mask; + else + chip->interrupt_mask |= channel_mask; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); spin_unlock_irq(&chip->reg_lock); return 0; diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h index 4dcd41b78258..63dc7a0ab555 100644 --- a/sound/pci/oxygen/oxygen_regs.h +++ b/sound/pci/oxygen/oxygen_regs.h @@ -139,9 +139,11 @@ #define OXYGEN_I2S_FORMAT_I2S 0x0000 #define OXYGEN_I2S_FORMAT_LJUST 0x0008 #define OXYGEN_I2S_MCLK_MASK 0x0030 /* MCLK/LRCK */ -#define OXYGEN_I2S_MCLK_128 0x0000 -#define OXYGEN_I2S_MCLK_256 0x0010 -#define OXYGEN_I2S_MCLK_512 0x0020 +#define OXYGEN_I2S_MCLK_SHIFT 4 +#define MCLK_128 0 +#define MCLK_256 1 +#define MCLK_512 2 +#define OXYGEN_I2S_MCLK(f) (((f) & 3) << OXYGEN_I2S_MCLK_SHIFT) #define OXYGEN_I2S_BITS_MASK 0x00c0 #define OXYGEN_I2S_BITS_16 0x0000 #define OXYGEN_I2S_BITS_20 0x0040 @@ -238,11 +240,11 @@ #define OXYGEN_SPI_DATA_LENGTH_MASK 0x02 #define OXYGEN_SPI_DATA_LENGTH_2 0x00 #define OXYGEN_SPI_DATA_LENGTH_3 0x02 -#define OXYGEN_SPI_CLOCK_MASK 0xc0 +#define OXYGEN_SPI_CLOCK_MASK 0x0c #define OXYGEN_SPI_CLOCK_160 0x00 /* ns */ -#define OXYGEN_SPI_CLOCK_320 0x40 -#define OXYGEN_SPI_CLOCK_640 0x80 -#define OXYGEN_SPI_CLOCK_1280 0xc0 +#define OXYGEN_SPI_CLOCK_320 0x04 +#define OXYGEN_SPI_CLOCK_640 0x08 +#define OXYGEN_SPI_CLOCK_1280 0x0c #define OXYGEN_SPI_CODEC_MASK 0x70 /* 0..5 */ #define OXYGEN_SPI_CODEC_SHIFT 4 #define OXYGEN_SPI_CEN_MASK 0x80 diff --git a/sound/pci/oxygen/xonar.h b/sound/pci/oxygen/xonar.h index b35343b0a9a5..0434c207e811 100644 --- a/sound/pci/oxygen/xonar.h +++ b/sound/pci/oxygen/xonar.h @@ -24,6 +24,8 @@ void xonar_init_ext_power(struct oxygen *chip); void xonar_init_cs53x1(struct oxygen *chip); void xonar_set_cs53x1_params(struct oxygen *chip, struct snd_pcm_hw_params *params); + +#define XONAR_GPIO_BIT_INVERT (1 << 16) int xonar_gpio_bit_switch_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value); int xonar_gpio_bit_switch_put(struct snd_kcontrol *ctl, diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index aa27c31049af..9f72d424969c 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -22,29 +22,28 @@ * * CMI8788: * - * I²C <-> CS4398 (front) - * <-> CS4362A (surround, center/LFE, back) + * I²C <-> CS4398 (addr 1001111) (front) + * <-> CS4362A (addr 0011000) (surround, center/LFE, back) * - * GPI 0 <- external power present (DX only) + * GPI 0 <- external power present (DX only) * - * GPIO 0 -> enable output to speakers - * GPIO 1 -> enable front panel I/O - * GPIO 2 -> M0 of CS5361 - * GPIO 3 -> M1 of CS5361 - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * GPIO 0 -> enable output to speakers + * GPIO 1 -> route output to front panel + * GPIO 2 -> M0 of CS5361 + * GPIO 3 -> M1 of CS5361 + * GPIO 6 -> ? + * GPIO 7 -> ? + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) * - * CS4398: - * - * AD0 <- 1 - * AD1 <- 1 + * CM9780: * - * CS4362A: + * LINE_OUT -> input of ADC * - * AD0 <- 0 + * AUX_IN <- aux + * MIC_IN <- mic + * FMIC_IN <- front mic * - * CM9780: - * - * GPO 0 -> route line-in (0) or AC97 output (1) to CS5361 input + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5361 input */ #include <linux/pci.h> @@ -63,6 +62,7 @@ #define GPI_EXT_POWER 0x01 #define GPIO_D1_OUTPUT_ENABLE 0x0001 #define GPIO_D1_FRONT_PANEL 0x0002 +#define GPIO_D1_MAGIC 0x00c0 #define GPIO_D1_INPUT_ROUTE 0x0100 #define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ @@ -169,12 +169,12 @@ static void xonar_d1_init(struct oxygen *chip) cs43xx_registers_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); + GPIO_D1_FRONT_PANEL | + GPIO_D1_MAGIC | + GPIO_D1_INPUT_ROUTE); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); - oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); - xonar_init_cs53x1(chip); xonar_enable_output(chip); @@ -284,7 +284,7 @@ static void update_cs43xx_center_lfe_mix(struct oxygen *chip, bool mixed) static const struct snd_kcontrol_new front_panel_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Front Panel Switch", + .name = "Front Panel Playback Switch", .info = snd_ctl_boolean_mono_info, .get = xonar_gpio_bit_switch_get, .put = xonar_gpio_bit_switch_put, @@ -298,13 +298,7 @@ static int rolloff_info(struct snd_kcontrol *ctl, "Fast Roll-off", "Slow Roll-off" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int rolloff_get(struct snd_kcontrol *ctl, @@ -380,6 +374,30 @@ static int xonar_d1_mixer_init(struct oxygen *chip) return 0; } +static void dump_cs4362a_registers(struct xonar_cs43xx *data, + struct snd_info_buffer *buffer) +{ + unsigned int i; + + snd_iprintf(buffer, "\nCS4362A:"); + for (i = 1; i <= 14; ++i) + snd_iprintf(buffer, " %02x", data->cs4362a_regs[i]); + snd_iprintf(buffer, "\n"); +} + +static void dump_d1_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_cs43xx *data = chip->model_data; + unsigned int i; + + snd_iprintf(buffer, "\nCS4398: 7?"); + for (i = 2; i <= 8; ++i) + snd_iprintf(buffer, " %02x", data->cs4398_regs[i]); + snd_iprintf(buffer, "\n"); + dump_cs4362a_registers(data, buffer); +} + static const struct oxygen_model model_xonar_d1 = { .longname = "Asus Virtuoso 100", .chip = "AV200", @@ -388,22 +406,26 @@ static const struct oxygen_model model_xonar_d1 = { .cleanup = xonar_d1_cleanup, .suspend = xonar_d1_suspend, .resume = xonar_d1_resume, - .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_cs43xx_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_cs43xx_volume, .update_dac_mute = update_cs43xx_mute, .update_center_lfe_mix = update_cs43xx_center_lfe_mix, .ac97_switch = xonar_d1_line_mic_ac97_switch, + .dump_registers = dump_d1_registers, .dac_tlv = cs4362a_db_scale, .model_data_size = sizeof(struct xonar_cs43xx), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, - .dac_channels = 8, + CAPTURE_0_FROM_I2S_2 | + AC97_FMIC_SWITCH, + .dac_channels_pcm = 8, + .dac_channels_mixer = 8, .dac_volume_min = 127 - 60, .dac_volume_max = 127, .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_mclks = OXYGEN_MCLKS(256, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c new file mode 100644 index 000000000000..e1fa602eba79 --- /dev/null +++ b/sound/pci/oxygen/xonar_dg.c @@ -0,0 +1,572 @@ +/* + * card driver for the Xonar DG + * + * Copyright (c) Clemens Ladisch <clemens@ladisch.de> + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * 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 driver; if not, see <http://www.gnu.org/licenses/>. + */ + +/* + * Xonar DG + * -------- + * + * CMI8788: + * + * SPI 0 -> CS4245 + * + * GPIO 3 <- ? + * GPIO 4 <- headphone detect + * GPIO 5 -> route input jack to line-in (0) or mic-in (1) + * GPIO 6 -> route input jack to line-in (0) or mic-in (1) + * GPIO 7 -> enable rear headphone amp + * GPIO 8 -> enable output to speakers + * + * CS4245: + * + * input 1 <- aux + * input 2 <- front mic + * input 4 <- line/mic + * aux out -> front panel headphones + */ + +#include <linux/pci.h> +#include <linux/delay.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include <sound/tlv.h> +#include "oxygen.h" +#include "xonar_dg.h" +#include "cs4245.h" + +#define GPIO_MAGIC 0x0008 +#define GPIO_HP_DETECT 0x0010 +#define GPIO_INPUT_ROUTE 0x0060 +#define GPIO_HP_REAR 0x0080 +#define GPIO_OUTPUT_ENABLE 0x0100 + +struct dg { + unsigned int output_sel; + s8 input_vol[4][2]; + unsigned int input_sel; + u8 hp_vol_att; + u8 cs4245_regs[0x11]; +}; + +static void cs4245_write(struct oxygen *chip, unsigned int reg, u8 value) +{ + struct dg *data = chip->model_data; + + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_3 | + OXYGEN_SPI_CLOCK_1280 | + (0 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_HI, + CS4245_SPI_ADDRESS | + CS4245_SPI_WRITE | + (reg << 8) | value); + data->cs4245_regs[reg] = value; +} + +static void cs4245_write_cached(struct oxygen *chip, unsigned int reg, u8 value) +{ + struct dg *data = chip->model_data; + + if (value != data->cs4245_regs[reg]) + cs4245_write(chip, reg, value); +} + +static void cs4245_registers_init(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + cs4245_write(chip, CS4245_POWER_CTRL, CS4245_PDN); + cs4245_write(chip, CS4245_DAC_CTRL_1, + data->cs4245_regs[CS4245_DAC_CTRL_1]); + cs4245_write(chip, CS4245_ADC_CTRL, + data->cs4245_regs[CS4245_ADC_CTRL]); + cs4245_write(chip, CS4245_SIGNAL_SEL, + data->cs4245_regs[CS4245_SIGNAL_SEL]); + cs4245_write(chip, CS4245_PGA_B_CTRL, + data->cs4245_regs[CS4245_PGA_B_CTRL]); + cs4245_write(chip, CS4245_PGA_A_CTRL, + data->cs4245_regs[CS4245_PGA_A_CTRL]); + cs4245_write(chip, CS4245_ANALOG_IN, + data->cs4245_regs[CS4245_ANALOG_IN]); + cs4245_write(chip, CS4245_DAC_A_CTRL, + data->cs4245_regs[CS4245_DAC_A_CTRL]); + cs4245_write(chip, CS4245_DAC_B_CTRL, + data->cs4245_regs[CS4245_DAC_B_CTRL]); + cs4245_write(chip, CS4245_DAC_CTRL_2, + CS4245_DAC_SOFT | CS4245_DAC_ZERO | CS4245_INVERT_DAC); + cs4245_write(chip, CS4245_INT_MASK, 0); + cs4245_write(chip, CS4245_POWER_CTRL, 0); +} + +static void cs4245_init(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + data->cs4245_regs[CS4245_DAC_CTRL_1] = + CS4245_DAC_FM_SINGLE | CS4245_DAC_DIF_LJUST; + data->cs4245_regs[CS4245_ADC_CTRL] = + CS4245_ADC_FM_SINGLE | CS4245_ADC_DIF_LJUST; + data->cs4245_regs[CS4245_SIGNAL_SEL] = + CS4245_A_OUT_SEL_HIZ | CS4245_ASYNCH; + data->cs4245_regs[CS4245_PGA_B_CTRL] = 0; + data->cs4245_regs[CS4245_PGA_A_CTRL] = 0; + data->cs4245_regs[CS4245_ANALOG_IN] = + CS4245_PGA_SOFT | CS4245_PGA_ZERO | CS4245_SEL_INPUT_4; + data->cs4245_regs[CS4245_DAC_A_CTRL] = 0; + data->cs4245_regs[CS4245_DAC_B_CTRL] = 0; + cs4245_registers_init(chip); + snd_component_add(chip->card, "CS4245"); +} + +static void dg_output_enable(struct oxygen *chip) +{ + msleep(2500); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); +} + +static void dg_init(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + data->output_sel = 0; + data->input_sel = 3; + data->hp_vol_att = 2 * 16; + + cs4245_init(chip); + + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_MAGIC | GPIO_HP_DETECT); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_INPUT_ROUTE | GPIO_HP_REAR | GPIO_OUTPUT_ENABLE); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_INPUT_ROUTE | GPIO_HP_REAR); + dg_output_enable(chip); +} + +static void dg_cleanup(struct oxygen *chip) +{ + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); +} + +static void dg_suspend(struct oxygen *chip) +{ + dg_cleanup(chip); +} + +static void dg_resume(struct oxygen *chip) +{ + cs4245_registers_init(chip); + dg_output_enable(chip); +} + +static void set_cs4245_dac_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct dg *data = chip->model_data; + u8 value; + + value = data->cs4245_regs[CS4245_DAC_CTRL_1] & ~CS4245_DAC_FM_MASK; + if (params_rate(params) <= 50000) + value |= CS4245_DAC_FM_SINGLE; + else if (params_rate(params) <= 100000) + value |= CS4245_DAC_FM_DOUBLE; + else + value |= CS4245_DAC_FM_QUAD; + cs4245_write_cached(chip, CS4245_DAC_CTRL_1, value); +} + +static void set_cs4245_adc_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct dg *data = chip->model_data; + u8 value; + + value = data->cs4245_regs[CS4245_ADC_CTRL] & ~CS4245_ADC_FM_MASK; + if (params_rate(params) <= 50000) + value |= CS4245_ADC_FM_SINGLE; + else if (params_rate(params) <= 100000) + value |= CS4245_ADC_FM_DOUBLE; + else + value |= CS4245_ADC_FM_QUAD; + cs4245_write_cached(chip, CS4245_ADC_CTRL, value); +} + +static int output_switch_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "Speakers", "Headphones", "FP Headphones" + }; + + return snd_ctl_enum_info(info, 1, 3, names); +} + +static int output_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.enumerated.item[0] = data->output_sel; + mutex_unlock(&chip->mutex); + return 0; +} + +static int output_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + u8 reg; + int changed; + + if (value->value.enumerated.item[0] > 2) + return -EINVAL; + + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != