diff options
| author | Steven Whitehouse <swhiteho@redhat.com> | 2006-09-25 12:26:59 -0400 |
|---|---|---|
| committer | Steven Whitehouse <swhiteho@redhat.com> | 2006-09-25 12:26:59 -0400 |
| commit | 363e065c02b1273364d5356711a83e7f548fc0c8 (patch) | |
| tree | 0df0e65da403ade33ade580c2770c97437b1b1af /sound/pci | |
| parent | 907b9bceb41fa46beae93f79cc4a2247df502c0f (diff) | |
| parent | 7c250413e5b7c3dfae89354725b70c76d7621395 (diff) | |
[GFS2] Fix up merge of Linus' kernel into GFS2
This fixes up a couple of conflicts when merging up with
Linus' latest kernel. This will hopefully allow GFS2 to
be more easily merged into forthcoming -mm and FC kernels
due to the "one line per header" format now used for the
kernel headers.
Signed-off-by: Steven Whitehouse <swhiteho@redhat.com>
Conflicts:
include/linux/Kbuild
include/linux/kernel.h
Diffstat (limited to 'sound/pci')
49 files changed, 2346 insertions, 601 deletions
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index e49c0fe21b0d..8a6b1803c763 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -475,6 +475,7 @@ config SND_FM801_TEA575X depends on SND_FM801_TEA575X_BOOL default SND_FM801 select VIDEO_V4L1 + select VIDEO_DEV config SND_HDA_INTEL tristate "Intel HD Audio" @@ -743,4 +744,17 @@ config SND_YMFPCI To compile this driver as a module, choose M here: the module will be called snd-ymfpci. +config SND_AC97_POWER_SAVE + bool "AC97 Power-Saving Mode" + depends on SND_AC97_CODEC && EXPERIMENTAL + default n + help + Say Y here to enable the aggressive power-saving support of + AC97 codecs. In this mode, the power-mode is dynamically + controlled at each open/close. + + The mode is activated by passing power_save=1 option to + snd-ac97-codec driver. You can toggle it dynamically over + sysfs, too. + endmenu diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 51e83d7a839a..a79e91850ba3 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -31,6 +31,7 @@ #include <linux/mutex.h> #include <sound/core.h> #include <sound/pcm.h> +#include <sound/tlv.h> #include <sound/ac97_codec.h> #include <sound/asoundef.h> #include <sound/initval.h> @@ -47,6 +48,11 @@ static int enable_loopback; module_param(enable_loopback, bool, 0444); MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control"); +#ifdef CONFIG_SND_AC97_POWER_SAVE +static int power_save; +module_param(power_save, bool, 0644); +MODULE_PARM_DESC(power_save, "Enable AC97 power-saving control"); +#endif /* */ @@ -151,7 +157,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x4e534300, 0xffffffff, "LM4540,43,45,46,48", NULL, NULL }, // only guess --jk { 0x4e534331, 0xffffffff, "LM4549", NULL, NULL }, { 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix -{ 0x50534304, 0xffffffff, "UCB1400", NULL, NULL }, +{ 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, @@ -187,6 +193,8 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { }; +static void update_power_regs(struct snd_ac97 *ac97); + /* * I/O routines */ @@ -554,6 +562,18 @@ int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value } err = snd_ac97_update_bits(ac97, reg, val_mask, val); snd_ac97_page_restore(ac97, page_save); +#ifdef CONFIG_SND_AC97_POWER_SAVE + /* check analog mixer power-down */ + if ((val_mask & 0x8000) && + (kcontrol->private_value & (1<<30))) { + if (val & 0x8000) + ac97->power_up &= ~(1 << (reg>>1)); + else + ac97->power_up |= 1 << (reg>>1); + if (power_save) + update_power_regs(ac97); + } +#endif return err; } @@ -962,6 +982,10 @@ static int snd_ac97_bus_dev_free(struct snd_device *device) static int snd_ac97_free(struct snd_ac97 *ac97) { if (ac97) { +#ifdef CONFIG_SND_AC97_POWER_SAVE + if (ac97->power_workq) + destroy_workqueue(ac97->power_workq); +#endif snd_ac97_proc_done(ac97); if (ac97->bus) ac97->bus->codec[ac97->num] = NULL; @@ -1117,7 +1141,9 @@ struct snd_kcontrol *snd_ac97_cnew(const struct snd_kcontrol_new *_template, str /* * create mute switch(es) for normal stereo controls */ -static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg, int check_stereo, struct snd_ac97 *ac97) +static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg, + int check_stereo, int check_amix, + struct snd_ac97 *ac97) { struct snd_kcontrol *kctl; int err; @@ -1137,10 +1163,14 @@ static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg, } if (mute_mask == 0x8080) { struct snd_kcontrol_new tmp = AC97_DOUBLE(name, reg, 15, 7, 1, 1); + if (check_amix) + tmp.private_value |= (1 << 30); tmp.index = ac97->num; kctl = snd_ctl_new1(&tmp, ac97); } else { struct snd_kcontrol_new tmp = AC97_SINGLE(name, reg, 15, 1, 1); + if (check_amix) + tmp.private_value |= (1 << 30); tmp.index = ac97->num; kctl = snd_ctl_new1(&tmp, ac97); } @@ -1153,6 +1183,32 @@ static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg, } /* + * set dB information + */ +static DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); +static DECLARE_TLV_DB_SCALE(db_scale_5bit, -4650, 150, 0); +static DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +static DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); +static DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); + +static unsigned int *find_db_scale(unsigned int maxval) +{ + switch (maxval) { + case 0x0f: return db_scale_4bit; + case 0x1f: return db_scale_5bit; + case 0x3f: return db_scale_6bit; + } + return NULL; +} + +static void set_tlv_db_scale(struct snd_kcontrol *kctl, unsigned int *tlv) +{ + kctl->tlv.p = tlv; + if (tlv) + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; +} + +/* * create a volume for normal stereo/mono controls */ static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigned int lo_max, @@ -1174,6 +1230,10 @@ static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigne tmp.index = ac97->num; kctl = snd_ctl_new1(&tmp, ac97); } + if (reg >= AC97_PHONE && reg <= AC97_PCM) + set_tlv_db_scale(kctl, db_scale_5bit_12db_max); + else + set_tlv_db_scale(kctl, find_db_scale(lo_max)); err = snd_ctl_add(card, kctl); if (err < 0) return err; @@ -1186,7 +1246,9 @@ static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigne /* * create a mute-switch and a volume for normal stereo/mono controls */ -static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, int reg, int check_stereo, struct snd_ac97 *ac97) +static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, + int reg, int check_stereo, int check_amix, + struct snd_ac97 *ac97) { int err; char name[44]; @@ -1197,7 +1259,9 @@ static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, int if (snd_ac97_try_bit(ac97, reg, 15)) { sprintf(name, "%s Switch", pfx); - if ((err = snd_ac97_cmute_new_stereo(card, name, reg, check_stereo, ac97)) < 0) + if ((err = snd_ac97_cmute_new_stereo(card, name, reg, + check_stereo, check_amix, + ac97)) < 0) return err; } check_volume_resolution(ac97, reg, &lo_max, &hi_max); @@ -1209,8 +1273,10 @@ static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx, int return 0; } -#define snd_ac97_cmix_new(card, pfx, reg, ac97) snd_ac97_cmix_new_stereo(card, pfx, reg, 0, ac97) -#define snd_ac97_cmute_new(card, name, reg, ac97) snd_ac97_cmute_new_stereo(card, name, reg, 0, ac97) +#define snd_ac97_cmix_new(card, pfx, reg, acheck, ac97) \ + snd_ac97_cmix_new_stereo(card, pfx, reg, 0, acheck, ac97) +#define snd_ac97_cmute_new(card, name, reg, acheck, ac97) \ + snd_ac97_cmute_new_stereo(card, name, reg, 0, acheck, ac97) static unsigned int snd_ac97_determine_spdif_rates(struct snd_ac97 *ac97); @@ -1226,9 +1292,11 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) /* AD claims to remove this control from AD1887, although spec v2.2 does not allow this */ if (snd_ac97_try_volume_mix(ac97, AC97_MASTER)) { if (ac97->flags & AC97_HAS_NO_MASTER_VOL) - err = snd_ac97_cmute_new(card, "Master Playback Switch", AC97_MASTER, ac97); + err = snd_ac97_cmute_new(card, "Master Playback Switch", + AC97_MASTER, 0, ac97); else - err = snd_ac97_cmix_new(card, "Master Playback", AC97_MASTER, ac97); + err = snd_ac97_cmix_new(card, "Master Playback", + AC97_MASTER, 0, ac97); if (err < 0) return err; } @@ -1245,6 +1313,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) snd_ac97_change_volume_params2(ac97, AC97_CENTER_LFE_MASTER, 0, &max); kctl->private_value &= ~(0xff << 16); kctl->private_value |= (int)max << 16; + set_tlv_db_scale(kctl, find_db_scale(max)); snd_ac97_write_cache(ac97, AC97_CENTER_LFE_MASTER, ac97->regs[AC97_CENTER_LFE_MASTER] | max); } @@ -1258,6 +1327,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) snd_ac97_change_volume_params2(ac97, AC97_CENTER_LFE_MASTER, 8, &max); kctl->private_value &= ~(0xff << 16); kctl->private_value |= (int)max << 16; + set_tlv_db_scale(kctl, find_db_scale(max)); snd_ac97_write_cache(ac97, AC97_CENTER_LFE_MASTER, ac97->regs[AC97_CENTER_LFE_MASTER] | max << 8); } @@ -1265,19 +1335,23 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) if ((snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER)) && !(ac97->flags & AC97_AD_MULTI)) { /* Surround Master (0x38) is with stereo mutes */ - if ((err = snd_ac97_cmix_new_stereo(card, "Surround Playback", AC97_SURROUND_MASTER, 1, ac97)) < 0) + if ((err = snd_ac97_cmix_new_stereo(card, "Surround Playback", + AC97_SURROUND_MASTER, 1, 0, + ac97)) < 0) return err; } /* build headphone controls */ if (snd_ac97_try_volume_mix(ac97, AC97_HEADPHONE)) { - if ((err = snd_ac97_cmix_new(card, "Headphone Playback", AC97_HEADPHONE, ac97)) < 0) + if ((err = snd_ac97_cmix_new(card, "Headphone Playback", + AC97_HEADPHONE, 0, ac97)) < 0) return err; } /* build master mono controls */ if (snd_ac97_try_volume_mix(ac97, AC97_MASTER_MONO)) { - if ((err = snd_ac97_cmix_new(card, "Master Mono Playback", AC97_MASTER_MONO, ac97)) < 0) + if ((err = snd_ac97_cmix_new(card, "Master Mono Playback", + AC97_MASTER_MONO, 0, ac97)) < 0) return err; } @@ -1301,8 +1375,9 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) ((ac97->flags & AC97_HAS_PC_BEEP) || snd_ac97_try_volume_mix(ac97, AC97_PC_BEEP))) { for (idx = 0; idx < 2; idx++) - if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_pc_beep[idx], ac97))) < 0) + if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_pc_beep[idx], ac97))) < 0) return err; + set_tlv_db_scale(kctl, db_scale_4bit); snd_ac97_write_cache(ac97, AC97_PC_BEEP, snd_ac97_read(ac97, AC97_PC_BEEP) | 0x801e); } @@ -1310,7 +1385,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) /* build Phone controls */ if (!(ac97->flags & AC97_HAS_NO_PHONE)) { if (snd_ac97_try_volume_mix(ac97, AC97_PHONE)) { - if ((err = snd_ac97_cmix_new(card, "Phone Playback", AC97_PHONE, ac97)) < 0) + if ((err = snd_ac97_cmix_new(card, "Phone Playback", + AC97_PHONE, 1, ac97)) < 0) return err; } } @@ -1318,7 +1394,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) /* build MIC controls */ if (!