data->output_sel; + if (changed) { + data->output_sel = value->value.enumerated.item[0]; + + reg = data->cs4245_regs[CS4245_SIGNAL_SEL] & + ~CS4245_A_OUT_SEL_MASK; + reg |= data->output_sel == 2 ? + CS4245_A_OUT_SEL_DAC : CS4245_A_OUT_SEL_HIZ; + cs4245_write_cached(chip, CS4245_SIGNAL_SEL, reg); + + cs4245_write_cached(chip, CS4245_DAC_A_CTRL, + data->output_sel ? data->hp_vol_att : 0); + cs4245_write_cached(chip, CS4245_DAC_B_CTRL, + data->output_sel ? data->hp_vol_att : 0); + + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + data->output_sel == 1 ? GPIO_HP_REAR : 0, + GPIO_HP_REAR); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int hp_volume_offset_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "< 64 ohms", "64-150 ohms", "150-300 ohms" + }; + + return snd_ctl_enum_info(info, 1, 3, names); +} + +static int hp_volume_offset_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + if (data->hp_vol_att > 2 * 7) + value->value.enumerated.item[0] = 0; + else if (data->hp_vol_att > 0) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + mutex_unlock(&chip->mutex); + return 0; +} + +static int hp_volume_offset_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + static const s8 atts[3] = { 2 * 16, 2 * 7, 0 }; + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + s8 att; + int changed; + + if (value->value.enumerated.item[0] > 2) + return -EINVAL; + att = atts[value->value.enumerated.item[0]]; + mutex_lock(&chip->mutex); + changed = att != data->hp_vol_att; + if (changed) { + data->hp_vol_att = att; + if (data->output_sel) { + cs4245_write_cached(chip, CS4245_DAC_A_CTRL, att); + cs4245_write_cached(chip, CS4245_DAC_B_CTRL, att); + } + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int input_vol_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 2 * -12; + info->value.integer.max = 2 * 12; + return 0; +} + +static int input_vol_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + unsigned int idx = ctl->private_value; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = data->input_vol[idx][0]; + value->value.integer.value[1] = data->input_vol[idx][1]; + mutex_unlock(&chip->mutex); + return 0; +} + +static int input_vol_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + unsigned int idx = ctl->private_value; + int changed = 0; + + if (value->value.integer.value[0] < 2 * -12 || + value->value.integer.value[0] > 2 * 12 || + value->value.integer.value[1] < 2 * -12 || + value->value.integer.value[1] > 2 * 12) + return -EINVAL; + mutex_lock(&chip->mutex); + changed = data->input_vol[idx][0] != value->value.integer.value[0] || + data->input_vol[idx][1] != value->value.integer.value[1]; + if (changed) { + data->input_vol[idx][0] = value->value.integer.value[0]; + data->input_vol[idx][1] = value->value.integer.value[1]; + if (idx == data->input_sel) { + cs4245_write_cached(chip, CS4245_PGA_A_CTRL, + data->input_vol[idx][0]); + cs4245_write_cached(chip, CS4245_PGA_B_CTRL, + data->input_vol[idx][1]); + } + } + mutex_unlock(&chip->mutex); + return changed; +} + +static DECLARE_TLV_DB_SCALE(cs4245_pga_db_scale, -1200, 50, 0); + +static int input_sel_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[4] = { + "Mic", "Aux", "Front Mic", "Line" + }; + + return snd_ctl_enum_info(info, 1, 4, names); +} + +static int input_sel_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.enumerated.item[0] = data->input_sel; + mutex_unlock(&chip->mutex); + return 0; +} + +static int input_sel_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + static const u8 sel_values[4] = { + CS4245_SEL_MIC, + CS4245_SEL_INPUT_1, + CS4245_SEL_INPUT_2, + CS4245_SEL_INPUT_4 + }; + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + int changed; + + if (value->value.enumerated.item[0] > 3) + return -EINVAL; + + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != data->input_sel; + if (changed) { + data->input_sel = value->value.enumerated.item[0]; + + cs4245_write(chip, CS4245_ANALOG_IN, + (data->cs4245_regs[CS4245_ANALOG_IN] & + ~CS4245_SEL_MASK) | + sel_values[data->input_sel]); + + cs4245_write_cached(chip, CS4245_PGA_A_CTRL, + data->input_vol[data->input_sel][0]); + cs4245_write_cached(chip, CS4245_PGA_B_CTRL, + data->input_vol[data->input_sel][1]); + + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + data->input_sel ? 0 : GPIO_INPUT_ROUTE, + GPIO_INPUT_ROUTE); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { "Active", "Frozen" }; + + return snd_ctl_enum_info(info, 1, 2, names); +} + +static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + value->value.enumerated.item[0] = + !!(data->cs4245_regs[CS4245_ADC_CTRL] & CS4245_HPF_FREEZE); + return 0; +} + +static int hpf_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + u8 reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->cs4245_regs[CS4245_ADC_CTRL] & ~CS4245_HPF_FREEZE; + if (value->value.enumerated.item[0]) + reg |= CS4245_HPF_FREEZE; + changed = reg != data->cs4245_regs[CS4245_ADC_CTRL]; + if (changed) + cs4245_write(chip, CS4245_ADC_CTRL, reg); + mutex_unlock(&chip->mutex); + return changed; +} + +#define INPUT_VOLUME(xname, index) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = input_vol_info, \ + .get = input_vol_get, \ + .put = input_vol_put, \ + .tlv = { .p = cs4245_pga_db_scale }, \ + .private_value = index, \ +} +static const struct snd_kcontrol_new dg_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Output Playback Enum", + .info = output_switch_info, + .get = output_switch_get, + .put = output_switch_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphones Impedance Playback Enum", + .info = hp_volume_offset_info, + .get = hp_volume_offset_get, + .put = hp_volume_offset_put, + }, + INPUT_VOLUME("Mic Capture Volume", 0), + INPUT_VOLUME("Aux Capture Volume", 1), + INPUT_VOLUME("Front Mic Capture Volume", 2), + INPUT_VOLUME("Line Capture Volume", 3), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = input_sel_info, + .get = input_sel_get, + .put = input_sel_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC High-pass Filter Capture Enum", + .info = hpf_info, + .get = hpf_get, + .put = hpf_put, + }, +}; + +static int dg_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "Master Playback ", 16)) + return 1; + return 0; +} + +static int dg_mixer_init(struct oxygen *chip) +{ + unsigned int i; + int err; + + for (i = 0; i < ARRAY_SIZE(dg_controls); ++i) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(&dg_controls[i], chip)); + if (err < 0) + return err; + } + return 0; +} + +static void dump_cs4245_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct dg *data = chip->model_data; + unsigned int i; + + snd_iprintf(buffer, "\nCS4245:"); + for (i = 1; i <= 0x10; ++i) + snd_iprintf(buffer, " %02x", data->cs4245_regs[i]); + snd_iprintf(buffer, "\n"); +} + +struct oxygen_model model_xonar_dg = { + .shortname = "Xonar DG", + .longname = "C-Media Oxygen HD Audio", + .chip = "CMI8786", + .init = dg_init, + .control_filter = dg_control_filter, + .mixer_init = dg_mixer_init, + .cleanup = dg_cleanup, + .suspend = dg_suspend, + .resume = dg_resume, + .set_dac_params = set_cs4245_dac_params, + .set_adc_params = set_cs4245_adc_params, + .dump_registers = dump_cs4245_registers, + .model_data_size = sizeof(struct dg), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2, + .dac_channels_pcm = 6, + .dac_channels_mixer = 0, + .function_flags = OXYGEN_FUNCTION_SPI, + .dac_mclks = OXYGEN_MCLKS(256, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; diff --git a/sound/pci/oxygen/xonar_dg.h b/sound/pci/oxygen/xonar_dg.h new file mode 100644 index 000000000000..5688d78609a9 --- /dev/null +++ b/sound/pci/oxygen/xonar_dg.h @@ -0,0 +1,8 @@ +#ifndef XONAR_DG_H_INCLUDED +#define XONAR_DG_H_INCLUDED + +#include "oxygen.h" + +extern struct oxygen_model model_xonar_dg; + +#endif diff --git a/sound/pci/oxygen/xonar_hdmi.c b/sound/pci/oxygen/xonar_hdmi.c index b12db1f1cea9..136dac6a3964 100644 --- a/sound/pci/oxygen/xonar_hdmi.c +++ b/sound/pci/oxygen/xonar_hdmi.c @@ -1,5 +1,5 @@ /* - * helper functions for HDMI models (Xonar HDAV1.3) + * helper functions for HDMI models (Xonar HDAV1.3/HDAV1.3 Slim) * * Copyright (c) Clemens Ladisch <clemens@ladisch.de> * diff --git a/sound/pci/oxygen/xonar_lib.c b/sound/pci/oxygen/xonar_lib.c index b3ff71316653..0ebe7f5916f9 100644 --- a/sound/pci/oxygen/xonar_lib.c +++ b/sound/pci/oxygen/xonar_lib.c @@ -104,9 +104,10 @@ int xonar_gpio_bit_switch_get(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; u16 bit = ctl->private_value; + bool invert = ctl->private_value & XONAR_GPIO_BIT_INVERT; value->value.integer.value[0] = - !