(ac97->flags & AC97_HAS_NO_MIC)) { if (snd_ac97_try_volume_mix(ac97, AC97_MIC)) { - if ((err = snd_ac97_cmix_new(card, "Mic Playback", AC97_MIC, ac97)) < 0) + if ((err = snd_ac97_cmix_new(card, "Mic Playback", + AC97_MIC, 1, ac97)) < 0) return err; if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic_boost, ac97))) < 0) return err; @@ -1327,14 +1404,16 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) /* build Line controls */ if (snd_ac97_try_volume_mix(ac97, AC97_LINE)) { - if ((err = snd_ac97_cmix_new(card, "Line Playback", AC97_LINE, ac97)) < 0) + if ((err = snd_ac97_cmix_new(card, "Line Playback", + AC97_LINE, 1, ac97)) < 0) return err; } /* build CD controls */ if (!(ac97->flags & AC97_HAS_NO_CD)) { if (snd_ac97_try_volume_mix(ac97, AC97_CD)) { - if ((err = snd_ac97_cmix_new(card, "CD Playback", AC97_CD, ac97)) < 0) + if ((err = snd_ac97_cmix_new(card, "CD Playback", + AC97_CD, 1, ac97)) < 0) return err; } } @@ -1342,7 +1421,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) /* build Video controls */ if (!(ac97->flags & AC97_HAS_NO_VIDEO)) { if (snd_ac97_try_volume_mix(ac97, AC97_VIDEO)) { - if ((err = snd_ac97_cmix_new(card, "Video Playback", AC97_VIDEO, ac97)) < 0) + if ((err = snd_ac97_cmix_new(card, "Video Playback", + AC97_VIDEO, 1, ac97)) < 0) return err; } } @@ -1350,7 +1430,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) /* build Aux controls */ if (!(ac97->flags & AC97_HAS_NO_AUX)) { if (snd_ac97_try_volume_mix(ac97, AC97_AUX)) { - if ((err = snd_ac97_cmix_new(card, "Aux Playback", AC97_AUX, ac97)) < 0) + if ((err = snd_ac97_cmix_new(card, "Aux Playback", + AC97_AUX, 1, ac97)) < 0) return err; } } @@ -1363,31 +1444,38 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) else init_val = 0x9f1f; for (idx = 0; idx < 2; idx++) - if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_ad18xx_pcm[idx], ac97))) < 0) + if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_pcm[idx], ac97))) < 0) return err; + set_tlv_db_scale(kctl, db_scale_5bit); ac97->spec.ad18xx.pcmreg[0] = init_val; if (ac97->scaps & AC97_SCAP_SURROUND_DAC) { for (idx = 0; idx < 2; idx++) - if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_ad18xx_surround[idx], ac97))) < 0) + if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_surround[idx], ac97))) < 0) return err; + set_tlv_db_scale(kctl, db_scale_5bit); ac97->spec.ad18xx.pcmreg[1] = init_val; } if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC) { for (idx = 0; idx < 2; idx++) - if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_ad18xx_center[idx], ac97))) < 0) + if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_center[idx], ac97))) < 0) return err; + set_tlv_db_scale(kctl, db_scale_5bit); for (idx = 0; idx < 2; idx++) - if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_ad18xx_lfe[idx], ac97))) < 0) + if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_lfe[idx], ac97))) < 0) return err; + set_tlv_db_scale(kctl, db_scale_5bit); ac97->spec.ad18xx.pcmreg[2] = init_val; } snd_ac97_write_cache(ac97, AC97_PCM, init_val); } else { if (!(ac97->flags & AC97_HAS_NO_STD_PCM)) { if (ac97->flags & AC97_HAS_NO_PCM_VOL) - err = snd_ac97_cmute_new(card, "PCM Playback Switch", AC97_PCM, ac97); + err = snd_ac97_cmute_new(card, + "PCM Playback Switch", + AC97_PCM, 0, ac97); else - err = snd_ac97_cmix_new(card, "PCM Playback", AC97_PCM, ac97); + err = snd_ac97_cmix_new(card, "PCM Playback", + AC97_PCM, 0, ac97); if (err < 0) return err; } @@ -1398,19 +1486,23 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_control_capture_src, ac97))) < 0) return err; if (snd_ac97_try_bit(ac97, AC97_REC_GAIN, 15)) { - if ((err = snd_ac97_cmute_new(card, "Capture Switch", AC97_REC_GAIN, ac97)) < 0) + err = snd_ac97_cmute_new(card, "Capture Switch", + AC97_REC_GAIN, 0, ac97); + if (err < 0) return err; } - if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_control_capture_vol, ac97))) < 0) + if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_control_capture_vol, ac97))) < 0) return err; + set_tlv_db_scale(kctl, db_scale_rec_gain); snd_ac97_write_cache(ac97, AC97_REC_SEL, 0x0000); snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x0000); } /* build MIC Capture controls */ if (snd_ac97_try_volume_mix(ac97, AC97_REC_GAIN_MIC)) { for (idx = 0; idx < 2; idx++) - if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic_capture[idx], ac97))) < 0) + if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_mic_capture[idx], ac97))) < 0) return err; + set_tlv_db_scale(kctl, db_scale_rec_gain); snd_ac97_write_cache(ac97, AC97_REC_GAIN_MIC, 0x0000); } @@ -1481,6 +1573,12 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build S/PDIF controls */ + + /* Hack for ASUS P5P800-VM, which does not indicate S/PDIF capability */ + if (ac97->subsystem_vendor == 0x1043 && + ac97->subsystem_device == 0x810f) + ac97->ext_id |= AC97_EI_SPDIF; + if ((ac97->ext_id & AC97_EI_SPDIF) && !(ac97->scaps & AC97_SCAP_NO_SPDIF)) { if (ac97->build_ops->build_spdif) { if ((err = ac97->build_ops->build_spdif(ac97)) < 0) @@ -1817,18 +1915,25 @@ static int snd_ac97_dev_register(struct snd_device *device) return 0; } -/* unregister ac97 codec */ -static int snd_ac97_dev_unregister(struct snd_device *device) +/* disconnect ac97 codec */ +static int snd_ac97_dev_disconnect(struct snd_device *device) { struct snd_ac97 *ac97 = device->device_data; if (ac97->dev.bus) device_unregister(&ac97->dev); - return snd_ac97_free(ac97); + return 0; } /* build_ops to do nothing */ static struct snd_ac97_build_ops null_build_ops; +#ifdef CONFIG_SND_AC97_POWER_SAVE +static void do_update_power(void *data) +{ + update_power_regs(data); +} +#endif + /** * snd_ac97_mixer - create an Codec97 component * @bus: the AC97 bus which codec is attached to @@ -1860,7 +1965,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, static struct snd_device_ops ops = { .dev_free = snd_ac97_dev_free, .dev_register = snd_ac97_dev_register, - .dev_unregister = snd_ac97_dev_unregister, + .dev_disconnect = snd_ac97_dev_disconnect, }; snd_assert(rac97 != NULL, return -EINVAL); @@ -1883,6 +1988,10 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, bus->codec[ac97->num] = ac97; mutex_init(&ac97->reg_mutex); mutex_init(&ac97->page_mutex); +#ifdef CONFIG_SND_AC97_POWER_SAVE + ac97->power_workq = create_workqueue("ac97"); + INIT_WORK(&ac97->power_work, do_update_power, ac97); +#endif #ifdef CONFIG_PCI if (ac97->pci) { @@ -2117,15 +2226,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, return -ENOMEM; } } - /* make sure the proper powerdown bits are cleared */ - if (ac97->scaps && ac97_is_audio(ac97)) { - reg = snd_ac97_read(ac97, AC97_EXTENDED_STATUS); - if (ac97->scaps & AC97_SCAP_SURROUND_DAC) - reg &= ~AC97_EA_PRJ; - if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC) - reg &= ~(AC97_EA_PRI | AC97_EA_PRK); - snd_ac97_write_cache(ac97, AC97_EXTENDED_STATUS, reg); - } + if (ac97_is_audio(ac97)) + update_power_regs(ac97); snd_ac97_proc_init(ac97); if ((err = snd_device_new(card, SNDRV_DEV_CODEC, ac97, &ops)) < 0) { snd_ac97_free(ac97); @@ -2153,19 +2255,152 @@ static void snd_ac97_powerdown(struct snd_ac97 *ac97) snd_ac97_write(ac97, AC97_HEADPHONE, 0x9f9f); } - power = ac97->regs[AC97_POWERDOWN] | 0x8000; /* EAPD */ - power |= 0x4000; /* Headphone amplifier powerdown */ - power |= 0x0300; /* ADC & DAC powerdown */ - snd_ac97_write(ac97, AC97_POWERDOWN, power); - udelay(100); - power |= 0x0400; /* Analog Mixer powerdown (Vref on) */ + /* surround, CLFE, mic powerdown */ + power = ac97->regs[AC97_EXTENDED_STATUS]; + if (ac97->scaps & AC97_SCAP_SURROUND_DAC) + power |= AC97_EA_PRJ; + if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC) + power |= AC97_EA_PRI | AC97_EA_PRK; + power |= AC97_EA_PRL; + snd_ac97_write(ac97, AC97_EXTENDED_STATUS, power); + + /* powerdown external amplifier */ + if (ac97->scaps & AC97_SCAP_INV_EAPD) + power = ac97->regs[AC97_POWERDOWN] & ~AC97_PD_EAPD; + else if (! (ac97->scaps & AC97_SCAP_EAPD_LED)) + power = ac97->regs[AC97_POWERDOWN] | AC97_PD_EAPD; + power |= AC97_PD_PR6; /* Headphone amplifier powerdown */ + power |= AC97_PD_PR0 | AC97_PD_PR1; /* ADC & DAC powerdown */ snd_ac97_write(ac97, AC97_POWERDOWN, power); udelay(100); -#if 0 - /* FIXME: this causes click noises on some boards at resume */ - power |= 0x3800; /* AC-link powerdown, internal Clk disable */ + power |= AC97_PD_PR2 | AC97_PD_PR3; /* Analog Mixer powerdown */ snd_ac97_write(ac97, AC97_POWERDOWN, power); +#ifdef CONFIG_SND_AC97_POWER_SAVE + if (power_save) { + udelay(100); + /* AC-link powerdown, internal Clk disable */ + /* FIXME: this may cause click noises on some boards */ + power |= AC97_PD_PR4 | AC97_PD_PR5; + snd_ac97_write(ac97, AC97_POWERDOWN, power); + } +#endif +} + + +struct ac97_power_reg { + unsigned short reg; + unsigned short power_reg; + unsigned short mask; +}; + +enum { PWIDX_ADC, PWIDX_FRONT, PWIDX_CLFE, PWIDX_SURR, PWIDX_MIC, PWIDX_SIZE }; + +static struct ac97_power_reg power_regs[PWIDX_SIZE] = { + [PWIDX_ADC] = { AC97_PCM_LR_ADC_RATE, AC97_POWERDOWN, AC97_PD_PR0}, + [PWIDX_FRONT] = { AC97_PCM_FRONT_DAC_RATE, AC97_POWERDOWN, AC97_PD_PR1}, + [PWIDX_CLFE] = { AC97_PCM_LFE_DAC_RATE, AC97_EXTENDED_STATUS, + AC97_EA_PRI | AC97_EA_PRK}, + [PWIDX_SURR] = { AC97_PCM_SURR_DAC_RATE, AC97_EXTENDED_STATUS, + AC97_EA_PRJ}, + [PWIDX_MIC] = { AC97_PCM_MIC_ADC_RATE, AC97_EXTENDED_STATUS, + AC97_EA_PRL}, +}; + +#ifdef CONFIG_SND_AC97_POWER_SAVE +/** + * snd_ac97_update_power - update the powerdown register + * @ac97: the codec instance + * @reg: the rate register, e.g. AC97_PCM_FRONT_DAC_RATE + * @powerup: non-zero when power up the part + * + * Update the AC97 powerdown register bits of the given part. + */ +int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup) +{ + int i; + + if (! ac97) + return 0; + + if (reg) { + /* SPDIF requires DAC power, too */ + if (reg == AC97_SPDIF) + reg = AC97_PCM_FRONT_DAC_RATE; + for (i = 0; i < PWIDX_SIZE; i++) { + if (power_regs[i].reg == reg) { + if (powerup) + ac97->power_up |= (1 << i); + else + ac97->power_up &= ~(1 << i); + break; + } + } + } + + if (! power_save) + return 0; + + if (! powerup && ac97->power_workq) + /* adjust power-down bits after two seconds delay + * (for avoiding loud click noises for many (OSS) apps + * that open/close frequently) + */ + queue_delayed_work(ac97->power_workq, &ac97->power_work, HZ*2); + else + update_power_regs(ac97); + + return 0; +} + +EXPORT_SYMBOL(snd_ac97_update_power); +#endif /* CONFIG_SND_AC97_POWER_SAVE */ + +static void update_power_regs(struct snd_ac97 *ac97) +{ + unsigned int power_up, bits; + int i; + +#ifdef CONFIG_SND_AC97_POWER_SAVE + if (power_save) + power_up = ac97->power_up; + else { #endif + power_up = (1 << PWIDX_FRONT) | (1 << PWIDX_ADC); + power_up |= (1 << PWIDX_MIC); + if (ac97->scaps & AC97_SCAP_SURROUND_DAC) + power_up |= (1 << PWIDX_SURR); + if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC) + power_up |= (1 << PWIDX_CLFE); +#ifdef CONFIG_SND_AC97_POWER_SAVE + } +#endif + if (power_up) { + if (ac97->regs[AC97_POWERDOWN] & AC97_PD_PR2) { + /* needs power-up analog mix and vref */ + snd_ac97_update_bits(ac97, AC97_POWERDOWN, + AC97_PD_PR3, 0); + msleep(1); + snd_ac97_update_bits(ac97, AC97_POWERDOWN, + AC97_PD_PR2, 0); + } + } + for (i = 0; i < PWIDX_SIZE; i++) { + if (power_up & (1 << i)) + bits = 0; + else + bits = power_regs[i].mask; + snd_ac97_update_bits(ac97, power_regs[i].power_reg, + power_regs[i].mask, bits); + } + if (! power_up) { + if (! (ac97->regs[AC97_POWERDOWN] & AC97_PD_PR2)) { + /* power down analog mix and vref */ + snd_ac97_update_bits(ac97, AC97_POWERDOWN, + AC97_PD_PR2, AC97_PD_PR2); + snd_ac97_update_bits(ac97, AC97_POWERDOWN, + AC97_PD_PR3, AC97_PD_PR3); + } + } } @@ -2484,6 +2719,7 @@ static int tune_mute_led(struct snd_ac97 *ac97) msw->put = master_mute_sw_put; snd_ac97_remove_ctl(ac97, "External Amplifier", NULL); snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, 0x8000); /* mute LED on */ + ac97->scaps |= AC97_SCAP_EAPD_LED; return 0; } diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 094cfc1f3a19..dc28b111a06d 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -32,6 +32,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/control.h> +#include <sound/tlv.h> #include <sound/ac97_codec.h> #include "ac97_patch.h" #include "ac97_id.h" @@ -51,6 +52,20 @@ static int patch_build_controls(struct snd_ac97 * ac97, const struct snd_kcontro return 0; } +/* replace with a new TLV */ +static void reset_tlv(struct snd_ac97 *ac97, const char *name, + unsigned int *tlv) +{ + struct snd_ctl_elem_id sid; + struct snd_kcontrol *kctl; + memset(&sid, 0, sizeof(sid)); + strcpy(sid.name, name); + sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kctl = snd_ctl_find_id(ac97->bus->card, &sid); + if (kctl && kctl->tlv.p) + kctl->tlv.p = tlv; +} + /* set to the page, update bits and restore the page */ static int ac97_update_bits_page(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value, unsigned short page) { @@ -466,7 +481,7 @@ int patch_wolfson05(struct snd_ac97 * ac97) ac97->build_ops = &patch_wolfson_wm9705_ops; #ifdef CONFIG_TOUCHSCREEN_WM9705 /* WM9705 touchscreen uses AUX and VIDEO for touch */ - ac97->flags |=3D AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX; + ac97->flags |= AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX; #endif return 0; } @@ -1380,6 +1395,17 @@ static void ad1888_resume(struct snd_ac97 *ac97) #endif +static const struct snd_ac97_res_table ad1819_restbl[] = { + { AC97_PHONE, 0x9f1f }, + { AC97_MIC, 0x9f1f }, + { AC97_LINE, 0x9f1f }, + { AC97_CD, 0x9f1f }, + { AC97_VIDEO, 0x9f1f }, + { AC97_AUX, 0x9f1f }, + { AC97_PCM, 0x9f1f }, + { } /* terminator */ +}; + int patch_ad1819(struct snd_ac97 * ac97) { unsigned short scfg; @@ -1387,6 +1413,7 @@ int patch_ad1819(struct snd_ac97 * ac97) // patch for Analog Devices scfg = snd_ac97_read(ac97, AC97_AD_SERIAL_CFG); snd_ac97_write_cache(ac97, AC97_AD_SERIAL_CFG, scfg | 0x7000); /* select all codecs */ + ac97->res_table = ad1819_restbl; return 0; } @@ -1522,12 +1549,16 @@ static const struct snd_kcontrol_new snd_ac97_controls_ad1885[] = { AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 8, 1, 1), /* inverted */ }; +static DECLARE_TLV_DB_SCALE(db_scale_6bit_6db_max, -8850, 150, 0); + static int patch_ad1885_specific(struct snd_ac97 * ac97) { int err; if ((err = patch_build_controls(ac97, snd_ac97_controls_ad1885, ARRAY_SIZE(snd_ac97_controls_ad1885))) < 0) return err; + reset_tlv(ac97, "Headphone Playback Volume", + db_scale_6bit_6db_max); return 0; } @@ -1551,12 +1582,27 @@ int patch_ad1885(struct snd_ac97 * ac97) return 0; } +static int patch_ad1886_specific(struct snd_ac97 * ac97) +{ + reset_tlv(ac97, "Headphone Playback Volume", + db_scale_6bit_6db_max); + return 0; +} + +static struct snd_ac97_build_ops patch_ad1886_build_ops = { + .build_specific = &patch_ad1886_specific, +#ifdef CONFIG_PM + .resume = ad18xx_resume +#endif +}; + int patch_ad1886(struct snd_ac97 * ac97) { patch_ad1881(ac97); /* Presario700 workaround */ /* for Jack Sense/SPDIF Register misetting causing */ snd_ac97_write_cache(ac97, AC97_AD_JACK_SPDIF, 0x0010); + ac97->build_ops = &patch_ad1886_build_ops; return 0; } @@ -2015,6 +2061,8 @@ static const struct snd_kcontrol_new snd_ac97_spdif_controls_alc650[] = { /* AC97_SINGLE("IEC958 Input Monitor", AC97_ALC650_MULTICH, 13, 1, 0), */ }; +static DECLARE_TLV_DB_SCALE(db_scale_5bit_3db_max, -4350, 150, 0); + static int patch_alc650_specific(struct snd_ac97 * ac97) { int err; @@ -2025,6 +2073,9 @@ static int patch_alc650_specific(struct snd_ac97 * ac97) if ((err = patch_build_controls(ac97, snd_ac97_spdif_controls_alc650, ARRAY_SIZE(snd_ac97_spdif_controls_alc650))) < 0) return err; } + if (ac97->id != AC97_ID_ALC650F) + reset_tlv(ac97, "Master Playback Volume", + db_scale_5bit_3db_max); return 0; } @@ -2208,7 +2259,8 @@ int patch_alc655(struct snd_ac97 * ac97) val &= ~(1 << 1); /* Pin 47 is spdif input pin */ else { /* ALC655 */ if (ac97->subsystem_vendor == 0x1462 && - ac97->subsystem_device == 0x0131) /* MSI S270 laptop */ + (ac97->subsystem_device == 0x0131 || /* MSI S270 laptop */ + ac97->subsystem_device == 0x0161)) /* LG K1 Express */ val &= ~(1 << 1); /* Pin 47 is EAPD (for internal speaker) */ else val |= (1 << 1); /* Pin 47 is spdif input pin */ @@ -2759,6 +2811,10 @@ int patch_vt1616(struct snd_ac97 * ac97) */ int patch_vt1617a(struct snd_ac97 * ac97) { + /* bring analog power consumption to normal, like WinXP driver + * for EPIA SP + */ + snd_ac97_write_cache(ac97, 0x5c, 0x20); ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; return 0; @@ -2872,3 +2928,41 @@ int patch_lm4550(struct snd_ac97 *ac97) ac97->res_table = lm4550_restbl; return 0; } + +/* + * UCB1400 codec (http://www.semiconductors.philips.com/acrobat_download/datasheets/UCB1400-02.pdf) + */ +static const struct snd_kcontrol_new snd_ac97_controls_ucb1400[] = { +/* enable/disable headphone driver which allows direct connection to + stereo headphone without the use of external DC blocking + capacitors */ +AC97_SINGLE("Headphone Driver", 0x6a, 6, 1, 0), +/* Filter used to compensate the DC offset is added in the ADC to remove idle + tones from the audio band. */ +AC97_SINGLE("DC Filter", 0x6a, 4, 1, 0), +/* Control smart-low-power mode feature. Allows automatic power down + of unused blocks in the ADC analog front end and the PLL. */ +AC97_SINGLE("Smart Low Power Mode", 0x6c, 4, 3, 0), +}; + +static int patch_ucb1400_specific(struct snd_ac97 * ac97) +{ + int idx, err; + for (idx = 0; idx < ARRAY_SIZE(snd_ac97_controls_ucb1400); idx++) + if ((err = snd_ctl_add(ac97->bus->card, snd_ctl_new1(&snd_ac97_controls_ucb1400[idx], ac97))) < 0) + return err; + return 0; +} + +static struct snd_ac97_build_ops patch_ucb1400_ops = { + .build_specific = patch_ucb1400_specific, +}; + +int patch_ucb1400(struct snd_ac97 * ac97) +{ + ac97->build_ops = &patch_ucb1400_ops; + /* enable headphone driver and smart low power mode by default */ + snd_ac97_write(ac97, 0x6a, 0x0050); + snd_ac97_write(ac97, 0x6c, 0x0030); + return 0; +} diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h index adcaa04586cb..741979217207 100644 --- a/sound/pci/ac97/ac97_patch.h +++ b/sound/pci/ac97/ac97_patch.h @@ -58,5 +58,6 @@ int patch_cm9780(struct snd_ac97 * ac97); int patch_vt1616(struct snd_ac97 * ac97); int patch_vt1617a(struct snd_ac97 * ac97); int patch_it2646(struct snd_ac97 * ac97); +int patch_ucb1400(struct snd_ac97 * ac97); int mpatch_si3036(struct snd_ac97 * ac97); int patch_lm4550(struct snd_ac97 * ac97); diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index f684aa2c0067..3758d07182f8 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -269,6 +269,7 @@ int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate) return -EINVAL; } + snd_ac97_update_power(ac97, reg, 1); switch (reg) { case AC97_PCM_MIC_ADC_RATE: if ((ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_VRM) == 0) /* MIC VRA */ @@ -606,6 +607,7 @@ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate, goto error; } } + pcm->cur_dbl = r; spin_unlock_irq(&pcm->bus->bus_lock); for (i = 3; i < 12; i++) { if (!(slots & (1 << i))) @@ -651,6 +653,21 @@ int snd_ac97_pcm_close(struct ac97_pcm *pcm) unsigned short slots = pcm->aslots; int i, cidx; +#ifdef CONFIG_SND_AC97_POWER_SAVE + int r = pcm->cur_dbl; + for (i = 3; i < 12; i++) { + if (!(slots & (1 << i))) + continue; + for (cidx = 0; cidx < 4; cidx++) { + if (pcm->r[r].rslots[cidx] & (1 << i)) { + int reg = get_slot_reg(pcm, cidx, i, r); + snd_ac97_update_power(pcm->r[r].codec[cidx], + reg, 0); + } + } + } +#endif + bus = pcm->bus; spin_lock_irq(&pcm->bus->bus_lock); for (i = 3; i < 12; i++) { @@ -660,6 +677,7 @@ int snd_ac97_pcm_close(struct ac97_pcm *pcm) bus->used_slots[pcm->stream][cidx] &= ~(1 << i); } pcm->aslots = 0; + pcm->cur_dbl = 0; spin_unlock_irq(&pcm->bus->bus_lock); return 0; } diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 2118df50b9d6..a3fdd7da911c 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -457,14 +457,10 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) void snd_ac97_proc_done(struct snd_ac97 * ac97) { - if (ac97->proc_regs) { - snd_info_unregister(ac97->proc_regs); - ac97->proc_regs = NULL; - } - if (ac97->proc) { - snd_info_unregister(ac97->proc); - ac97->proc = NULL; - } + snd_info_free_entry(ac97->proc_regs); + ac97->proc_regs = NULL; + snd_info_free_entry(ac97->proc); + ac97->proc = NULL; } void snd_ac97_bus_proc_init(struct snd_ac97_bus * bus) @@ -485,8 +481,6 @@ void snd_ac97_bus_proc_init(struct snd_ac97_bus * bus) void snd_ac97_bus_proc_done(struct snd_ac97_bus * bus) { - if (bus->proc) { - snd_info_unregister(bus->proc); - bus->proc = NULL; - } + snd_info_free_entry(bus->proc); + bus->proc = NULL; } diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c index 94c26ec05882..c153cb79c518 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ac97/ak4531_codec.c @@ -27,6 +27,7 @@ #include <sound/core.h> #include <sound/ak4531_codec.h> +#include <sound/tlv.h> MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>"); MODULE_DESCRIPTION("Universal routines for AK4531 codec"); @@ -63,6 +64,14 @@ static void snd_ak4531_dump(struct snd_ak4531 *ak4531) .info = snd_ak4531_info_single, \ .get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \ .private_value = reg | (shift << 16) | (mask << 24) | (invert << 22) } +#define AK4531_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, .index = xindex, \ + .info = snd_ak4531_info_single, \ + .get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \ + .private_value = reg | (shift << 16) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } static int snd_ak4531_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -122,6 +131,14 @@ static int snd_ak4531_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e .info = snd_ak4531_info_double, \ .get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \ .private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22) } +#define AK4531_DOUBLE_TLV(xname, xindex, left_reg, right_reg, left_shift, right_shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, .index = xindex, \ + .info = snd_ak4531_info_double, \ + .get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \ + .private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } static int