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & bit); + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & bit) ^ invert; return 0; } @@ -115,12 +116,13 @@ int xonar_gpio_bit_switch_put(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; u16 bit = ctl->private_value; + bool invert = ctl->private_value & XONAR_GPIO_BIT_INVERT; u16 old_bits, new_bits; int changed; spin_lock_irq(&chip->reg_lock); old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); - if (value->value.integer.value[0]) + if (!!value->value.integer.value[0] ^ invert) new_bits = old_bits | bit; else new_bits = old_bits & ~bit; diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index d491fd6c0be2..54cad38ec30a 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -22,20 +22,26 @@ * * CMI8788: * - * SPI 0 -> 1st PCM1796 (front) - * SPI 1 -> 2nd PCM1796 (surround) - * SPI 2 -> 3rd PCM1796 (center/LFE) - * SPI 4 -> 4th PCM1796 (back) + * SPI 0 -> 1st PCM1796 (front) + * SPI 1 -> 2nd PCM1796 (surround) + * SPI 2 -> 3rd PCM1796 (center/LFE) + * SPI 4 -> 4th PCM1796 (back) * - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 5 <- external power present (D2X only) - * GPIO 7 -> ALT - * GPIO 8 -> enable output to speakers + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 5 <- external power present (D2X only) + * GPIO 7 -> ALT + * GPIO 8 -> enable output to speakers * * CM9780: * - * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * LINE_OUT -> input of ADC + * + * AUX_IN <- aux + * VIDEO_IN <- CD + * FMIC_IN <- mic + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input */ /* @@ -44,52 +50,53 @@ * * CMI8788: * - * I²C <-> PCM1796 (front) + * I²C <-> PCM1796 (addr 1001100) (front) * - * GPI 0 <- external power present + * GPI 0 <- external power present * - * GPIO 0 -> enable output to speakers - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * GPIO 0 -> enable HDMI (0) or speaker (1) output + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 4 <- daughterboard detection + * GPIO 5 <- daughterboard detection + * GPIO 6 -> ? + * GPIO 7 -> ? + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) * - * TXD -> HDMI controller - * RXD <- HDMI controller - * - * PCM1796 front: AD1,0 <- 0,0 + * UART <-> HDMI controller * * CM9780: * - * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * LINE_OUT -> input of ADC + * + * AUX_IN <- aux + * CD_IN <- CD + * MIC_IN <- mic + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input * * no daughterboard * ---------------- * - * GPIO 4 <- 1 + * GPIO 4 <- 1 * * H6 daughterboard * ---------------- * - * GPIO 4 <- 0 - * GPIO 5 <- 0 - * - * I²C <-> PCM1796 (surround) - * <-> PCM1796 (center/LFE) - * <-> PCM1796 (back) + * GPIO 4 <- 0 + * GPIO 5 <- 0 * - * PCM1796 surround: AD1,0 <- 0,1 - * PCM1796 center/LFE: AD1,0 <- 1,0 - * PCM1796 back: AD1,0 <- 1,1 + * I²C <-> PCM1796 (addr 1001101) (surround) + * <-> PCM1796 (addr 1001110) (center/LFE) + * <-> PCM1796 (addr 1001111) (back) * * unknown daughterboard * --------------------- * - * GPIO 4 <- 0 - * GPIO 5 <- 1 - * - * I²C <-> CS4362A (surround, center/LFE, back) + * GPIO 4 <- 0 + * GPIO 5 <- 1 * - * CS4362A: AD0 <- 0 + * I²C <-> CS4362A (addr 0011000) (surround, center/LFE, back) */ /* @@ -98,32 +105,35 @@ * * CMI8788: * - * I²C <-> PCM1792A - * <-> CS2000 (ST only) + * I²C <-> PCM1792A (addr 1001100) + * <-> CS2000 (addr 1001110) (ST only) * - * ADC1 MCLK -> REF_CLK of CS2000 (ST only) + * ADC1 MCLK -> REF_CLK of CS2000 (ST only) * - * GPI 0 <- external power present (STX only) + * GPI 0 <- external power present (STX only) * - * GPIO 0 -> enable output to speakers - * GPIO 1 -> route HP to front panel (0) or rear jack (1) - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 7 -> route output to speaker jacks (0) or HP (1) - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * GPIO 0 -> enable output to speakers + * GPIO 1 -> route HP to front panel (0) or rear jack (1) + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 4 <- daughterboard detection + * GPIO 5 <- daughterboard detection + * GPIO 6 -> ? + * GPIO 7 -> route output to speaker jacks (0) or HP (1) + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) * * PCM1792A: * - * AD1,0 <- 0,0 - * SCK <- CLK_OUT of CS2000 (ST only) + * SCK <- CLK_OUT of CS2000 (ST only) * - * CS2000: + * CM9780: * - * AD0 <- 0 + * LINE_OUT -> input of ADC * - * CM9780: + * AUX_IN <- aux + * MIC_IN <- mic * - * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input * * H6 daughterboard * ---------------- @@ -133,15 +143,39 @@ */ /* - * Xonar HDAV1.3 Slim - * ------------------ + * Xonar Xense + * ----------- * * CMI8788: * - * GPIO 1 -> enable output + * I²C <-> PCM1796 (addr 1001100) (front) + * <-> CS4362A (addr 0011000) (surround, center/LFE, back) + * <-> CS2000 (addr 1001110) + * + * ADC1 MCLK -> REF_CLK of CS2000 + * + * GPI 0 <- external power present + * + * GPIO 0 -> enable output + * GPIO 1 -> route HP to front panel (0) or rear jack (1) + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 4 -> enable output + * GPIO 5 -> enable output + * GPIO 6 -> ? + * GPIO 7 -> route output to HP (0) or speaker (1) + * GPIO 8 -> route input jack to mic-in (0) or line-in (1) * - * TXD -> HDMI controller - * RXD <- HDMI controller + * CM9780: + * + * LINE_OUT -> input of ADC + * + * AUX_IN <- aux + * VIDEO_IN <- ? + * FMIC_IN <- mic + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * GPO 1 -> route mic-in from input jack (0) or front panel header (1) */ #include <linux/pci.h> @@ -150,6 +184,7 @@ #include <sound/ac97_codec.h> #include <sound/control.h> #include <sound/core.h> +#include <sound/info.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/tlv.h> @@ -167,12 +202,14 @@ #define GPIO_INPUT_ROUTE 0x0100 #define GPIO_HDAV_OUTPUT_ENABLE 0x0001 +#define GPIO_HDAV_MAGIC 0x00c0 #define GPIO_DB_MASK 0x0030 #define GPIO_DB_H6 0x0000 #define GPIO_ST_OUTPUT_ENABLE 0x0001 #define GPIO_ST_HP_REAR 0x0002 +#define GPIO_ST_MAGIC 0x0040 #define GPIO_ST_HP 0x0080 #define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ii, /W=0 */ @@ -186,11 +223,12 @@ struct xonar_pcm179x { unsigned int dacs; u8 pcm1796_regs[4][5]; unsigned int current_rate; - bool os_128; + bool h6; bool hp_active; s8 hp_gain_offset; bool has_cs2000; - u8 cs2000_fun_cfg_1; + u8 cs2000_regs[0x1f]; + bool broken_i2c; }; struct xonar_hdav { @@ -249,16 +287,14 @@ static void cs2000_write(struct oxygen *chip, u8 reg, u8 value) struct xonar_pcm179x *data = chip->model_data; oxygen_write_i2c(chip, I2C_DEVICE_CS2000, reg, value); - if (reg == CS2000_FUN_CFG_1) - data->cs2000_fun_cfg_1 = value; + data->cs2000_regs[reg] = value; } static void cs2000_write_cached(struct oxygen *chip, u8 reg, u8 value) { struct xonar_pcm179x *data = chip->model_data; - if (reg != CS2000_FUN_CFG_1 || - value != data->cs2000_fun_cfg_1) + if (value != data->cs2000_regs[reg]) cs2000_write(chip, reg, value); } @@ -268,6 +304,7 @@ static void pcm1796_registers_init(struct oxygen *chip) unsigned int i; s8 gain_offset; + msleep(1); gain_offset = data->hp_active ? data->hp_gain_offset : 0; for (i = 0; i < data->dacs; ++i) { /* set ATLD before ATL/ATR */ @@ -282,6 +319,7 @@ static void pcm1796_registers_init(struct oxygen *chip) pcm1796_write(chip, i, 20, data->pcm1796_regs[0][20 - PCM1796_REG_BASE]); pcm1796_write(chip, i, 21, 0); + gain_offset = 0; } } @@ -290,10 +328,11 @@ static void pcm1796_init(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; data->pcm1796_regs[0][18 - PCM1796_REG_BASE] = PCM1796_MUTE | - PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; data->pcm1796_regs[0][19 - PCM1796_REG_BASE] = PCM1796_FLT_SHARP | PCM1796_ATS_1; - data->pcm1796_regs[0][20 - PCM1796_REG_BASE] = PCM1796_OS_64; + data->pcm1796_regs[0][20 - PCM1796_REG_BASE] = + data->h6 ? PCM1796_OS_64 : PCM1796_OS_128; pcm1796_registers_init(chip); data->current_rate = 48000; } @@ -339,18 +378,20 @@ static void xonar_hdav_init(struct oxygen *chip) oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); + OXYGEN_2WIRE_SPEED_STANDARD); data->pcm179x.generic.anti_pop_delay = 100; data->pcm179x.generic.output_enable_bit = GPIO_HDAV_OUTPUT_ENABLE; data->pcm179x.generic.ext_power_reg = OXYGEN_GPI_DATA; data->pcm179x.generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->pcm179x.generic.ext_power_bit = GPI_EXT_POWER; - data->pcm179x.dacs = chip->model.private_data ? 4 : 1; + data->pcm179x.dacs = chip->model.dac_channels_mixer / 2; + data->pcm179x.h6 = chip->model.dac_channels_mixer > 2; pcm1796_init(chip); - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_HDAV_MAGIC | GPIO_INPUT_ROUTE); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_INPUT_ROUTE); xonar_init_cs53x1(chip); @@ -367,7 +408,7 @@ static void xonar_st_init_i2c(struct oxygen *chip) oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); + OXYGEN_2WIRE_SPEED_STANDARD); } static void xonar_st_init_common(struct oxygen *chip) @@ -375,13 +416,14 @@ static void xonar_st_init_common(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; data->generic.output_enable_bit = GPIO_ST_OUTPUT_ENABLE; - data->dacs = chip->model.private_data ? 4 : 1; + data->dacs = chip->model.dac_channels_mixer / 2; data->hp_gain_offset = 2*-18; pcm1796_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | + GPIO_ST_MAGIC | GPIO_ST_HP); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); @@ -410,9 +452,11 @@ static void cs2000_registers_init(struct oxygen *chip) cs2000_write(chip, CS2000_RATIO_0 + 1, 0x10); cs2000_write(chip, CS2000_RATIO_0 + 2, 0x00); cs2000_write(chip, CS2000_RATIO_0 + 3, 0x00); - cs2000_write(chip, CS2000_FUN_CFG_1, data->cs2000_fun_cfg_1); + cs2000_write(chip, CS2000_FUN_CFG_1, + data->cs2000_regs[CS2000_FUN_CFG_1]); cs2000_write(chip, CS2000_FUN_CFG_2, 0); cs2000_write(chip, CS2000_GLOBAL_CFG, CS2000_EN_DEV_CFG_2); + msleep(3); /* PLL lock delay */ } static void xonar_st_init(struct oxygen *chip) @@ -420,13 +464,18 @@ static void xonar_st_init(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; data->generic.anti_pop_delay = 100; + data->h6 = chip->model.dac_channels_mixer > 2; data->has_cs2000 = 1; - data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_1; + data->cs2000_regs[CS2000_FUN_CFG_1] = CS2000_REF_CLK_DIV_1; + data->broken_i2c = true; oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, - OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_I2S | - OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | - OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + OXYGEN_RATE_48000 | + OXYGEN_I2S_FORMAT_I2S | + OXYGEN_I2S_MCLK(data->h6 ? MCLK_256 : MCLK_512) | + OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | + OXYGEN_I2S_BCLK_64); xonar_st_init_i2c(chip); cs2000_registers_init(chip); @@ -507,44 +556,16 @@ static void xonar_st_resume(struct oxygen *chip) xonar_stx_resume(chip); } -static unsigned int mclk_from_rate(struct oxygen *chip, unsigned int rate) -{ - struct xonar_pcm179x *data = chip->model_data; - - if (rate <= 32000) - return OXYGEN_I2S_MCLK_512; - else if (rate <= 48000 && data->os_128) - return OXYGEN_I2S_MCLK_512; - else if (rate <= 96000) - return OXYGEN_I2S_MCLK_256; - else - return OXYGEN_I2S_MCLK_128; -} - -static unsigned int get_pcm1796_i2s_mclk(struct oxygen *chip, - unsigned int channel, - struct snd_pcm_hw_params *params) -{ - if (channel == PCM_MULTICH) - return mclk_from_rate(chip, params_rate(params)); - else - return oxygen_default_i2s_mclk(chip, channel, params); -} - static void update_pcm1796_oversampling(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; unsigned int i; u8 reg; - if (data->current_rate <= 32000) + if (data->current_rate <= 48000 && !data->h6) reg = PCM1796_OS_128; - else if (data->current_rate <= 48000 && data->os_128) - reg = PCM1796_OS_128; - else if (data->current_rate <= 96000 || data->os_128) - reg = PCM1796_OS_64; else - reg = PCM1796_OS_32; + reg = PCM1796_OS_64; for (i = 0; i < data->dacs; ++i) pcm1796_write_cached(chip, i, 20, reg); } @@ -554,6 +575,7 @@ static void set_pcm1796_params(struct oxygen *chip, { struct xonar_pcm179x *data = chip->model_data; + msleep(1); data->current_rate = params_rate(params); update_pcm1796_oversampling(chip); } @@ -570,6 +592,7 @@ static void update_pcm1796_volume(struct oxygen *chip) + gain_offset); pcm1796_write_cached(chip, i, 17, chip->dac_volume[i * 2 + 1] + gain_offset); + gain_offset = 0; } } @@ -579,7 +602,7 @@ static void update_pcm1796_mute(struct oxygen *chip) unsigned int i; u8 value; - value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; if (chip->dac_mute) value |= PCM1796_MUTE; for (i = 0; i < data->dacs; ++i) @@ -592,45 +615,35 @@ static void update_cs2000_rate(struct oxygen *chip, unsigned int rate) u8 rate_mclk, reg; switch (rate) { - /* XXX Why is the I2S A MCLK half the actual I2S MCLK? */ case 32000: - rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; - break; - case 44100: - if (data->os_128) - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; - else - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_128; - break; - default: /* 48000 */ - if (data->os_128) - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; - else - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_128; - break; case 64000: - rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_32000; break; + case 44100: case 88200: - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; - break; - case 96000: - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; - break; case 176400: - rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_44100; break; + default: + case 48000: + case 96000: case 192000: - rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + rate_mclk = OXYGEN_RATE_48000; break; } - oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, rate_mclk, - OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_MCLK_MASK); - if ((rate_mclk & OXYGEN_I2S_MCLK_MASK) <= OXYGEN_I2S_MCLK_128) + + if (rate <= 96000 && (rate > 48000 || data->h6)) { + rate_mclk |= OXYGEN_I2S_MCLK(MCLK_256); reg = CS2000_REF_CLK_DIV_1; - else + } else { + rate_mclk |= OXYGEN_I2S_MCLK(MCLK_512); reg = CS2000_REF_CLK_DIV_2; + } + + oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, rate_mclk, + OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_MCLK_MASK); cs2000_write_cached(chip, CS2000_FUN_CFG_1, reg); + msleep(3); /* PLL lock delay */ } static void set_st_params(struct oxygen *chip, @@ -665,13 +678,7 @@ static int rolloff_info(struct snd_kcontrol *ctl, "Sharp Roll-off", "Slow Roll-off" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int rolloff_get(struct snd_kcontrol *ctl, @@ -719,57 +726,13 @@ static const struct snd_kcontrol_new rolloff_control = { .put = rolloff_put, }; -static int os_128_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) -{ - static const char *const names[2] = { "64x", "128x" }; - - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; -} - -static int os_128_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct xonar_pcm179x *data = chip->model_data; - - value->value.enumerated.item[0] = data->os_128; - return 0; -} - -static int os_128_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct xonar_pcm179x *data = chip->model_data; - int changed; - - mutex_lock(&chip->mutex); - changed = value->value.enumerated.item[0] != data->os_128; - if (changed) { - data->os_128 = value->value.enumerated.item[0]; - if (data->has_cs2000) - update_cs2000_rate(chip, data->current_rate); - oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, - mclk_from_rate(chip, data->current_rate), - OXYGEN_I2S_MCLK_MASK); - update_pcm1796_oversampling(chip); - } - mutex_unlock(&chip->mutex); - return changed; -} - -static const struct snd_kcontrol_new os_128_control = { +static const struct snd_kcontrol_new hdav_hdmi_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "DAC Oversampling Playback Enum", - .info = os_128_info, - .get = os_128_get, - .put = os_128_put, + .name = "HDMI Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = xonar_gpio_bit_switch_get, + .put = xonar_gpio_bit_switch_put, + .private_value = GPIO_HDAV_OUTPUT_ENABLE | XONAR_GPIO_BIT_INVERT, }; static int st_output_switch_info(struct snd_kcontrol *ctl, @@ -779,13 +742,7 @@ static int st_output_switch_info(struct snd_kcontrol *ctl, "Speakers", "Headphones", "FP Headphones" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item >= 3) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 3, names); } static int st_output_switch_get(struct snd_kcontrol *ctl, @@ -840,13 +797,7 @@ static int st_hp_volume_offset_info(struct snd_kcontrol *ctl, "< 64 ohms", "64-300 ohms", "300-600 ohms" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item > 2) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 3, names); } static int st_hp_volume_offset_get(struct snd_kcontrol *ctl, @@ -928,16 +879,25 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template) return 0; } +static int xonar_st_h6_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "Master Playback ", 16)) + /* no volume/mute, as I²C to the third DAC does not work */ + return 1; + return 0; +} + static int add_pcm1796_controls(struct oxygen *chip) { + struct xonar_pcm179x *data = chip->model_data; int err; - err = snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); - if (err < 0) - return err; - err = snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); - if (err < 0) - return err; + if (!data->broken_i2c) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(&rolloff_control, chip)); + if (err < 0) + return err; + } return 0; } @@ -956,7 +916,15 @@ static int xonar_d2_mixer_init(struct oxygen *chip) static int xonar_hdav_mixer_init(struct oxygen *chip) { - return add_pcm1796_controls(chip); + int err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&hdav_hdmi_control, chip)); + if (err < 0) + return err; + err = add_pcm1796_controls(chip); + if (err < 0) + return err; + return 0; } static int xonar_st_mixer_init(struct oxygen *chip) @@ -976,6 +944,45 @@ static int xonar_st_mixer_init(struct oxygen *chip) return 0; } +static void dump_pcm1796_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int dac, i; + + for (dac = 0; dac < data->dacs; ++dac) { + snd_iprintf(buffer, "\nPCM1796 %u:", dac + 1); + for (i = 0; i < 5; ++i) + snd_iprintf(buffer, " %02x", + data->pcm1796_regs[dac][i]); + } + snd_iprintf(buffer, "\n"); +} + +static void dump_cs2000_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + + if (data->has_cs2000) { + snd_iprintf(buffer, "\nCS2000:\n00: "); + for (i = 1; i < 0x10; ++i) + snd_iprintf(buffer, " %02x", data->cs2000_regs[i]); + snd_iprintf(buffer, "\n10:"); + for (i = 0x10; i < 0x1f; ++i) + snd_iprintf(buffer, " %02x", data->cs2000_regs[i]); + snd_iprintf(buffer, "\n"); + } +} + +static void dump_st_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + dump_pcm1796_registers(chip, buffer); + dump_cs2000_registers(chip, buffer); +} + static const struct oxygen_model model_xonar_d2 = { .longname = "Asus Virtuoso 200", .chip = "AV200", @@ -985,11 +992,11 @@ static const struct oxygen_model model_xonar_d2 = { .cleanup = xonar_d2_cleanup, .suspend = xonar_d2_suspend, .resume = xonar_d2_resume, - .get_i2s_mclk = get_pcm1796_i2s_mclk, .set_dac_params = set_pcm1796_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, + .dump_registers = dump_pcm1796_registers, .dac_tlv = pcm1796_db_scale, .model_data_size = sizeof(struct xonar_pcm179x), .device_config = PLAYBACK_0_TO_I2S | @@ -999,13 +1006,16 @@ static const struct oxygen_model model_xonar_d2 = { MIDI_OUTPUT | MIDI_INPUT | AC97_CD_INPUT, - .dac_channels = 8, + .dac_channels_pcm = 8, + .dac_channels_mixer = 8, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .dac_mclks = OXYGEN_MCLKS(512, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), + .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1018,25 +1028,28 @@ static const struct oxygen_model model_xonar_hdav = { .suspend = xonar_hdav_suspend, .resume = xonar_hdav_resume, .pcm_hardware_filter = xonar_hdmi_pcm_hardware_filter, - .get_i2s_mclk = get_pcm1796_i2s_mclk, .set_dac_params = set_hdav_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, .uart_input = xonar_hdmi_uart_input, .ac97_switch = xonar_line_mic_ac97_switch, + .dump_registers = dump_pcm1796_registers, .dac_tlv = pcm1796_db_scale, .model_data_size = sizeof(struct xonar_hdav), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF, - .dac_channels = 8, + .dac_channels_pcm = 8, + .dac_channels_mixer = 2, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .dac_mclks = OXYGEN_MCLKS(512, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), + .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1048,22 +1061,26 @@ static const struct oxygen_model model_xonar_st = { .cleanup = xonar_st_cleanup, .suspend = xonar_st_suspend, .resume = xonar_st_resume, - .get_i2s_mclk = get_pcm1796_i2s_mclk, .set_dac_params = set_st_params, .set_adc_params = xonar_set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, .update_dac_mute = update_pcm1796_mute, .ac97_switch = xonar_line_mic_ac97_switch, + .dump_registers = dump_st_registers, .dac_tlv = pcm1796_db_scale, .model_data_size = sizeof(struct xonar_pcm179x), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, - .dac_channels = 2, + CAPTURE_0_FROM_I2S_2 | + AC97_FMIC_SWITCH, + .dac_channels_pcm = 2, + .dac_channels_mixer = 2, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .dac_mclks = OXYGEN_MCLKS(512, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), + .