snd_ak4531_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -250,50 +267,62 @@ static int snd_ak4531_put_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl return change; } +static DECLARE_TLV_DB_SCALE(db_scale_master, -6200, 200, 0); +static DECLARE_TLV_DB_SCALE(db_scale_mono, -2800, 400, 0); +static DECLARE_TLV_DB_SCALE(db_scale_input, -5000, 200, 0); + static struct snd_kcontrol_new snd_ak4531_controls[] = { -AK4531_DOUBLE("Master Playback Switch", 0, AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1), +AK4531_DOUBLE_TLV("Master Playback Switch", 0, + AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1, + db_scale_master), AK4531_DOUBLE("Master Playback Volume", 0, AK4531_LMASTER, AK4531_RMASTER, 0, 0, 0x1f, 1), -AK4531_SINGLE("Master Mono Playback Switch", 0, AK4531_MONO_OUT, 7, 1, 1), +AK4531_SINGLE_TLV("Master Mono Playback Switch", 0, AK4531_MONO_OUT, 7, 1, 1, + db_scale_mono), AK4531_SINGLE("Master Mono Playback Volume", 0, AK4531_MONO_OUT, 0, 0x07, 1), AK4531_DOUBLE("PCM Switch", 0, AK4531_LVOICE, AK4531_RVOICE, 7, 7, 1, 1), -AK4531_DOUBLE("PCM Volume", 0, AK4531_LVOICE, AK4531_RVOICE, 0, 0, 0x1f, 1), +AK4531_DOUBLE_TLV("PCM Volume", 0, AK4531_LVOICE, AK4531_RVOICE, 0, 0, 0x1f, 1, + db_scale_input), AK4531_DOUBLE("PCM Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 3, 2, 1, 0), AK4531_DOUBLE("PCM Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 2, 2, 1, 0), AK4531_DOUBLE("PCM Switch", 1, AK4531_LFM, AK4531_RFM, 7, 7, 1, 1), -AK4531_DOUBLE("PCM Volume", 1, AK4531_LFM, AK4531_RFM, 0, 0, 0x1f, 1), +AK4531_DOUBLE_TLV("PCM Volume", 1, AK4531_LFM, AK4531_RFM, 0, 0, 0x1f, 1, + db_scale_input), AK4531_DOUBLE("PCM Playback Switch", 1, AK4531_OUT_SW1, AK4531_OUT_SW1, 6, 5, 1, 0), AK4531_INPUT_SW("PCM Capture Route", 1, AK4531_LIN_SW1, AK4531_RIN_SW1, 6, 5), AK4531_DOUBLE("CD Switch", 0, AK4531_LCD, AK4531_RCD, 7, 7, 1, 1), -AK4531_DOUBLE("CD Volume", 0, AK4531_LCD, AK4531_RCD, 0, 0, 0x1f, 1), +AK4531_DOUBLE_TLV("CD Volume", 0, AK4531_LCD, AK4531_RCD, 0, 0, 0x1f, 1, + db_scale_input), AK4531_DOUBLE("CD Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 2, 1, 1, 0), AK4531_INPUT_SW("CD Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 2, 1), AK4531_DOUBLE("Line Switch", 0, AK4531_LLINE, AK4531_RLINE, 7, 7, 1, 1), -AK4531_DOUBLE("Line Volume", 0, AK4531_LLINE, AK4531_RLINE, 0, 0, 0x1f, 1), +AK4531_DOUBLE_TLV("Line Volume", 0, AK4531_LLINE, AK4531_RLINE, 0, 0, 0x1f, 1, + db_scale_input), AK4531_DOUBLE("Line Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 4, 3, 1, 0), AK4531_INPUT_SW("Line Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 4, 3), AK4531_DOUBLE("Aux Switch", 0, AK4531_LAUXA, AK4531_RAUXA, 7, 7, 1, 1), -AK4531_DOUBLE("Aux Volume", 0, AK4531_LAUXA, AK4531_RAUXA, 0, 0, 0x1f, 1), +AK4531_DOUBLE_TLV("Aux Volume", 0, AK4531_LAUXA, AK4531_RAUXA, 0, 0, 0x1f, 1, + db_scale_input), AK4531_DOUBLE("Aux Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 5, 4, 1, 0), AK4531_INPUT_SW("Aux Capture Route", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 4, 3), AK4531_SINGLE("Mono Switch", 0, AK4531_MONO1, 7, 1, 1), -AK4531_SINGLE("Mono Volume", 0, AK4531_MONO1, 0, 0x1f, 1), +AK4531_SINGLE_TLV("Mono Volume", 0, AK4531_MONO1, 0, 0x1f, 1, db_scale_input), AK4531_SINGLE("Mono Playback Switch", 0, AK4531_OUT_SW2, 0, 1, 0), AK4531_DOUBLE("Mono Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 0, 0, 1, 0), AK4531_SINGLE("Mono Switch", 1, AK4531_MONO2, 7, 1, 1), -AK4531_SINGLE("Mono Volume", 1, AK4531_MONO2, 0, 0x1f, 1), +AK4531_SINGLE_TLV("Mono Volume", 1, AK4531_MONO2, 0, 0x1f, 1, db_scale_input), AK4531_SINGLE("Mono Playback Switch", 1, AK4531_OUT_SW2, 1, 1, 0), AK4531_DOUBLE("Mono Capture Switch", 1, AK4531_LIN_SW2, AK4531_RIN_SW2, 1, 1, 1, 0), -AK4531_SINGLE("Mic Volume", 0, AK4531_MIC, 0, 0x1f, 1), +AK4531_SINGLE_TLV("Mic Volume", 0, AK4531_MIC, 0, 0x1f, 1, db_scale_input), AK4531_SINGLE("Mic Switch", 0, AK4531_MIC, 7, 1, 1), AK4531_SINGLE("Mic Playback Switch", 0, AK4531_OUT_SW1, 0, 1, 0), AK4531_DOUBLE("Mic Capture Switch", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 0, 0, 1, 0), diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 146eed70dce6..9855f528ea78 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -70,9 +70,13 @@ #include <sound/pcm.h> #include <sound/ac97_codec.h> #include <sound/info.h> +#include <sound/tlv.h> #include "ca0106.h" +static DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1); +static DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1); + static int snd_ca0106_shared_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -469,18 +473,24 @@ static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol, #define CA_VOLUME(xname,chid,reg) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_ca0106_volume_info, \ .get = snd_ca0106_volume_get, \ .put = snd_ca0106_volume_put, \ + .tlv = { .p = snd_ca0106_db_scale1 }, \ .private_value = ((chid) << 8) | (reg) \ } #define I2C_VOLUME(xname,chid) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_ca0106_i2c_volume_info, \ .get = snd_ca0106_i2c_volume_get, \ .put = snd_ca0106_i2c_volume_put, \ + .tlv = { .p = snd_ca0106_db_scale2 }, \ .private_value = chid \ } diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 9631456ec3de..1990430a21c1 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -33,6 +33,7 @@ #include <sound/pcm.h> #include <sound/rawmidi.h> #include <sound/ac97_codec.h> +#include <sound/tlv.h> #include <sound/opl3.h> #include <sound/initval.h> @@ -1054,6 +1055,8 @@ static int snd_cs4281_put_volume(struct snd_kcontrol *kcontrol, return change; } +static DECLARE_TLV_DB_SCALE(db_scale_dsp, -4650, 150, 0); + static struct snd_kcontrol_new snd_cs4281_fm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1062,6 +1065,7 @@ static struct snd_kcontrol_new snd_cs4281_fm_vol = .get = snd_cs4281_get_volume, .put = snd_cs4281_put_volume, .private_value = ((BA0_FMLVC << 16) | BA0_FMRVC), + .tlv = { .p = db_scale_dsp }, }; static struct snd_kcontrol_new snd_cs4281_pcm_vol = @@ -1072,6 +1076,7 @@ static struct snd_kcontrol_new snd_cs4281_pcm_vol = .get = snd_cs4281_get_volume, .put = snd_cs4281_put_volume, .private_value = ((BA0_PPLVC << 16) | BA0_PPRVC), + .tlv = { .p = db_scale_dsp }, }; static void snd_cs4281_mixer_free_ac97_bus(struct snd_ac97_bus *bus) diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 5c9711c0265c..89c402770a1d 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -868,35 +868,23 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; int i; - if (ins->proc_sym_info_entry) { - snd_info_unregister(ins->proc_sym_info_entry); - ins->proc_sym_info_entry = NULL; - } - - if (ins->proc_modules_info_entry) { - snd_info_unregister(ins->proc_modules_info_entry); - ins->proc_modules_info_entry = NULL; - } - - if (ins->proc_parameter_dump_info_entry) { - snd_info_unregister(ins->proc_parameter_dump_info_entry); - ins->proc_parameter_dump_info_entry = NULL; - } - - if (ins->proc_sample_dump_info_entry) { - snd_info_unregister(ins->proc_sample_dump_info_entry); - ins->proc_sample_dump_info_entry = NULL; - } - - if (ins->proc_scb_info_entry) { - snd_info_unregister(ins->proc_scb_info_entry); - ins->proc_scb_info_entry = NULL; - } - - if (ins->proc_task_info_entry) { - snd_info_unregister(ins->proc_task_info_entry); - ins->proc_task_info_entry = NULL; - } + snd_info_free_entry(ins->proc_sym_info_entry); + ins->proc_sym_info_entry = NULL; + + snd_info_free_entry(ins->proc_modules_info_entry); + ins->proc_modules_info_entry = NULL; + + snd_info_free_entry(ins->proc_parameter_dump_info_entry); + ins->proc_parameter_dump_info_entry = NULL; + + snd_info_free_entry(ins->proc_sample_dump_info_entry); + ins->proc_sample_dump_info_entry = NULL; + + snd_info_free_entry(ins->proc_scb_info_entry); + ins->proc_scb_info_entry = NULL; + + snd_info_free_entry(ins->proc_task_info_entry); + ins->proc_task_info_entry = NULL; mutex_lock(&chip->spos_mutex); for (i = 0; i < ins->nscb; ++i) { @@ -905,10 +893,8 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) } mutex_unlock(&chip->spos_mutex); - if (ins->proc_dsp_dir) { - snd_info_unregister (ins->proc_dsp_dir); - ins->proc_dsp_dir = NULL; - } + snd_info_free_entry(ins->proc_dsp_dir); + ins->proc_dsp_dir = NULL; return 0; } diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 232b337852ff..343f51d5311b 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -233,7 +233,7 @@ void cs46xx_dsp_proc_free_scb_desc (struct dsp_scb_descriptor * scb) snd_printdd("cs46xx_dsp_proc_free_scb_desc: freeing %s\n",scb->scb_name); - snd_info_unregister(scb->proc_info); + snd_info_free_entry(scb->proc_info); scb->proc_info = NULL; snd_assert (scb_info != NULL, return); diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile index 2911a8adc1f2..ad947b4c04cc 100644 --- a/sound/pci/cs5535audio/Makefile +++ b/sound/pci/cs5535audio/Makefile @@ -4,7 +4,7 @@ snd-cs5535audio-objs := cs5535audio.o cs5535audio_pcm.o -ifdef CONFIG_PM +ifeq ($(CONFIG_PM),y) snd-cs5535audio-objs += cs5535audio_pm.o endif diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 289bcd99c19c..493ec0816bb3 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -232,7 +232,7 @@ static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state) return 0; } -int snd_emu10k1_resume(struct pci_dev *pci) +static int snd_emu10k1_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct snd_emu10k1 *emu = card->private_data; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 79f24cdf5fbf..be65d4db8e27 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -927,6 +927,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .adc_1361t = 1, /* 24 bit capture instead of 16bit */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102, .driver = "Audigy2", .name = "Audigy 2 EX [1005]", diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index bda8bdf59935..da1610a571b8 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1626,12 +1626,7 @@ static struct pci_driver driver = { // initialization of the module static int __init alsa_card_emu10k1x_init(void) { - int err; - - if ((err = pci_register_driver(&driver)) > 0) - return err; - - return 0; + return pci_register_driver(&driver); } // clean up the module diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index dfba00230d4d..13cd6ce89811 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -35,6 +35,7 @@ #include <linux/mutex.h> #include <sound/core.h> +#include <sound/tlv.h> #include <sound/emu10k1.h> #if 0 /* for testing purposes - digital out -> capture */ @@ -266,6 +267,7 @@ static const u32 treble_table[41][5] = { { 0x37c4448b, 0xa45ef51d, 0x262f3267, 0x081e36dc, 0xfd8f5d14 } }; +/* dB gain = (float) 20 * log10( float(db_table_value) / 0x8000000 ) */ static const u32 db_table[101] = { 0x00000000, 0x01571f82, 0x01674b41, 0x01783a1b, 0x0189f540, 0x019c8651, 0x01aff763, 0x01c45306, 0x01d9a446, 0x01eff6b8, @@ -290,6 +292,9 @@ static const u32 db_table[101] = { 0x7fffffff, }; +/* EMU10k1/EMU10k2 DSP control db gain */ +static DECLARE_TLV_DB_SCALE(snd_emu10k1_db_scale1, -4000, 40, 