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1089,7 +1106,8 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, break; case GPIO_DB_H6: chip->model.shortname = "Xonar HDAV1.3+H6"; - chip->model.private_data = 1; + chip->model.dac_channels_mixer = 8; + chip->model.dac_mclks = OXYGEN_MCLKS(256, 128, 128); break; } break; @@ -1102,8 +1120,10 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, break; case GPIO_DB_H6: chip->model.shortname = "Xonar ST+H6"; - chip->model.dac_channels = 8; - chip->model.private_data = 1; + chip->model.control_filter = xonar_st_h6_control_filter; + chip->model.dac_channels_pcm = 8; + chip->model.dac_channels_mixer = 8; + chip->model.dac_mclks = OXYGEN_MCLKS(256, 128, 128); break; } break; @@ -1114,9 +1134,6 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; - case 0x835e: - snd_printk(KERN_ERR "the HDAV1.3 Slim is not supported\n"); - return -ENODEV; default: return -EINVAL; } diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 200f7601276f..42d1ab136217 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -1,5 +1,5 @@ /* - * card driver for models with WM8776/WM8766 DACs (Xonar DS) + * card driver for models with WM8776/WM8766 DACs (Xonar DS/HDAV1.3 Slim) * * Copyright (c) Clemens Ladisch <clemens@ladisch.de> * @@ -22,26 +22,48 @@ * * CMI8788: * - * SPI 0 -> WM8766 (surround, center/LFE, back) - * SPI 1 -> WM8776 (front, input) + * SPI 0 -> WM8766 (surround, center/LFE, back) + * SPI 1 -> WM8776 (front, input) * - * GPIO 4 <- headphone detect, 0 = plugged - * GPIO 6 -> route input jack to mic-in (0) or line-in (1) - * GPIO 7 -> enable output to front L/R speaker channels - * GPIO 8 -> enable output to other speaker channels and front panel headphone + * GPIO 4 <- headphone detect, 0 = plugged + * GPIO 6 -> route input jack to mic-in (0) or line-in (1) + * GPIO 7 -> enable output to front L/R speaker channels + * GPIO 8 -> enable output to other speaker channels and front panel headphone * - * WM8766: + * WM8776: * - * input 1 <- line - * input 2 <- mic - * input 3 <- front mic - * input 4 <- aux + * input 1 <- line + * input 2 <- mic + * input 3 <- front mic + * input 4 <- aux + */ + +/* + * Xonar HDAV1.3 Slim + * ------------------ + * + * CMI8788: + * + * I²C <-> WM8776 (addr 0011010) + * + * GPIO 0 -> disable HDMI output + * GPIO 1 -> enable HP output + * GPIO 6 -> firmware EEPROM I²C clock + * GPIO 7 <-> firmware EEPROM I²C data + * + * UART <-> HDMI controller + * + * WM8776: + * + * input 1 <- mic + * input 2 <- aux */ #include <linux/pci.h> #include <linux/delay.h> #include <sound/control.h> #include <sound/core.h> +#include <sound/info.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -55,6 +77,13 @@ #define GPIO_DS_OUTPUT_FRONTLR 0x0080 #define GPIO_DS_OUTPUT_ENABLE 0x0100 +#define GPIO_SLIM_HDMI_DISABLE 0x0001 +#define GPIO_SLIM_OUTPUT_ENABLE 0x0002 +#define GPIO_SLIM_FIRMWARE_CLK 0x0040 +#define GPIO_SLIM_FIRMWARE_DATA 0x0080 + +#define I2C_DEVICE_WM8776 0x34 /* 001101, 0, /W=0 */ + #define LC_CONTROL_LIMITER 0x40000000 #define LC_CONTROL_ALC 0x20000000 @@ -66,19 +95,37 @@ struct xonar_wm87x6 { struct snd_kcontrol *mic_adcmux_control; struct snd_kcontrol *lc_controls[13]; struct snd_jack *hp_jack; + struct xonar_hdmi hdmi; }; -static void wm8776_write(struct oxygen *chip, - unsigned int reg, unsigned int value) +static void wm8776_write_spi(struct oxygen *chip, + unsigned int reg, unsigned int value) { - struct xonar_wm87x6 *data = chip->model_data; - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | OXYGEN_SPI_CLOCK_160 | (1 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); +} + +static void wm8776_write_i2c(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + oxygen_write_i2c(chip, I2C_DEVICE_WM8776, + (reg << 1) | (value >> 8), value); +} + +static void wm8776_write(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + if ((chip->model.function_flags & OXYGEN_FUNCTION_2WIRE_SPI_MASK) == + OXYGEN_FUNCTION_SPI) + wm8776_write_spi(chip, reg, value); + else + wm8776_write_i2c(chip, reg, value); if (reg < ARRAY_SIZE(data->wm8776_regs)) { if (reg >= WM8776_HPLVOL && reg <= WM8776_DACMASTER) value &= ~WM8776_UPDATE; @@ -245,17 +292,50 @@ static void xonar_ds_init(struct oxygen *chip) snd_component_add(chip->card, "WM8766"); } +static void xonar_hdav_slim_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + data->generic.anti_pop_delay = 300; + data->generic.output_enable_bit = GPIO_SLIM_OUTPUT_ENABLE; + + wm8776_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_SLIM_HDMI_DISABLE | + GPIO_SLIM_FIRMWARE_CLK | + GPIO_SLIM_FIRMWARE_DATA); + + xonar_hdmi_init(chip, &data->hdmi); + xonar_enable_output(chip); + + snd_component_add(chip->card, "WM8776"); +} + static void xonar_ds_cleanup(struct oxygen *chip) { xonar_disable_output(chip); wm8776_write(chip, WM8776_RESET, 0); } +static void xonar_hdav_slim_cleanup(struct oxygen *chip) +{ + xonar_hdmi_cleanup(chip); + xonar_disable_output(chip); + wm8776_write(chip, WM8776_RESET, 0); + msleep(2); +} + static void xonar_ds_suspend(struct oxygen *chip) { xonar_ds_cleanup(chip); } +static void xonar_hdav_slim_suspend(struct oxygen *chip) +{ + xonar_hdav_slim_cleanup(chip); +} + static void xonar_ds_resume(struct oxygen *chip) { wm8776_registers_init(chip); @@ -264,6 +344,15 @@ static void xonar_ds_resume(struct oxygen *chip) xonar_ds_handle_hp_jack(chip); } +static void xonar_hdav_slim_resume(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + wm8776_registers_init(chip); + xonar_hdmi_resume(chip, &data->hdmi); + xonar_enable_output(chip); +} + static void wm8776_adc_hardware_filter(unsigned int channel, struct snd_pcm_hardware *hardware) { @@ -278,6 +367,13 @@ static void wm8776_adc_hardware_filter(unsigned int channel, } } +static void xonar_hdav_slim_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware) +{ + wm8776_adc_hardware_filter(channel, hardware); + xonar_hdmi_pcm_hardware_filter(channel, hardware); +} + static void set_wm87x6_dac_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -294,6 +390,14 @@ static void set_wm8776_adc_params(struct oxygen *chip, wm8776_write_cached(chip, WM8776_MSTRCTRL, reg); } +static void set_hdav_slim_dac_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct xonar_wm87x6 *data = chip->model_data; + + xonar_set_hdmi_params(chip, &data->hdmi, params); +} + static void update_wm8776_volume(struct oxygen *chip) { struct xonar_wm87x6 *data = chip->model_data; @@ -473,11 +577,6 @@ static int wm8776_field_enum_info(struct snd_kcontrol *ctl, const char *const *names; max = (ctl->private_value >> 12) & 0xf; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = max + 1; - if (info->value.enumerated.item > max) - info->value.enumerated.item = max; switch ((ctl->private_value >> 24) & 0x1f) { case WM8776_ALCCTRL2: names = hld; @@ -501,8 +600,7 @@ static int wm8776_field_enum_info(struct snd_kcontrol *ctl, default: return -ENXIO; } - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, max + 1, names); } static int wm8776_field_volume_info(struct snd_kcontrol *ctl, @@ -759,13 +857,8 @@ static int wm8776_level_control_info(struct snd_kcontrol *ctl, static const char *const names[3] = { "None", "Peak Limiter", "Automatic Level Control" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item >= 3) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + + return snd_ctl_enum_info(info, 1, 3, names); } static int wm8776_level_control_get(struct snd_kcontrol *ctl, @@ -851,13 +944,7 @@ static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) "None", "High-pass Filter" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item >= 2) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) @@ -985,6 +1072,53 @@ static const struct snd_kcontrol_new ds_controls[] = { .private_value = 0, }, }; +static const struct snd_kcontrol_new hdav_slim_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "HDMI Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = xonar_gpio_bit_switch_get, + .put = xonar_gpio_bit_switch_put, + .private_value = GPIO_SLIM_HDMI_DISABLE | XONAR_GPIO_BIT_INVERT, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .info = wm8776_hp_vol_info, + .get = wm8776_hp_vol_get, + .put = wm8776_hp_vol_put, + .tlv = { .p = wm8776_hp_db_scale }, + }, + WM8776_BIT_SWITCH("Headphone Playback Switch", + WM8776_PWRDOWN, WM8776_HPPD, 1, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Capture Volume", + .info = wm8776_input_vol_info, + .get = wm8776_input_vol_get, + .put = wm8776_input_vol_put, + .tlv = { .p = wm8776_adc_db_scale }, + }, + WM8776_BIT_SWITCH("Mic Capture Switch", + WM8776_ADCMUX, 1 << 0, 0, 0), + WM8776_BIT_SWITCH("Aux Capture Switch", + WM8776_ADCMUX, 1 << 1, 0, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC Filter Capture Enum", + .info = hpf_info, + .get = hpf_get, + .put = hpf_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Level Control Capture Enum", + .info = wm8776_level_control_info, + .get = wm8776_level_control_get, + .put = wm8776_level_control_put, + .private_value = 0, + }, +}; static const struct snd_kcontrol_new lc_controls[] = { WM8776_FIELD_CTL_VOLUME("Limiter Threshold", WM8776_ALCCTRL1, 0, 11, 0, 15, 0xf, @@ -1028,6 +1162,26 @@ static const struct snd_kcontrol_new lc_controls[] = { LC_CONTROL_ALC, wm8776_ngth_db_scale), }; +static int add_lc_controls(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + struct snd_kcontrol *ctl; + int err; + + BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); + for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { + ctl = snd_ctl_new1(&lc_controls[i], chip); + if (!ctl) + return -ENOMEM; + err = snd_ctl_add(chip->card, ctl); + if (err < 0) + return err; + data->lc_controls[i] = ctl; + } + return 0; +} + static int xonar_ds_mixer_init(struct oxygen *chip) { struct xonar_wm87x6 *data = chip->model_data; @@ -1049,17 +1203,54 @@ static int xonar_ds_mixer_init(struct oxygen *chip) } if (!data->line_adcmux_control || !data->mic_adcmux_control) return -ENXIO; - BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); - for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { - ctl = snd_ctl_new1(&lc_controls[i], chip); + + return add_lc_controls(chip); +} + +static int xonar_hdav_slim_mixer_init(struct oxygen *chip) +{ + unsigned int i; + struct snd_kcontrol *ctl; + int err; + + for (i = 0; i < ARRAY_SIZE(hdav_slim_controls); ++i) { + ctl = snd_ctl_new1(&hdav_slim_controls[i], chip); if (!ctl) return -ENOMEM; err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; - data->lc_controls[i] = ctl; } - return 0; + + return add_lc_controls(chip); +} + +static void dump_wm8776_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + + snd_iprintf(buffer, "\nWM8776:\n00:"); + for (i = 0; i < 0x10; ++i) + snd_iprintf(buffer, " %03x", data->wm8776_regs[i]); + snd_iprintf(buffer, "\n10:"); + for (i = 0x10; i < 0x17; ++i) + snd_iprintf(buffer, " %03x", data->wm8776_regs[i]); + snd_iprintf(buffer, "\n"); +} + +static void dump_wm87x6_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + + dump_wm8776_registers(chip, buffer); + snd_iprintf(buffer, "\nWM8766:\n00:"); + for (i = 0; i < 0x10; ++i) + snd_iprintf(buffer, " %03x", data->wm8766_regs[i]); + snd_iprintf(buffer, "\n"); } static const struct oxygen_model model_xonar_ds = { @@ -1072,22 +1263,57 @@ static const struct oxygen_model model_xonar_ds = { .suspend = xonar_ds_suspend, .resume = xonar_ds_resume, .pcm_hardware_filter = wm8776_adc_hardware_filter, - .