1); + static const u32 onoff_table[2] = { 0x00000000, 0x00000001 }; @@ -755,6 +760,11 @@ static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu, knew.device = gctl->id.device; knew.subdevice = gctl->id.subdevice; knew.info = snd_emu10k1_gpr_ctl_info; + if (gctl->tlv.p) { + knew.tlv.p = gctl->tlv.p; + knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ; + } knew.get = snd_emu10k1_gpr_ctl_get; knew.put = snd_emu10k1_gpr_ctl_put; memset(nctl, 0, sizeof(*nctl)); @@ -1013,6 +1023,7 @@ snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl, ctl->gpr[0] = gpr + 0; ctl->value[0] = defval; ctl->min = 0; ctl->max = 100; + ctl->tlv.p = snd_emu10k1_db_scale1; ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100; } @@ -1027,6 +1038,7 @@ snd_emu10k1_init_stereo_control(struct snd_emu10k1_fx8010_control_gpr *ctl, ctl->gpr[1] = gpr + 1; ctl->value[1] = defval; ctl->min = 0; ctl->max = 100; + ctl->tlv.p = snd_emu10k1_db_scale1; ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100; } diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 9905651935fb..4e0f95438f47 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -100,6 +100,7 @@ #include <sound/pcm.h> #include <sound/ac97_codec.h> #include <sound/info.h> +#include <sound/tlv.h> #include <sound/emu10k1.h> #include "p16v.h" @@ -784,12 +785,16 @@ static int snd_p16v_capture_channel_put(struct snd_kcontrol *kcontrol, } return change; } +static DECLARE_TLV_DB_SCALE(snd_p16v_db_scale1, -5175, 25, 1); #define P16V_VOL(xname,xreg,xhl) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_p16v_volume_info, \ .get = snd_p16v_volume_get, \ .put = snd_p16v_volume_put, \ + .tlv = { .p = snd_p16v_db_scale1 }, \ .private_value = ((xreg) | ((xhl) << 8)) \ } diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index cc0f34f68185..3ce5a4e7e31f 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -62,6 +62,7 @@ #include <sound/opl3.h> #include <sound/mpu401.h> #include <sound/initval.h> +#include <sound/tlv.h> #include <asm/io.h> @@ -1164,6 +1165,14 @@ static int snd_es1938_reg_read(struct es1938 *chip, unsigned char reg) return snd_es1938_read(chip, reg); } +#define ES1938_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,\ + .name = xname, .index = xindex, \ + .info = snd_es1938_info_single, \ + .get = snd_es1938_get_single, .put = snd_es1938_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ + .tlv = { .p = xtlv } } #define ES1938_SINGLE(xname, xindex, reg, shift, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_es1938_info_single, \ @@ -1217,6 +1226,14 @@ static int snd_es1938_put_single(struct snd_kcontrol *kcontrol, return snd_es1938_reg_bits(chip, reg, mask, val) != val; } +#define ES1938_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,\ + .name = xname, .index = xindex, \ + .info = snd_es1938_info_double, \ + .get = snd_es1938_get_double, .put = snd_es1938_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = xtlv } } #define ES1938_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_es1938_info_double, \ @@ -1297,8 +1314,41 @@ static int snd_es1938_put_double(struct snd_kcontrol *kcontrol, return change; } +static unsigned int db_scale_master[] = { + TLV_DB_RANGE_HEAD(2), + 0, 54, TLV_DB_SCALE_ITEM(-3600, 50, 1), + 54, 63, TLV_DB_SCALE_ITEM(-900, 100, 0), +}; + +static unsigned int db_scale_audio1[] = { + TLV_DB_RANGE_HEAD(2), + 0, 8, TLV_DB_SCALE_ITEM(-3300, 300, 1), + 8, 15, TLV_DB_SCALE_ITEM(-900, 150, 0), +}; + +static unsigned int db_scale_audio2[] = { + TLV_DB_RANGE_HEAD(2), + 0, 8, TLV_DB_SCALE_ITEM(-3450, 300, 1), + 8, 15, TLV_DB_SCALE_ITEM(-1050, 150, 0), +}; + +static unsigned int db_scale_mic[] = { + TLV_DB_RANGE_HEAD(2), + 0, 8, TLV_DB_SCALE_ITEM(-2400, 300, 1), + 8, 15, TLV_DB_SCALE_ITEM(0, 150, 0), +}; + +static unsigned int db_scale_line[] = { + TLV_DB_RANGE_HEAD(2), + 0, 8, TLV_DB_SCALE_ITEM(-3150, 300, 1), + 8, 15, TLV_DB_SCALE_ITEM(-750, 150, 0), +}; + +static DECLARE_TLV_DB_SCALE(db_scale_capture, 0, 150, 0); + static struct snd_kcontrol_new snd_es1938_controls[] = { -ES1938_DOUBLE("Master Playback Volume", 0, 0x60, 0x62, 0, 0, 63, 0), +ES1938_DOUBLE_TLV("Master Playback Volume", 0, 0x60, 0x62, 0, 0, 63, 0, + db_scale_master), ES1938_DOUBLE("Master Playback Switch", 0, 0x60, 0x62, 6, 6, 1, 1), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1309,19 +1359,27 @@ ES1938_DOUBLE("Master Playback Switch", 0, 0x60, 0x62, 6, 6, 1, 1), }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Hardware Master Playback Switch", - .access = SNDRV_CTL_ELEM_ACCESS_READ, .info = snd_es1938_info_hw_switch, .get = snd_es1938_get_hw_switch, + .tlv = { .p = db_scale_master }, }, ES1938_SINGLE("Hardware Volume Split", 0, 0x64, 7, 1, 0), -ES1938_DOUBLE("Line Playback Volume", 0, 0x3e, 0x3e, 4, 0, 15, 0), +ES1938_DOUBLE_TLV("Line Playback Volume", 0, 0x3e, 0x3e, 4, 0, 15, 0, + db_scale_line), ES1938_DOUBLE("CD Playback Volume", 0, 0x38, 0x38, 4, 0, 15, 0), -ES1938_DOUBLE("FM Playback Volume", 0, 0x36, 0x36, 4, 0, 15, 0), -ES1938_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), -ES1938_DOUBLE("Mic Playback Volume", 0, 0x1a, 0x1a, 4, 0, 15, 0), -ES1938_DOUBLE("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0), -ES1938_DOUBLE("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0), +ES1938_DOUBLE_TLV("FM Playback Volume", 0, 0x36, 0x36, 4, 0, 15, 0, + db_scale_mic), +ES1938_DOUBLE_TLV("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0, + db_scale_line), +ES1938_DOUBLE_TLV("Mic Playback Volume", 0, 0x1a, 0x1a, 4, 0, 15, 0, + db_scale_mic), +ES1938_DOUBLE_TLV("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0, + db_scale_line), +ES1938_DOUBLE_TLV("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0, + db_scale_capture), ES1938_SINGLE("PC Speaker Volume", 0, 0x3c, 0, 7, 0), ES1938_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0), ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), @@ -1332,16 +1390,26 @@ ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), .get = snd_es1938_get_mux, .put = snd_es1938_put_mux, }, -ES1938_DOUBLE("Mono Input Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), -ES1938_DOUBLE("PCM Capture Volume", 0, 0x69, 0x69, 4, 0, 15, 0), -ES1938_DOUBLE("Mic Capture Volume", 0, 0x68, 0x68, 4, 0, 15, 0), -ES1938_DOUBLE("Line Capture Volume", 0, 0x6e, 0x6e, 4, 0, 15, 0), -ES1938_DOUBLE("FM Capture Volume", 0, 0x6b, 0x6b, 4, 0, 15, 0), -ES1938_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0), -ES1938_DOUBLE("CD Capture Volume", 0, 0x6a, 0x6a, 4, 0, 15, 0), -ES1938_DOUBLE("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0), -ES1938_DOUBLE("PCM Playback Volume", 0, 0x7c, 0x7c, 4, 0, 15, 0), -ES1938_DOUBLE("PCM Playback Volume", 1, 0x14, 0x14, 4, 0, 15, 0), +ES1938_DOUBLE_TLV("Mono Input Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0, + db_scale_line), +ES1938_DOUBLE_TLV("PCM Capture Volume", 0, 0x69, 0x69, 4, 0, 15, 0, + db_scale_audio2), +ES1938_DOUBLE_TLV("Mic Capture Volume", 0, 0x68, 0x68, 4, 0, 15, 0, + db_scale_mic), +ES1938_DOUBLE_TLV("Line Capture Volume", 0, 0x6e, 0x6e, 4, 0, 15, 0, + db_scale_line), +ES1938_DOUBLE_TLV("FM Capture Volume", 0, 0x6b, 0x6b, 4, 0, 15, 0, + db_scale_mic), +ES1938_DOUBLE_TLV("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0, + db_scale_line), +ES1938_DOUBLE_TLV("CD Capture Volume", 0, 0x6a, 0x6a, 4, 0, 15, 0, + db_scale_line), +ES1938_DOUBLE_TLV("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0, + db_scale_line), +ES1938_DOUBLE_TLV("PCM Playback Volume", 0, 0x7c, 0x7c, 4, 0, 15, 0, + db_scale_audio2), +ES1938_DOUBLE_TLV("PCM Playback Volume", 1, 0x14, 0x14, 4, 0, 15, 0, + db_scale_audio1), ES1938_SINGLE("3D Control - Level", 0, 0x52, 0, 63, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 3c5ab7c2e72d..f3c40385c87d 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1905,7 +1905,7 @@ static void es1968_update_hw_volume(unsigned long private_data) /* Figure out which volume control button was pushed, based on differences from the default register values. */ - x = inb(chip->io_port + 0x1c); + x = inb(chip->io_port + 0x1c) & 0xee; /* Reset the volume control registers. */ outb(0x88, chip->io_port + 0x1c); outb(0x88, chip->io_port + 0x1d); @@ -1921,7 +1921,8 @@ static void es1968_update_hw_volume(unsigned long private_data) /* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */ spin_lock_irqsave(&chip->ac97_lock, flags); val = chip->ac97->regs[AC97_MASTER]; - if (x & 1) { + switch (x) { + case 0x88: /* mute */ val ^= 0x8000; chip->ac97->regs[AC97_MASTER] = val; @@ -1929,26 +1930,31 @@ static void es1968_update_hw_volume(unsigned long private_data) outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX); snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); - } else { - val &= 0x7fff; - if (((x>>1) & 7) > 4) { - /* volume up */ - if ((val & 0xff) > 0) - val--; - if ((val & 0xff00) > 0) - val -= 0x0100; - } else { - /* volume down */ - if ((val & 0xff) < 0x1f) - val++; - if ((val & 0xff00) < 0x1f00) - val += 0x0100; - } + break; + case 0xaa: + /* volume up */ + if ((val & 0x7f) > 0) + val--; + if ((val & 0x7f00) > 0) + val -= 0x0100; + chip->ac97->regs[AC97_MASTER] = val; + outw(val, chip->io_port + ESM_AC97_DATA); + outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX); + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_volume->id); + break; + case 0x66: + /* volume down */ + if ((val & 0x7f) < 0x1f) + val++; + if ((val & 0x7f00) < 0x1f00) + val += 0x0100; chip->ac97->regs[AC97_MASTER] = val; outw(val, chip->io_port + ESM_AC97_DATA); outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX); snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); + break; } spin_unlock_irqrestore(&chip->ac97_lock, flags); } diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 13868c985126..bdfda1997d5b 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -2,6 +2,7 @@ * The driver for the ForteMedia FM801 based soundcards * Copyright (c) by Jaroslav Kysela <perex@suse.cz> * + * Support FM only card by Andy Shevchenko <andy@smile.org.ua> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -28,6 +29,7 @@ #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/pcm.h> +#include <sound/tlv.h> #include <sound/ac97_codec.h> #include <sound/mpu401.h> #include <sound/opl3.h> @@ -54,6 +56,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * * 1 = MediaForte 256-PCS * 2 = MediaForte 256-PCPR * 3 = MediaForte 64-PCR + * 16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card * High 16-bits are video (radio) device number + 1 */ static int tea575x_tuner[SNDRV_CARDS]; @@ -158,6 +161,7 @@ struct fm801 { unsigned int multichannel: 1, /* multichannel support */ secondary: 1; /* secondary codec */ unsigned char secondary_addr; /* address of the secondary codec */ + unsigned int tea575x_tuner; /* tuner flags */ unsigned short ply_ctrl; /* playback control */ unsigned short cap_ctrl; /* capture control */ @@ -318,10 +322,8 @@ static unsigned int channels[] = { 2, 4, 6 }; -#define CHANNELS sizeof(channels) / sizeof(channels[0]) - static struct snd_pcm_hw_constraint_list hw_constraints_channels = { - .count = CHANNELS, + .count = ARRAY_SIZE(channels), .list = channels, .mask = 0, }; @@ -1052,6 +1054,13 @@ static int snd_fm801_put_single(struct snd_kcontrol *kcontrol, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_fm801_info_double, \ .get = snd_fm801_get_double, .put = snd_fm801_put_double, \ .private_value = reg | (shift_left << 8) | (shift_right << 12) | (mask << 16) | (invert << 24) } +#define FM801_DOUBLE_TLV(xname, reg, shift_left, shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, .info = snd_fm801_info_double, \ + .get = snd_fm801_get_double, .put = snd_fm801_put_double, \ + .private_value = reg | (shift_left << 8) | (shift_right << 12) | (mask << 16) | (invert << 24), \ + .tlv = { .p = (xtlv) } } static int snd_fm801_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1148,14 +1157,19 @@ static int snd_fm801_put_mux(struct snd_kcontrol *kcontrol, return snd_fm801_update_bits(chip, FM801_REC_SRC, 7, val); } +static DECLARE_TLV_DB_SCALE(db_scale_dsp, -3450, 150, 0); + #define FM801_CONTROLS ARRAY_SIZE(snd_fm801_controls) static struct snd_kcontrol_new snd_fm801_controls[] __devinitdata = { -FM801_DOUBLE("Wave Playback Volume", FM801_PCM_VOL, 0, 8, 31, 1), +FM801_DOUBLE_TLV("Wave Playback Volume", FM801_PCM_VOL, 0, 8, 31, 1, + db_scale_dsp), FM801_SINGLE("Wave Playback Switch", FM801_PCM_VOL, 15, 1, 1), -FM801_DOUBLE("I2S Playback Volume", FM801_I2S_VOL, 0, 8, 31, 1), +FM801_DOUBLE_TLV("I2S Playback Volume", FM801_I2S_VOL, 0, 8, 31, 1, + db_scale_dsp), FM801_SINGLE("I2S Playback Switch", FM801_I2S_VOL, 15, 1, 1), -FM801_DOUBLE("FM Playback Volume", FM801_FM_VOL, 0, 8, 31, 1), +FM801_DOUBLE_TLV("FM Playback Volume", FM801_FM_VOL, 0, 8, 31, 1, + db_scale_dsp), FM801_SINGLE("FM Playback Switch", FM801_FM_VOL, 15, 1, 1), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1253,6 +1267,9 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) int id; unsigned short cmdw; + if (chip->tea575x_tuner & 0x0010) + goto __ac97_ok; + /* codec cold reset + AC'97 warm reset */ outw((1<<5) | (1<<6), FM801_REG(chip, CODEC_CTRL)); inw(FM801_REG(chip, CODEC_CTRL)); /* flush posting data */ @@ -1290,6 +1307,8 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) wait_for_codec(chip, 0, AC97_VENDOR_ID1, msecs_to_jiffies(750)); } + __ac97_ok: + /* init volume */ outw(0x0808, FM801_REG(chip, PCM_VOL)); outw(0x9f1f, FM801_REG(chip, FM_VOL)); @@ -1298,9 +1317,12 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) /* I2S control - I2S mode */ outw(0x0003, FM801_REG(chip, I2S_MODE)); - /* interrupt setup - unmask MPU, PLAYBACK & CAPTURE */ + /* interrupt setup */ cmdw = inw(FM801_REG(chip, IRQ_MASK)); - cmdw &= ~0x0083; + if (chip->irq < 0) + cmdw |= 0x00c3; /* mask everything, no PCM nor MPU */ + else + cmdw &= ~0x0083; /* unmask MPU, PLAYBACK & CAPTURE */ outw(cmdw, FM801_REG(chip, IRQ_MASK)); /* interrupt clear */ @@ -1365,20 +1387,23 @@ static int __devinit snd_fm801_create(struct snd_card *card, chip->card = card; chip->pci = pci; chip->irq = -1; + chip->tea575x_tuner = tea575x_tuner; if ((err = pci_request_regions(pci, "FM801")) < 0) { kfree(chip); pci_disable_device(pci); return err; } chip->port = pci_resource_start(pci, 0); - if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_DISABLED|IRQF_SHARED, - "FM801", chip)) { - snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq); - snd_fm801_free(chip); - return -EBUSY; + if ((tea575x_tuner & 0x0010) == 0) { + if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_DISABLED|IRQF_SHARED, + "FM801", chip)) { + snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq); + snd_fm801_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + pci_set_master(pci); } - chip->irq = pci->irq; - pci_set_master(pci); pci_read_config_byte(pci, PCI_REVISION_ID, &rev); if (rev >= 0xb1) /* FM801-AU */ @@ -1394,12 +1419,12 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO - if (tea575x_tuner > 0 && (tea575x_tuner & 0xffff) < 4) { + if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) { chip->tea.dev_nr = tea575x_tuner >> 16; chip->tea.card = card; chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0xffff) - 1]; + chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1]; snd_tea575x_init(&chip->tea); } #endif @@ -1439,6 +1464,9 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->port, chip->irq); + if (tea575x_tuner[dev] & 0x0010) + goto __fm801_tuner_only; + if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) { snd_card_free(card); return err; @@ -1465,6 +1493,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, return err; } + __fm801_tuner_only: if ((err = snd_card_register(card)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 23201f3eeb12..9c3d7ac08068 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -29,6 +29,7 @@ #include <sound/core.h> #include "hda_codec.h" #include <sound/asoundef.h> +#include <sound/tlv.h> #include <sound/initval.h> #include "hda_local.h" @@ -50,8 +51,10 @@ struct hda_vendor_id { /* codec vendor labels */ static struct hda_vendor_id hda_vendor_ids[] = { { 0x10ec, "Realtek" }, + { 0x1057, "Motorola" }, { 0x11d4, "Analog Devices" }, { 0x13f6, "C-Media" }, + { 0x14f1, "Conexant" }, { 0x434d, "C-Media" }, { 0x8384, "SigmaTel" }, {} /* terminator */ @@ -841,6 +844,31 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } +int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *_tlv) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = get_amp_nid(kcontrol); + int dir = get_amp_direction(kcontrol); + u32 caps, val1, val2; + + if (size < 4 * sizeof(unsigned int)) + return -ENOMEM; + caps = query_amp_caps(codec, nid, dir); + val2 = (((caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT) + 1) * 25; + val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); + val1 = ((int)val1) * ((int)val2); + if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) + return -EFAULT; + if (put_user(2 * sizeof(unsigned int), _tlv + 1)) + return -EFAULT; + if (put_user(val1, _tlv + 2)) + return -EFAULT; + if (put_user(val2, _tlv + 3)) + return -EFAULT; + return 0; +} + /* switch */ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1477,10 +1505,10 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, formats |= SNDRV_PCM_FMTBIT_S32_LE; if (val & AC_SUPPCM_BITS_32) bps = 32; - else if (val & AC_SUPPCM_BITS_20) - bps = 20; else if (val & AC_SUPPCM_BITS_24) bps = 24; + else if (val & AC_SUPPCM_BITS_20) + bps = 20; } } else if (streams == AC_SUPFMT_FLOAT32) { /* should be exclusive */ @@ -1916,7 +1944,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_o /* front */ snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); - if (mout->hp_nid) + if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT]) /* headphone out will just decode front left/right (stereo) */ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); /* extra outputs copied from front */ @@ -1984,7 +2012,7 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) * in the order of front, rear, CLFE, side, ... * * If more extra outputs (speaker and headphone) are found, the pins are - * assisnged to hp_pin and speaker_pins[], respectively. If no line-out jack + * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack * is detected, one of speaker of HP pins is assigned as the primary * output, i.e. to line_out_pins[0]. So, line_outs is always positive * if any analog output exists. @@ -2046,14 +2074,26 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c cfg->speaker_outs++; break; case AC_JACK_HP_OUT: - cfg->hp_pin = nid; + if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins)) + continue; + cfg->hp_pins[cfg->hp_outs] = nid; + cfg->hp_outs++; break; - case AC_JACK_MIC_IN: - if (loc == AC_JACK_LOC_FRONT) - cfg->input_pins[AUTO_PIN_FRONT_MIC] = nid; - else - cfg->input_pins[AUTO_PIN_MIC] = nid; + case AC_JACK_MIC_IN: { + int preferred, alt; + if (loc == AC_JACK_LOC_FRONT) { + preferred = AUTO_PIN_FRONT_MIC; + alt = AUTO_PIN_MIC; + } else { + preferred = AUTO_PIN_MIC; + alt = AUTO_PIN_FRONT_MIC; + } + if (!cfg->input_pins[preferred]) + cfg->input_pins[preferred] = nid; + else if (!cfg->input_pins[alt]) + cfg->input_pins[alt] = nid; break; + } case AC_JACK_LINE_IN: if (loc == AC_JACK_LOC_FRONT) cfg->input_pins[AUTO_PIN_FRONT_LINE] = nid; @@ -2119,8 +2159,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c cfg->speaker_outs, cfg->speaker_pins[0], cfg->speaker_pins[1], cfg->speaker_pins[2], cfg->speaker_pins[3], cfg->speaker_pins[4]); - snd_printd(" hp=0x%x, dig_out=0x%x, din_in=0x%x\n", - cfg->hp_pin, cfg->dig_out_pin, cfg->dig_in_pin); + snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->hp_outs, cfg->hp_pins[0], + cfg->hp_pins[1], cfg->hp_pins[2], + cfg->hp_pins[3], cfg->hp_pins[4]); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], @@ -2141,10 +2183,12 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c sizeof(cfg->speaker_pins)); cfg->speaker_outs = 0; memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); - } else if (cfg->hp_pin) { - cfg->line_outs = 1; - cfg->line_out_pins[0] = cfg->hp_pin; - cfg->hp_pin = 0; + } else if (cfg->hp_outs) { + cfg->line_outs = cfg->hp_outs; + memcpy(cfg->line_out_pins, cfg->hp_pins, + sizeof(cfg->hp_pins)); + cfg->hp_outs = 0; + memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); } } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 40520e9d5a4b..c12bc4e8840f 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -479,7 +479,7 @@ struct hda_codec_ops { struct hda_amp_info { u32 key; /* hash key */ u32 amp_caps; /* amp capabilities */ - u16 vol[2]; /* current volume & mute*/ + u16 vol[2]; /* current volume & mute */ u16 status; /* update flag */ u16 next; /* next link */ }; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 85ad164ada59..97e9af130b71 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -46,11 +46,18 @@ struct hda_gnode { }; /* patch-specific record */ + +#define MAX_PCM_VOLS 2 +struct pcm_vol { + struct hda_gnode *node; /* Node for PCM volume */ + unsigned int index; /* connection of PCM volume */ +}; + struct hda_gspec { struct hda_gnode *dac_node[2]; /* DAC node */ struct hda_gnode *out_pin_node[2]; /* Output pin (Line-Out) node */ - struct hda_gnode *pcm_vol_node[2]; /* Node for PCM volume */ - unsigned int pcm_vol_index[2]; /* connection of PCM volume */ + struct pcm_vol pcm_vol[MAX_PCM_VOLS]; /* PCM volumes */ + unsigned int pcm_vol_nodes; /* number of PCM volumes */ struct hda_gnode *adc_node; /* ADC node */ struct hda_gnode *cap_vol_node; /* Node for capture volume */ @@ -285,9 +292,11 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, return node == spec->dac_node[dac_idx]; } spec->dac_node[dac_idx] = node; - if (node->wid_caps & AC_WCAP_OUT_AMP) { - spec->pcm_vol_node[dac_idx] = node; - spec->pcm_vol_index[dac_idx] = 0; + if ((node->wid_caps & AC_WCAP_OUT_AMP) && + spec->pcm_vol_nodes < MAX_PCM_VOLS) { + spec->pcm_vol[spec->pcm_vol_nodes].node = node; + spec->pcm_vol[spec->pcm_vol_nodes].index = 0; + spec->pcm_vol_nodes++; } return 1; /* found */ } @@ -307,13 +316,16 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, select_input_connection(codec, node, i); unmute_input(codec, node, i); unmute_output(codec, node); - if (! spec->pcm_vol_node[dac_idx]) { - if (node->wid_caps & AC_WCAP_IN_AMP) { - spec->pcm_vol_node[dac_idx] = node; - spec->pcm_vol_index[dac_idx] = i; - } else if (node->wid_caps & AC_WCAP_OUT_AMP) { - spec->pcm_vol_node[dac_idx] = node; - spec->pcm_vol_index[dac_idx] = 0; + if (spec->dac_node[dac_idx] && + spec->pcm_vol_nodes < MAX_PCM_VOLS && + !