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_wm87x6_dac_params, .set_adc_params = set_wm8776_adc_params, .update_dac_volume = update_wm87x6_volume, .update_dac_mute = update_wm87x6_mute, .update_center_lfe_mix = update_wm8766_center_lfe_mix, .gpio_changed = xonar_ds_gpio_changed, + .dump_registers = dump_wm87x6_registers, .dac_tlv = wm87x6_dac_db_scale, .model_data_size = sizeof(struct xonar_wm87x6), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_1, - .dac_channels = 8, + .dac_channels_pcm = 8, + .dac_channels_mixer = 8, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, .function_flags = OXYGEN_FUNCTION_SPI, + .dac_mclks = OXYGEN_MCLKS(256, 256, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +static const struct oxygen_model model_xonar_hdav_slim = { + .shortname = "Xonar HDAV1.3 Slim", + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_hdav_slim_init, + .mixer_init = xonar_hdav_slim_mixer_init, + .cleanup = xonar_hdav_slim_cleanup, + .suspend = xonar_hdav_slim_suspend, + .resume = xonar_hdav_slim_resume, + .pcm_hardware_filter = xonar_hdav_slim_hardware_filter, + .set_dac_params = set_hdav_slim_dac_params, + .set_adc_params = set_wm8776_adc_params, + .update_dac_volume = update_wm8776_volume, + .update_dac_mute = update_wm8776_mute, + .uart_input = xonar_hdmi_uart_input, + .dump_registers = dump_wm8776_registers, + .dac_tlv = wm87x6_dac_db_scale, + .model_data_size = sizeof(struct xonar_wm87x6), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_1, + .dac_channels_pcm = 8, + .dac_channels_mixer = 2, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_mclks = OXYGEN_MCLKS(256, 256, 128), + .adc_mclks = OXYGEN_MCLKS(256, 256, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -1099,6 +1325,9 @@ int __devinit get_xonar_wm87x6_model(struct oxygen *chip, case 0x838e: chip->model = model_xonar_ds; break; + case 0x835e: + chip->model = model_xonar_hdav_slim; + break; default: return -EINVAL; } diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 0b720cf7783e..2d8332416c83 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -60,6 +60,7 @@ MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP}," "{RME HDSP-9652}," "{RME HDSP-9632}}"); #ifdef HDSP_FW_LOADER +MODULE_FIRMWARE("rpm_firmware.bin"); MODULE_FIRMWARE("multiface_firmware.bin"); MODULE_FIRMWARE("multiface_firmware_rev11.bin"); MODULE_FIRMWARE("digiface_firmware.bin"); @@ -81,6 +82,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define H9632_SS_CHANNELS 12 #define H9632_DS_CHANNELS 8 #define H9632_QS_CHANNELS 4 +#define RPM_CHANNELS 6 /* Write registers. These are defined as byte-offsets from the iobase value. */ @@ -191,6 +193,25 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_PhoneGain1 (1<<30) #define HDSP_QuadSpeed (1<<31) +/* RPM uses some of the registers for special purposes */ +#define HDSP_RPM_Inp12 0x04A00 +#define HDSP_RPM_Inp12_Phon_6dB 0x00800 /* Dolby */ +#define HDSP_RPM_Inp12_Phon_0dB 0x00000 /* .. */ +#define HDSP_RPM_Inp12_Phon_n6dB 0x04000 /* inp_0 */ +#define HDSP_RPM_Inp12_Line_0dB 0x04200 /* Dolby+PRO */ +#define HDSP_RPM_Inp12_Line_n6dB 0x00200 /* PRO */ + +#define HDSP_RPM_Inp34 0x32000 +#define HDSP_RPM_Inp34_Phon_6dB 0x20000 /* SyncRef1 */ +#define HDSP_RPM_Inp34_Phon_0dB 0x00000 /* .. */ +#define HDSP_RPM_Inp34_Phon_n6dB 0x02000 /* SyncRef2 */ +#define HDSP_RPM_Inp34_Line_0dB 0x30000 /* SyncRef1+SyncRef0 */ +#define HDSP_RPM_Inp34_Line_n6dB 0x10000 /* SyncRef0 */ + +#define HDSP_RPM_Bypass 0x01000 + +#define HDSP_RPM_Disconnect 0x00001 + #define HDSP_ADGainMask (HDSP_ADGain0|HDSP_ADGain1) #define HDSP_ADGainMinus10dBV HDSP_ADGainMask #define HDSP_ADGainPlus4dBu (HDSP_ADGain0) @@ -450,7 +471,7 @@ struct hdsp { u32 creg_spdif; u32 creg_spdif_stream; int clock_source_locked; - char *card_name; /* digiface/multiface */ + char *card_name; /* digiface/multiface/rpm */ enum HDSP_IO_Type io_type; /* ditto, but for code use */ unsigned short firmware_rev; unsigned short state; /* stores state bits */ @@ -612,6 +633,7 @@ static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out) switch (hdsp->io_type) { case Multiface: case Digiface: + case RPM: default: if (hdsp->firmware_rev == 0xa) return (64 * out) + (32 + (in)); @@ -629,6 +651,7 @@ static int hdsp_input_to_output_key (struct hdsp *hdsp, int in, int out) switch (hdsp->io_type) { case Multiface: case Digiface: + case RPM: default: if (hdsp->firmware_rev == 0xa) return (64 * out) + in; @@ -655,7 +678,7 @@ static int hdsp_check_for_iobox (struct hdsp *hdsp) { if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { - snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n"); + snd_printk("Hammerfall-DSP: no IO box connected!\n"); hdsp->state &= ~HDSP_FirmwareLoaded; return -EIO; } @@ -680,7 +703,7 @@ static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops, } } - snd_printk("Hammerfall-DSP: no Digiface or Multiface connected!\n"); + snd_printk("Hammerfall-DSP: no IO box connected!\n"); hdsp->state &= ~HDSP_FirmwareLoaded; return -EIO; } @@ -752,17 +775,21 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); hdsp_write (hdsp, HDSP_fifoData, 0); - if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT)) { - hdsp->io_type = Multiface; - hdsp_write (hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); - hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); - hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) { + hdsp_write(hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); + hdsp_write(hdsp, HDSP_control2Reg, HDSP_S_LOAD); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) + hdsp->io_type = RPM; + else + hdsp->io_type = Multiface; } else { hdsp->io_type = Digiface; } } else { /* firmware was already loaded, get iobox type */ - if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) + if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version2) + hdsp->io_type = RPM; + else if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; else hdsp->io_type = Digiface; @@ -1184,6 +1211,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) hdsp->channel_map = channel_map_ds; } else { switch (hdsp->io_type) { + case RPM: case Multiface: hdsp->channel_map = channel_map_mf_ss; break; @@ -3231,6 +3259,318 @@ HDSP_PRECISE_POINTER("Precise Pointer", 0), HDSP_USE_MIDI_TASKLET("Use Midi Tasklet", 0), }; + +static int hdsp_rpm_input12(struct hdsp *hdsp) +{ + switch (hdsp->control_register & HDSP_RPM_Inp12) { + case HDSP_RPM_Inp12_Phon_6dB: + return 0; + case HDSP_RPM_Inp12_Phon_n6dB: + return 2; + case HDSP_RPM_Inp12_Line_0dB: + return 3; + case HDSP_RPM_Inp12_Line_n6dB: + return 4; + } + return 1; +} + + +static int snd_hdsp_get_rpm_input12(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = hdsp_rpm_input12(hdsp); + return 0; +} + + +static int hdsp_set_rpm_input12(struct hdsp *hdsp, int mode) +{ + hdsp->control_register &= ~HDSP_RPM_Inp12; + switch (mode) { + case 0: + hdsp->control_register |= HDSP_RPM_Inp12_Phon_6dB; + break; + case 1: + break; + case 2: + hdsp->control_register |= HDSP_RPM_Inp12_Phon_n6dB; + break; + case 3: + hdsp->control_register |= HDSP_RPM_Inp12_Line_0dB; + break; + case 4: + hdsp->control_register |= HDSP_RPM_Inp12_Line_n6dB; + break; + default: + return -1; + } + + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); + return 0; +} + + +static int snd_hdsp_put_rpm_input12(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + int change; + int val; + + if (!snd_hdsp_use_is_exclusive(hdsp)) + return -EBUSY; + val = ucontrol->value.enumerated.item[0]; + if (val < 0) + val = 0; + if (val > 4) + val = 4; + spin_lock_irq(&hdsp->lock); + if (val != hdsp_rpm_input12(hdsp)) + change = (hdsp_set_rpm_input12(hdsp, val) == 0) ? 1 : 0; + else + change = 0; + spin_unlock_irq(&hdsp->lock); + return change; +} + + +static int snd_hdsp_info_rpm_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = {"Phono +6dB", "Phono 0dB", "Phono -6dB", "Line 0dB", "Line -6dB"}; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 5; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + + +static int hdsp_rpm_input34(struct hdsp *hdsp) +{ + switch (hdsp->control_register & HDSP_RPM_Inp34) { + case HDSP_RPM_Inp34_Phon_6dB: + return 0; + case HDSP_RPM_Inp34_Phon_n6dB: + return 2; + case HDSP_RPM_Inp34_Line_0dB: + return 3; + case HDSP_RPM_Inp34_Line_n6dB: + return 4; + } + return 1; +} + + +static int snd_hdsp_get_rpm_input34(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = hdsp_rpm_input34(hdsp); + return 0; +} + + +static int hdsp_set_rpm_input34(struct hdsp *hdsp, int mode) +{ + hdsp->control_register &= ~HDSP_RPM_Inp34; + switch (mode) { + case 0: + hdsp->control_register |= HDSP_RPM_Inp34_Phon_6dB; + break; + case 1: + break; + case 2: + hdsp->control_register |= HDSP_RPM_Inp34_Phon_n6dB; + break; + case 3: + hdsp->control_register |= HDSP_RPM_Inp34_Line_0dB; + break; + case 4: + hdsp->control_register |= HDSP_RPM_Inp34_Line_n6dB; + break; + default: + return -1; + } + + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); + return 0; +} + + +static int snd_hdsp_put_rpm_input34(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + int change; + int val; + + if (!snd_hdsp_use_is_exclusive(hdsp)) + return -EBUSY; + val = ucontrol->value.enumerated.item[0]; + if (val < 0) + val = 0; + if (val > 4) + val = 4; + spin_lock_irq(&hdsp->lock); + if (val != hdsp_rpm_input34(hdsp)) + change = (hdsp_set_rpm_input34(hdsp, val) == 0) ? 1 : 0; + else + change = 0; + spin_unlock_irq(&hdsp->lock); + return change; +} + + +/* RPM Bypass switch */ +static int hdsp_rpm_bypass(struct hdsp *hdsp) +{ + return (hdsp->control_register & HDSP_RPM_Bypass) ? 