(spec->dac_node[dac_idx]->wid_caps & + AC_WCAP_OUT_AMP)) { + if ((node->wid_caps & AC_WCAP_IN_AMP) || + (node->wid_caps & AC_WCAP_OUT_AMP)) { + int n = spec->pcm_vol_nodes; + spec->pcm_vol[n].node = node; + spec->pcm_vol[n].index = i; + spec->pcm_vol_nodes++; } } return 1; @@ -370,7 +382,9 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec, /* set PIN-Out enable */ snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + AC_PINCTL_OUT_EN | + ((node->pin_caps & AC_PINCAP_HP_DRV) ? + AC_PINCTL_HP_EN : 0)); return node; } } @@ -461,14 +475,19 @@ static const char *get_input_type(struct hda_gnode *node, unsigned int *pinctl) return "Front Line"; return "Line"; case AC_JACK_CD: +#if 0 if (pinctl) *pinctl |= AC_PINCTL_VREF_GRD; +#endif return "CD"; case AC_JACK_AUX: if ((location & 0x0f) == AC_JACK_LOC_FRONT) return "Front Aux"; return "Aux"; case AC_JACK_MIC_IN: + if (node->pin_caps & + (AC_PINCAP_VREF_80 << AC_PINCAP_VREF_SHIFT)) + *pinctl |= AC_PINCTL_VREF_80; if ((location & 0x0f) == AC_JACK_LOC_FRONT) return "Front Mic"; return "Mic"; @@ -556,6 +575,29 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, return 1; /* found */ } +/* add a capture source element */ +static void add_cap_src(struct hda_gspec *spec, int idx) +{ + struct hda_input_mux_item *csrc; + char *buf; + int num, ocap; + + num = spec->input_mux.num_items; + csrc = &spec->input_mux.items[num]; + buf = spec->cap_labels[num]; + for (ocap = 0; ocap < num; ocap++) { + if (! strcmp(buf, spec->cap_labels[ocap])) { + /* same label already exists, + * put the index number to be unique + */ + sprintf(buf, "%s %d", spec->cap_labels[ocap], num); + break; + } + } + csrc->index = idx; + spec->input_mux.num_items++; +} + /* * parse input */ @@ -576,28 +618,26 @@ static int parse_input_path(struct hda_codec *codec, struct hda_gnode *adc_node) * if it reaches to a proper input PIN, add the path as the * input path. */ + /* first, check the direct connections to PIN widgets */ for (i = 0; i < adc_node->nconns; i++) { node = hda_get_node(spec, adc_node->conn_list[i]); - if (! node) - continue; - err = parse_adc_sub_nodes(codec, spec, node); - if (err < 0) - return err; - else if (err > 0) { - struct hda_input_mux_item *csrc = &spec->input_mux.items[spec->input_mux.num_items]; - char *buf = spec->cap_labels[spec->input_mux.num_items]; - int ocap; - for (ocap = 0; ocap < spec->input_mux.num_items; ocap++) { - if (! strcmp(buf, spec->cap_labels[ocap])) { - /* same label already exists, - * put the index number to be unique - */ - sprintf(buf, "%s %d", spec->cap_labels[ocap], - spec->input_mux.num_items); - } - } - csrc->index = i; - spec->input_mux.num_items++; + if (node && node->type == AC_WID_PIN) { + err = parse_adc_sub_nodes(codec, spec, node); + if (err < 0) + return err; + else if (err > 0) + add_cap_src(spec, i); + } + } + /* ... then check the rests, more complicated connections */ + for (i = 0; i < adc_node->nconns; i++) { + node = hda_get_node(spec, adc_node->conn_list[i]); + if (node && node->type != AC_WID_PIN) { + err = parse_adc_sub_nodes(codec, spec, node); + if (err < 0) + return err; + else if (err > 0) + add_cap_src(spec, i); } } @@ -647,9 +687,6 @@ static int parse_input(struct hda_codec *codec) /* * create mixer controls if possible */ -#define DIR_OUT 0x1 -#define DIR_IN 0x2 - static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, unsigned int index, const char *type, const char *dir_sfx) { @@ -722,49 +759,97 @@ static int check_existing_control(struct hda_codec *codec, const char *type, con /* * build output mixer controls */ -static int build_output_controls(struct hda_codec *codec) +static int create_output_mixers(struct hda_codec *codec, const char **names) { struct hda_gspec *spec = codec->spec; - static const char *types[2] = { "Master", "Headphone" }; int i, err; - for (i = 0; i < 2 && spec->pcm_vol_node[i]; i++) { - err = create_mixer(codec, spec->pcm_vol_node[i], - spec->pcm_vol_index[i], - types[i], "Playback"); + for (i = 0; i < spec->pcm_vol_nodes; i++) { + err = create_mixer(codec, spec->pcm_vol[i].node, + spec->pcm_vol[i].index, + names[i], "Playback"); if (err < 0) return err; } return 0; } +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" }; + + switch (spec->pcm_vol_nodes) { + case 1: + return create_mixer(codec, spec->pcm_vol[0].node, + spec->pcm_vol[0].index, + "Master", "Playback"); + case 2: + if (defcfg_type(spec->out_pin_node[0]) == AC_JACK_SPEAKER) + return create_output_mixers(codec, types_speaker); + else + return create_output_mixers(codec, types_line); + } + return 0; +} + /* create capture volume/switch */ static int build_input_controls(struct hda_codec *codec) { struct hda_gspec *spec = codec->spec; struct hda_gnode *adc_node = spec->adc_node; - int err; - - if (! adc_node) + int i, err; + static struct snd_kcontrol_new cap_sel = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = capture_source_info, + .get = capture_source_get, + .put = capture_source_put, + }; + + if (! adc_node || ! spec->input_mux.num_items) return 0; /* not found */ + spec->cur_cap_src = 0; + select_input_connection(codec, adc_node, + spec->input_mux.items[0].index); + /* create capture volume and switch controls if the ADC has an amp */ - err = create_mixer(codec, adc_node, 0, NULL, "Capture"); + /* do we have only a single item? */ + if (spec->input_mux.num_items == 1) { + err = create_mixer(codec, adc_node, + spec->input_mux.items[0].index, + NULL, "Capture"); + if (err < 0) + return err; + return 0; + } /* create input MUX if multiple sources are available */ - if (spec->input_mux.num_items > 1) { - static struct snd_kcontrol_new cap_sel = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = capture_source_info, - .get = capture_source_get, - .put = capture_source_put, - }; - if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&cap_sel, codec))) < 0) + if ((err = snd_ctl_add(codec->bus->card, + snd_ctl_new1(&cap_sel, codec))) < 0) + return err; + + /* no volume control? */ + if (! (adc_node->wid_caps & AC_WCAP_IN_AMP) || + ! (adc_node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) + return 0; + + for (i = 0; i < spec->input_mux.num_items; i++) { + struct snd_kcontrol_new knew; + char name[32]; + sprintf(name, "%s Capture Volume", + spec->input_mux.items[i].label); + knew = (struct snd_kcontrol_new) + HDA_CODEC_VOLUME(name, adc_node->nid, + spec->input_mux.items[i].index, + HDA_INPUT); + if ((err = snd_ctl_add(codec->bus->card, + snd_ctl_new1(&knew, codec))) < 0) return err; - spec->cur_cap_src = 0; - select_input_connection(codec, adc_node, spec->input_mux.items[0].index); } + return 0; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79d63c99f092..e9d4cb4d07e1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -55,6 +55,7 @@ static char *model; static int position_fix; static int probe_mask = -1; static int single_cmd; +static int disable_msi; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -68,6 +69,8 @@ module_param(probe_mask, int, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); module_param(single_cmd, bool, 0444); MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs (for debugging only)."); +module_param(disable_msi, int, 0); +MODULE_PARM_DESC(disable_msi, "Disable Message Signaled Interrupt (MSI)"); /* just for backward compatibility */ @@ -252,7 +255,7 @@ enum { struct azx_dev { u32 *bdl; /* virtual address of the BDL */ dma_addr_t bdl_addr; /* physical address of the BDL */ - volatile u32 *posbuf; /* position buffer pointer */ + u32 *posbuf; /* position buffer pointer */ unsigned int bufsize; /* size of the play buffer in bytes */ unsigned int fragsize; /* size of each period in bytes */ @@ -271,8 +274,8 @@ struct azx_dev { /* for sanity check of position buffer */ unsigned int period_intr; - unsigned int opened: 1; - unsigned int running: 1; + unsigned int opened :1; + unsigned int running :1; }; /* CORB/RIRB */ @@ -330,8 +333,9 @@ struct azx { /* flags */ int position_fix; - unsigned int initialized: 1; - unsigned int single_cmd: 1; + unsigned int initialized :1; + unsigned int single_cmd :1; + unsigned int polling_mode :1; }; /* driver types */ @@ -516,23 +520,36 @@ static void azx_update_rirb(struct azx *chip) static unsigned int azx_rirb_get_response(struct hda_codec *codec) { struct azx *chip = codec->bus->private_data; - int timeout = 50; + unsigned long timeout; - while (chip->rirb.cmds) { - if (! --timeout) { - snd_printk(KERN_ERR - "hda_intel: azx_get_response timeout, " - "switching to single_cmd mode...\n"); - chip->rirb.rp = azx_readb(chip, RIRBWP); - chip->rirb.cmds = 0; - /* switch to single_cmd mode */ - chip->single_cmd = 1; - azx_free_cmd_io(chip); - return -1; + again: + timeout = jiffies + msecs_to_jiffies(1000); + do { + if (chip->polling_mode) { + spin_lock_irq(&chip->reg_lock); + azx_update_rirb(chip); + spin_unlock_irq(&chip->reg_lock); } - msleep(1); + if (! chip->rirb.cmds) + return chip->rirb.res; /* the last value */ + schedule_timeout_interruptible(1); + } while (time_after_eq(timeout, jiffies)); + + if (!chip->polling_mode) { + snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, " + "switching to polling mode...\n"); + chip->polling_mode = 1; + goto again; } - return chip->rirb.res; /* the last value */ + + snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, " + "switching to single_cmd mode...\n"); + chip->rirb.rp = azx_readb(chip, RIRBWP); + chip->rirb.cmds = 0; + /* switch to single_cmd mode */ + chip->single_cmd = 1; + azx_free_cmd_io(chip); + return -1; } /* @@ -642,14 +659,14 @@ static int azx_reset(struct azx *chip) azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); count = 50; - while (! azx_readb(chip, GCTL) && --count) + while (!azx_readb(chip, GCTL) && --count) msleep(1); - /* Brent Chartrand said to wait >= 540us for codecs to intialize */ + /* Brent Chartrand said to wait >= 540us for codecs to initialize */ msleep(1); /* check to see if controller is ready */ - if (! azx_readb(chip, GCTL)) { + if (!azx_readb(chip, GCTL)) { snd_printd("azx_reset: controller not ready!