1 : 0; +} + + +static int snd_hdsp_get_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = hdsp_rpm_bypass(hdsp); + return 0; +} + + +static int hdsp_set_rpm_bypass(struct hdsp *hdsp, int on) +{ + if (on) + hdsp->control_register |= HDSP_RPM_Bypass; + else + hdsp->control_register &= ~HDSP_RPM_Bypass; + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); + return 0; +} + + +static int snd_hdsp_put_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + int change; + unsigned int val; + + if (!snd_hdsp_use_is_exclusive(hdsp)) + return -EBUSY; + val = ucontrol->value.integer.value[0] & 1; + spin_lock_irq(&hdsp->lock); + change = (int)val != hdsp_rpm_bypass(hdsp); + hdsp_set_rpm_bypass(hdsp, val); + spin_unlock_irq(&hdsp->lock); + return change; +} + + +static int snd_hdsp_info_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = {"On", "Off"}; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + + +/* RPM Disconnect switch */ +static int hdsp_rpm_disconnect(struct hdsp *hdsp) +{ + return (hdsp->control_register & HDSP_RPM_Disconnect) ? 1 : 0; +} + + +static int snd_hdsp_get_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = hdsp_rpm_disconnect(hdsp); + return 0; +} + + +static int hdsp_set_rpm_disconnect(struct hdsp *hdsp, int on) +{ + if (on) + hdsp->control_register |= HDSP_RPM_Disconnect; + else + hdsp->control_register &= ~HDSP_RPM_Disconnect; + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); + return 0; +} + + +static int snd_hdsp_put_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); + int change; + unsigned int val; + + if (!snd_hdsp_use_is_exclusive(hdsp)) + return -EBUSY; + val = ucontrol->value.integer.value[0] & 1; + spin_lock_irq(&hdsp->lock); + change = (int)val != hdsp_rpm_disconnect(hdsp); + hdsp_set_rpm_disconnect(hdsp, val); + spin_unlock_irq(&hdsp->lock); + return change; +} + +static int snd_hdsp_info_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = {"On", "Off"}; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static struct snd_kcontrol_new snd_hdsp_rpm_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "RPM Bypass", + .get = snd_hdsp_get_rpm_bypass, + .put = snd_hdsp_put_rpm_bypass, + .info = snd_hdsp_info_rpm_bypass + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "RPM Disconnect", + .get = snd_hdsp_get_rpm_disconnect, + .put = snd_hdsp_put_rpm_disconnect, + .info = snd_hdsp_info_rpm_disconnect + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 1/2", + .get = snd_hdsp_get_rpm_input12, + .put = snd_hdsp_put_rpm_input12, + .info = snd_hdsp_info_rpm_input + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 3/4", + .get = snd_hdsp_get_rpm_input34, + .put = snd_hdsp_put_rpm_input34, + .info = snd_hdsp_info_rpm_input + }, + HDSP_SYSTEM_SAMPLE_RATE("System Sample Rate", 0), + HDSP_MIXER("Mixer", 0) +}; + static struct snd_kcontrol_new snd_hdsp_96xx_aeb = HDSP_AEB("Analog Extension Board", 0); static struct snd_kcontrol_new snd_hdsp_adat_sync_check = HDSP_ADAT_SYNC_CHECK; @@ -3240,6 +3580,16 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) int err; struct snd_kcontrol *kctl; + if (hdsp->io_type == RPM) { + /* RPM Bypass, Disconnect and Input switches */ + for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_rpm_controls); idx++) { + err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_rpm_controls[idx], hdsp)); + if (err < 0) + return err; + } + return 0; + } + for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_controls); idx++) { if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_controls[idx], hdsp))) < 0) return err; @@ -3459,48 +3809,102 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "\n"); - switch (hdsp_spdif_in(hdsp)) { - case HDSP_SPDIFIN_OPTICAL: - snd_iprintf(buffer, "IEC958 input: Optical\n"); - break; - case HDSP_SPDIFIN_COAXIAL: - snd_iprintf(buffer, "IEC958 input: Coaxial\n"); - break; - case HDSP_SPDIFIN_INTERNAL: - snd_iprintf(buffer, "IEC958 input: Internal\n"); - break; - case HDSP_SPDIFIN_AES: - snd_iprintf(buffer, "IEC958 input: AES\n"); - break; - default: - snd_iprintf(buffer, "IEC958 input: ???\n"); - break; + if (hdsp->io_type != RPM) { + switch (hdsp_spdif_in(hdsp)) { + case HDSP_SPDIFIN_OPTICAL: + snd_iprintf(buffer, "IEC958 input: Optical\n"); + break; + case HDSP_SPDIFIN_COAXIAL: + snd_iprintf(buffer, "IEC958 input: Coaxial\n"); + break; + case HDSP_SPDIFIN_INTERNAL: + snd_iprintf(buffer, "IEC958 input: Internal\n"); + break; + case HDSP_SPDIFIN_AES: + snd_iprintf(buffer, "IEC958 input: AES\n"); + break; + default: + snd_iprintf(buffer, "IEC958 input: ???\n"); + break; + } } - if (hdsp->control_register & HDSP_SPDIFOpticalOut) - snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); - else - snd_iprintf(buffer, "IEC958 output: Coaxial only\n"); + if (RPM == hdsp->io_type) { + if (hdsp->control_register & HDSP_RPM_Bypass) + snd_iprintf(buffer, "RPM Bypass: disabled\n"); + else + snd_iprintf(buffer, "RPM Bypass: enabled\n"); + if (hdsp->control_register & HDSP_RPM_Disconnect) + snd_iprintf(buffer, "RPM disconnected\n"); + else + snd_iprintf(buffer, "RPM connected\n"); - if (hdsp->control_register & HDSP_SPDIFProfessional) - snd_iprintf(buffer, "IEC958 quality: Professional\n"); - else - snd_iprintf(buffer, "IEC958 quality: Consumer\n"); + switch (hdsp->control_register & HDSP_RPM_Inp12) { + case HDSP_RPM_Inp12_Phon_6dB: + snd_iprintf(buffer, "Input 1/2: Phono, 6dB\n"); + break; + case HDSP_RPM_Inp12_Phon_0dB: + snd_iprintf(buffer, "Input 1/2: Phono, 0dB\n"); + break; + case HDSP_RPM_Inp12_Phon_n6dB: + snd_iprintf(buffer, "Input 1/2: Phono, -6dB\n"); + break; + case HDSP_RPM_Inp12_Line_0dB: + snd_iprintf(buffer, "Input 1/2: Line, 0dB\n"); + break; + case HDSP_RPM_Inp12_Line_n6dB: + snd_iprintf(buffer, "Input 1/2: Line, -6dB\n"); + break; + default: + snd_iprintf(buffer, "Input 1/2: ???\n"); + } - if (hdsp->control_register & HDSP_SPDIFEmphasis) - snd_iprintf(buffer, "IEC958 emphasis: on\n"); - else - snd_iprintf(buffer, "IEC958 emphasis: off\n"); + switch (hdsp->control_register & HDSP_RPM_Inp34) { + case HDSP_RPM_Inp34_Phon_6dB: + snd_iprintf(buffer, "Input 3/4: Phono, 6dB\n"); + break; + case HDSP_RPM_Inp34_Phon_0dB: + snd_iprintf(buffer, "Input 3/4: Phono, 0dB\n"); + break; + case HDSP_RPM_Inp34_Phon_n6dB: + snd_iprintf(buffer, "Input 3/4: Phono, -6dB\n"); + break; + case HDSP_RPM_Inp34_Line_0dB: + snd_iprintf(buffer, "Input 3/4: Line, 0dB\n"); + break; + case HDSP_RPM_Inp34_Line_n6dB: + snd_iprintf(buffer, "Input 3/4: Line, -6dB\n"); + break; + default: + snd_iprintf(buffer, "Input 3/4: ???\n"); + } - if (hdsp->control_register & HDSP_SPDIFNonAudio) - snd_iprintf(buffer, "IEC958 NonAudio: on\n"); - else - snd_iprintf(buffer, "IEC958 NonAudio: off\n"); - if ((x = hdsp_spdif_sample_rate (hdsp)) != 0) - snd_iprintf (buffer, "IEC958 sample rate: %d\n", x); - else - snd_iprintf (buffer, "IEC958 sample rate: Error flag set\n"); + } else { + if (hdsp->control_register & HDSP_SPDIFOpticalOut) + snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); + else + snd_iprintf(buffer, "IEC958 output: Coaxial only\n"); + + if (hdsp->control_register & HDSP_SPDIFProfessional) + snd_iprintf(buffer, "IEC958 quality: Professional\n"); + else + snd_iprintf(buffer, "IEC958 quality: Consumer\n"); + + if (hdsp->control_register & HDSP_SPDIFEmphasis) + snd_iprintf(buffer, "IEC958 emphasis: on\n"); + else + snd_iprintf(buffer, "IEC958 emphasis: off\n"); + if (hdsp->control_register & HDSP_SPDIFNonAudio) + snd_iprintf(buffer, "IEC958 NonAudio: on\n"); + else + snd_iprintf(buffer, "IEC958 NonAudio: off\n"); + x = hdsp_spdif_sample_rate(hdsp); + if (x != 0) + snd_iprintf(buffer, "IEC958 sample rate: %d\n", x); + else + snd_iprintf(buffer, "IEC958 sample rate: Error flag set\n"); + } snd_iprintf(buffer, "\n"); /* Sync Check */ @@ -3765,7 +4169,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) snd_hdsp_midi_input_read (&hdsp->midi[0]); } } - if (hdsp->io_type != Multiface && hdsp->io_type != H9632 && midi1 && midi1status) { + if (hdsp->io_type != Multiface && hdsp->io_type != RPM && hdsp->io_type != H9632 && midi1 && midi1status) { if (hdsp->use_midi_tasklet) { /* we disable interrupts for this input until processing is done */ hdsp->control_register &= ~HDSP_Midi1InterruptEnable; @@ -4093,7 +4497,7 @@ static struct snd_pcm_hardware snd_hdsp_playback_subinfo = SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, - .channels_min = 14, + .channels_min = 6, .channels_max = HDSP_MAX_CHANNELS, .buffer_bytes_max = HDSP_CHANNEL_BUFFER_BYTES * HDSP_MAX_CHANNELS, .period_bytes_min = (64 * 4) * 10, @@ -4122,7 +4526,7 @@ static struct snd_pcm_hardware snd_hdsp_capture_subinfo = SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, - .channels_min = 14, + .channels_min = 5, .channels_max = HDSP_MAX_CHANNELS, .buffer_bytes_max = HDSP_CHANNEL_BUFFER_BYTES * HDSP_MAX_CHANNELS, .period_bytes_min = (64 * 4) * 10, @@ -4357,10 +4761,12 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream) snd_hdsp_hw_rule_rate_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - hdsp->creg_spdif_stream = hdsp->creg_spdif; - hdsp->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; - snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE | - SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id); + if (RPM != hdsp->io_type) { + hdsp->creg_spdif_stream = hdsp->creg_spdif; + hdsp->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id); + } return 0; } @@ -4375,9 +4781,11 @@ static int snd_hdsp_playback_release(struct snd_pcm_substream *substream) spin_unlock_irq(&hdsp->lock); - hdsp->spdif_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; - snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE | - SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id); + if (RPM != hdsp->io_type) { + hdsp->spdif_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id); + } return 0; } @@ -4616,7 +5024,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (hdsp->io_type != H9632) info.adatsync_sync_check = (unsigned char)hdsp_adatsync_sync_check(hdsp); info.spdif_sync_check = (unsigned char)hdsp_spdif_sync_check(hdsp); - for (i = 0; i < ((hdsp->io_type != Multiface && hdsp->io_type != H9632) ? 3 : 1); ++i) + for (i = 0; i < ((hdsp->io_type != Multiface && hdsp->io_type != RPM && hdsp->io_type != H9632) ? 3 : 1); ++i) info.adat_sync_check[i] = (unsigned char)hdsp_adat_sync_check(hdsp, i); info.spdif_in = (unsigned char)hdsp_spdif_in(hdsp); info.spdif_out = (unsigned char)hdsp_spdif_out(hdsp); @@ -4636,6 +5044,9 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp); info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp); + } else if (hdsp->io_type == RPM) { + info.da_gain = (unsigned char) hdsp_rpm_input12(hdsp); + info.ad_gain = (unsigned char) hdsp_rpm_input34(hdsp); } if (hdsp->io_type == H9632 || hdsp->io_type == H9652) info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp); @@ -4844,6 +5255,14 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS; break; + case RPM: + hdsp->card_name = "RME Hammerfall DSP + RPM"; + hdsp->ss_in_channels = RPM_CHANNELS-1; + hdsp->ss_out_channels = RPM_CHANNELS; + hdsp->ds_in_channels = RPM_CHANNELS-1; + hdsp->ds_out_channels = RPM_CHANNELS; + break; + default: /* should never get here */ break; @@ -4930,6 +5349,9 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) /* caution: max length of firmware filename is 30! */ switch (hdsp->io_type) { + case RPM: + fwfile = "rpm_firmware.bin"; + break; case Multiface: if (hdsp->firmware_rev == 0xa) fwfile = "multiface_firmware.bin"; @@ -5100,7 +5522,9 @@ static int __devinit snd_hdsp_create(struct snd_card *card, return 0; } else { snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); - if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) + if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version2) + hdsp->io_type = RPM; + else if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; else hdsp->io_type = Digiface; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 0c98ef9156d8..f5eadfc0672a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -487,7 +487,7 @@ struct hdspm { struct snd_kcontrol *playback_mixer_ctls[HDSPM_MAX_CHANNELS]; /* but input to much, so not used */ struct snd_kcontrol *input_mixer_ctls[HDSPM_MAX_CHANNELS]; - /* full mixer accessable over mixer ioctl or hwdep-device */ + /* full mixer accessible over mixer ioctl or hwdep-device */ struct hdspm_mixer *mixer; }; @@ -550,7 +550,7 @@ static inline int HDSPM_bit2freq(int n) return bit2freq_tab[n]; } -/* Write/read to/from HDSPM with Adresses in Bytes +/* Write/read to/from HDSPM with Addresses in Bytes not words but only 32Bit writes are allowed */ static inline void hdspm_write(struct hdspm * hdspm, unsigned int reg, @@ -2908,7 +2908,7 @@ static int snd_hdspm_create_controls(struct snd_card *card, struct hdspm * hdspm /* Channel playback mixer as default control Note: the whole matrix would be 128*HDSPM_MIXER_CHANNELS Faders, - thats too * big for any alsamixer they are accesible via special + thats too * big for any alsamixer they are accessible via special IOCTL on hwdep and the mixer 2dimensional mixer control */ diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 5518371db13f..c94c051ad0c8 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1389,15 +1389,9 @@ static struct snd_kcontrol_new snd_ymfpci_spdif_stream __devinitdata = static int snd_ymfpci_drec_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *info) { - static char *texts[3] = {"AC'97", "IEC958", "ZV Port"}; - - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item > 2) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, texts[info->value.enumerated.item]); - return 0; + static const char *const texts[3] = {"AC'97", "IEC958", "ZV Port"}; + + return snd_ctl_enum_info(info, 1, 3, texts); } static int snd_ymfpci_drec_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *value) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 581a670e8261..edce8a27e3ee 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -51,7 +51,7 @@ static struct snd_ps3_card_info the_card; static int snd_ps3_start_delay = CONFIG_SND_PS3_DEFAULT_START_DELAY; module_param_named(start_delay, snd_ps3_start_delay, uint, 0644); -MODULE_PARM_DESC(start_delay, "time to insert silent data in milisec"); +MODULE_PARM_DESC(start_delay, "time to insert silent data in ms"); static int index = SNDRV_DEFAULT_IDX1; static char *id = SNDRV_DEFAULT_STR1; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index b6ecc7e89673..bd0517cb7980 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1958,7 +1958,7 @@ static int max98088_probe(struct snd_soc_codec *codec) return ret; } - /* initalize private data */ + /* initialize private data */ max98088->sysclk = (unsigned)-1; max98088->eq_textcnt = 0; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b439eee462cb..8ad93ee2e92b 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -20,6 +20,7 @@ #include <linux/platform_device.h> #include <linux/clk.h> #include <linux/io.h> +#include <linux/pxa2xx_ssp.h> #include <asm/irq.h> @@ -33,7 +34,6 @@ #include <mach/hardware.h> #include <mach/dma.h> #include <mach/audio.h> -#include <plat/ssp.h> #include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index d42fe8df7144..e8ac961c6ba1 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -56,7 +56,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev) goto out3; } - /* Set audio clock heirachy for S/PDIF */ + /* Set audio clock hierarchy for S/PDIF */ clk_set_parent(mout_epll, fout_epll); clk_set_parent(sclk_audio0, mout_epll); clk_set_parent(sclk_spdif, sclk_audio0); @@ -74,7 +74,7 @@ out1: /* We should haved to set clock directly on this part because of clock * scheme of Samsudng SoCs did not support to set rates from abstrct - * clock of it's heirachy. + * clock of it's hierarchy. */ static int set_audio_clock_rate(unsigned long epll_rate, unsigned long audio_rate) @@ -190,7 +190,7 @@ static int __init smdk_init(void) if (ret) goto err3; - /* Set audio clock heirachy manually */ + /* Set audio clock hierarchy manually */ ret = set_audio_clock_heirachy(smdk_snd_spdif_device); if (ret) goto err4; diff --git a/sound/usb/format.c b/sound/usb/format.c index 69148212aa70..5b792d2c8061 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -76,7 +76,10 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format = 1 << UAC_FORMAT_TYPE_I_PCM; } if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) { - if (sample_width > sample_bytes * 8) { + if (chip->usb_id == USB_ID(0x0582, 0x0016) /* Edirol SD-90 */ && + sample_width == 24 && sample_bytes == 2) + sample_bytes = 3; + else if (sample_width > sample_bytes * 8) { snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", chip->dev->devnum, fp->iface, fp->altsetting, sample_width, sample_bytes); diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 25bce7e5b1af..db2dc5ffe6dd 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -850,8 +850,8 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, return; } - memset(urb->transfer_buffer + count, 0xFD, 9 - count); - urb->transfer_buffer_length = count; + memset(urb->transfer_buffer + count, 0xFD, ep->max_transfer - count); + urb->transfer_buffer_length = ep->max_transfer; } static struct usb_protocol_ops snd_usbmidi_122l_ops = { @@ -1295,6 +1295,13 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */ ep->max_transfer = 4; break; + /* + * Some devices only work with 9 bytes packet size: + */ + case USB_ID(0x0644, 0x800E): /* Tascam US-122L */ + case USB_ID(0x0644, 0x800F): /* Tascam US-144 */ + ep->max_transfer = 9; + break; } for (i = 0; i < OUTPUT_URBS; ++i) { buffer = usb_alloc_coherent(umidi->dev, @@ -1729,13 +1736,7 @@ static int roland_load_info(struct snd_kcontrol *kcontrol, { static const char *const names[] = { "High Load", "Light Load" }; - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 2; - if (info->value.enumerated.item > 1) - info->value.enumerated.item = 1; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(info, 1, 2, names); } static int roland_load_get(struct snd_kcontrol *kcontrol, diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f2d74d654b3c..7df89b3d7ded 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1633,18 +1633,11 @@ static int parse_audio_extension_unit(struct mixer_build *state, int unitid, voi static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *cval = kcontrol->private_data; - char **itemlist = (char **)kcontrol->private_value; + const char **itemlist = (const char **)kcontrol->private_value; if (snd_BUG_ON(!itemlist)) return -EINVAL; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = cval->max; - if (uinfo->value.enumerated.item >= cval->max) - uinfo->value.enumerated.item = cval->max - 1; - strlcpy(uinfo->value.enumerated.name, itemlist[uinfo->value.enumerated.item], - sizeof(uinfo->value.enumerated.name)); - return 0; + return snd_ctl_enum_info(uinfo, 1, cval->max, itemlist); } /* get callback for selector unit */ diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index ad7079d1676c..35999874d301 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -705,11 +705,11 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE + .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE + .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 6ef68e42138e..084e6fc8d5bf 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -273,29 +273,26 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, struct file *file, poll_table *wait) { struct us122l *us122l = hw->private_data; - struct usb_stream *s = us122l->sk.s; unsigned *polled; unsigned int mask; poll_wait(file, &us122l->sk.sleep, wait); - switch (s->state) { - case usb_stream_ready: - if (us122l->first == file) - polled = &s->periods_polled; - else - polled = &us122l->second_periods_polled; - if (*polled != s->periods_done) { - *polled = s->periods_done; - mask = POLLIN | POLLOUT | POLLWRNORM; - break; + mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; + if (mutex_trylock(&us122l->mutex)) { + struct usb_stream *s = us122l->sk.s; + if (s && s->state == usb_stream_ready) { + if (us122l->first == file) + polled = &s->periods_polled; + else + polled = &us122l->second_periods_polled; + if (*polled != s->periods_done) { + *polled = s->periods_done; + mask = POLLIN | POLLOUT | POLLWRNORM; + } else + mask = 0; } - /* Fall through */ - mask = 0; - break; - default: - mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; - break; + mutex_unlock(&us122l->mutex); } return mask; } @@ -381,6 +378,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, { struct usb_stream_config *cfg; struct us122l *us122l = hw->private_data; + struct usb_stream *s; unsigned min_period_frames; int err = 0; bool high_speed; @@ -426,18 +424,18 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, snd_power_wait(hw->card, SNDRV_CTL_POWER_D0); mutex_lock(&us122l->mutex); + s = us122l->sk.s; if (!us122l->master) us122l->master = file; else if (us122l->master != file) { - if (memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg))) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg))) { err = -EIO; goto unlock; } us122l->slave = file; } - if (!us122l->sk.s || - memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg)) || - us122l->sk.s->state == usb_stream_xrun) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg)) || + s->state == usb_stream_xrun) { us122l_stop(us122l); if (!us122l_start(us122l, cfg->sample_rate, cfg->period_frames)) err = -EIO; @@ -448,6 +446,7 @@ unlock: mutex_unlock(&us122l->mutex); free: kfree(cfg); + wake_up_all(&us122l->sk.sleep); return err; } |