\n"); return -EBUSY; } @@ -658,7 +675,7 @@ static int azx_reset(struct azx *chip) azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UREN); /* detect codecs */ - if (! chip->codec_mask) { + if (!chip->codec_mask) { chip->codec_mask = azx_readw(chip, STATESTS); snd_printdd("codec_mask = 0x%x\n", chip->codec_mask); } @@ -766,7 +783,7 @@ static void azx_init_chip(struct azx *chip) azx_int_enable(chip); /* initialize the codec command I/O */ - if (! chip->single_cmd) + if (!chip->single_cmd) azx_init_cmd_io(chip); /* program the position buffer */ @@ -794,7 +811,7 @@ static void azx_init_chip(struct azx *chip) /* * interrupt handler */ -static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs) +static irqreturn_t azx_interrupt(int irq, void *dev_id, struct pt_regs *regs) { struct azx *chip = dev_id; struct azx_dev *azx_dev; @@ -999,8 +1016,9 @@ static struct snd_pcm_hardware azx_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE /*|*/ - /*SNDRV_PCM_INFO_RESUME*/), + /* No full-resume yet implemented */ + /* SNDRV_PCM_INFO_RESUME |*/ + SNDRV_PCM_INFO_PAUSE), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, @@ -1178,7 +1196,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) if (chip->position_fix == POS_FIX_POSBUF || chip->position_fix == POS_FIX_AUTO) { /* use the position buffer */ - pos = *azx_dev->posbuf; + pos = le32_to_cpu(*azx_dev->posbuf); if (chip->position_fix == POS_FIX_AUTO && azx_dev->period_intr == 1 && ! pos) { printk(KERN_WARNING @@ -1222,7 +1240,12 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, struct snd_pcm *pcm; struct azx_pcm *apcm; - snd_assert(cpcm->stream[0].substreams || cpcm->stream[1].substreams, return -EINVAL); + /* if no substreams are defined for both playback and capture, + * it's just a placeholder. ignore it. + */ + if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) + return 0; + snd_assert(cpcm->name, return -EINVAL); err = snd_pcm_new(chip->card, cpcm->name, pcm_dev, @@ -1248,7 +1271,8 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, snd_dma_pci_data(chip->pci), 1024 * 64, 1024 * 128); chip->pcm[pcm_dev] = pcm; - chip->pcm_devs = pcm_dev + 1; + if (chip->pcm_devs < pcm_dev + 1) + chip->pcm_devs = pcm_dev + 1; return 0; } @@ -1326,7 +1350,7 @@ static int __devinit azx_init_stream(struct azx *chip) struct azx_dev *azx_dev = &chip->azx_dev[i]; azx_dev->bdl = (u32 *)(chip->bdl.area + off); azx_dev->bdl_addr = chip->bdl.addr + off; - azx_dev->posbuf = (volatile u32 *)(chip->posbuf.area + i * 8); + azx_dev->posbuf = (u32 __iomem *)(chip->posbuf.area + i * 8); /* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ @@ -1355,6 +1379,10 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus, state); azx_free_cmd_io(chip); + if (chip->irq >= 0) + free_irq(chip->irq, chip); + if (!disable_msi) + pci_disable_msi(chip->pci); pci_disable_device(pci); pci_save_state(pci); return 0; @@ -1367,6 +1395,12 @@ static int azx_resume(struct pci_dev *pci) pci_restore_state(pci); pci_enable_device(pci); + if (!disable_msi) + pci_enable_msi(pci); + /* FIXME: need proper error handling */ + request_irq(pci->irq, azx_interrupt, IRQF_DISABLED|IRQF_SHARED, + "HDA Intel", chip); + chip->irq = pci->irq; pci_set_master(pci); azx_init_chip(chip); snd_hda_resume(chip->bus); @@ -1398,12 +1432,14 @@ static int azx_free(struct azx *chip) azx_writel(chip, DPLBASE, 0); azx_writel(chip, DPUBASE, 0); - /* wait a little for interrupts to finish */ - msleep(1); + synchronize_irq(chip->irq); } - if (chip->irq >= 0) + if (chip->irq >= 0) { free_irq(chip->irq, (void*)chip); + if (!disable_msi) + pci_disable_msi(chip->pci); + } if (chip->remap_addr) iounmap(chip->remap_addr); @@ -1434,19 +1470,19 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, struct azx **rchip) { struct azx *chip; - int err = 0; + int err; static struct snd_device_ops ops = { .dev_free = azx_dev_free, }; *rchip = NULL; - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; chip = kzalloc(sizeof(*chip), GFP_KERNEL); - - if (NULL == chip) { + if (!chip) { snd_printk(KERN_ERR SFX "cannot allocate chip\n"); pci_disable_device(pci); return -ENOMEM; @@ -1472,13 +1508,14 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } #endif - if ((err = pci_request_regions(pci, "ICH HD audio")) < 0) { + err = pci_request_regions(pci, "ICH HD audio"); + if (err < 0) { kfree(chip); pci_disable_device(pci); return err; } - chip->addr = pci_resource_start(pci,0); + chip->addr = pci_resource_start(pci, 0); chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0)); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR SFX "ioremap error\n"); @@ -1486,6 +1523,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, goto errout; } + if (!disable_msi) + pci_enable_msi(pci); + if (request_irq(pci->irq, azx_interrupt, IRQF_DISABLED|IRQF_SHARED, "HDA Intel", (void*)chip)) { snd_printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq); @@ -1519,7 +1559,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } chip->num_streams = chip->playback_streams + chip->capture_streams; chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); - if (! chip->azx_dev) { + if (!chip->azx_dev) { snd_printk(KERN_ERR "cannot malloc azx_dev\n"); goto errout; } @@ -1550,7 +1590,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->initialized = 1; /* codec detection */ - if (! chip->codec_mask) { + if (!chip->codec_mask) { snd_printk(KERN_ERR SFX "no codecs found!\n"); err = -ENODEV; goto errout; @@ -1577,16 +1617,16 @@ static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id * { struct snd_card *card; struct azx *chip; - int err = 0; + int err; card = snd_card_new(index, id, THIS_MODULE, 0); - if (NULL == card) { + if (!card) { snd_printk(KERN_ERR SFX "Error creating card!\n"); return -ENOMEM; } - if ((err = azx_create(card, pci, pci_id->driver_data, - &chip)) < 0) { + err = azx_create(card, pci, pci_id->driver_data, &chip); + if (err < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 14e8aa2806ed..f9416c36396e 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -30,9 +30,13 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ .info = snd_hda_mixer_amp_volume_info, \ .get = snd_hda_mixer_amp_volume_get, \ .put = snd_hda_mixer_amp_volume_put, \ + .tlv = { .c = snd_hda_mixer_amp_tlv }, \ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) } /* stereo volume with index */ #define HDA_CODEC_VOLUME_IDX(xname, xcidx, nid, xindex, direction) \ @@ -63,6 +67,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv); int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); @@ -224,7 +229,8 @@ struct auto_pin_cfg { hda_nid_t line_out_pins[5]; /* sorted in the order of Front/Surr/CLFE/Side */ int speaker_outs; hda_nid_t speaker_pins[5]; - hda_nid_t hp_pin; + int hp_outs; + hda_nid_t hp_pins[5]; hda_nid_t input_pins[AUTO_PIN_LAST]; hda_nid_t dig_out_pin; hda_nid_t dig_in_pin; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index c2f0fe85bf35..d737f17695a3 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -52,10 +52,9 @@ static void print_amp_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid, int dir) { unsigned int caps; - if (dir == HDA_OUTPUT) - caps = snd_hda_param_read(codec, nid, AC_PAR_AMP_OUT_CAP); - else - caps = snd_hda_param_read(codec, nid, AC_PAR_AMP_IN_CAP); + caps = snd_hda_param_read(codec, nid, + dir == HDA_OUTPUT ? + AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); if (caps == -1 || caps == 0) { snd_iprintf(buffer, "N/A\n"); return; @@ -74,10 +73,7 @@ static void print_amp_vals(struct snd_info_buffer *buffer, unsigned int val; int i; - if (dir == HDA_OUTPUT) - dir = AC_AMP_GET_OUTPUT; - else - dir = AC_AMP_GET_INPUT; + dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT; for (i = 0; i < indices; i++) { snd_iprintf(buffer, " ["); if (stereo) { diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 6823f2bc10b3..511df07fa2a3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -488,9 +488,13 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, .info = ad1986a_pcm_amp_vol_info, .get = ad1986a_pcm_amp_vol_get, .put = ad1986a_pcm_amp_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, .private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT) }, { @@ -637,6 +641,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = ad1986a_laptop_master_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), }, { @@ -791,6 +796,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { .config = AD1986A_3STACK }, /* ASUS A8N-VM CSM */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x81b3, .config = AD1986A_3STACK }, /* ASUS P5RD2-VM / P5GPL-X SE */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x81cb, + .config = AD1986A_3STACK }, /* ASUS M2NPV-VM */ { .modelname = "laptop", .config = AD1986A_LAPTOP }, { .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e, .config = AD1986A_LAPTOP }, /* FSC V2060 */ @@ -803,6 +810,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { .config = AD1986A_LAPTOP_EAPD }, /* Samsung X60 Chane */ { .pci_subvendor = 0x144d, .pci_subdevice = 0xc024, .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */ + { .pci_subvendor = 0x144d, .pci_subdevice = 0xc026, + .config = AD1986A_LAPTOP_EAPD }, /* Samsung X10-T2300 Culesa */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1153, .config = AD1986A_LAPTOP_EAPD }, /* ASUS M9 */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1213, @@ -1626,10 +1635,12 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *spec = codec->spec; - if (spec->need_dac_fix) + int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, + &spec->multiout.max_channels); + if (! err && spec->need_dac_fix) spec->multiout.num_dacs = spec->multiout.max_channels / 2; - return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, &spec->multiout.max_channels); + return err; } /* 6-stack mode */ @@ -2460,7 +2471,7 @@ static void ad1988_auto_init_extra_out(struct hda_codec *codec) pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ ad1988_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ ad1988_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); } @@ -2512,7 +2523,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) (err = ad1988_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pin, + (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pins[0], "Headphone")) < 0 || (err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 18d105263fea..d08d2e399c8f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -79,6 +79,7 @@ enum { ALC262_BASIC, ALC262_FUJITSU, ALC262_HP_BPC, + ALC262_BENQ_ED8, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -89,6 +90,7 @@ enum { ALC660_3ST, ALC861_3ST_DIG, ALC861_6ST_DIG, + ALC861_UNIWILL_M31, ALC861_AUTO, ALC861_MODEL_LAST, }; @@ -97,6 +99,7 @@ enum { enum { ALC882_3ST_DIG, ALC882_6ST_DIG, + ALC882_ARIMA, ALC882_AUTO, ALC882_MODEL_LAST, }; @@ -108,6 +111,7 @@ enum { ALC883_3ST_6ch, ALC883_6ST_DIG, ALC888_DEMO_BOARD, + ALC883_ACER, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -153,6 +157,7 @@ struct alc_spec { /* channel model */ const struct hda_channel_mode *channel_mode; int num_channel_mode; + int need_dac_fix; /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ @@ -190,6 +195,7 @@ struct alc_config_preset { hda_nid_t dig_in_nid; unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; + int need_dac_fix; unsigned int num_mux_defs; const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); @@ -262,9 +268,12 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->multiout.max_channels); + int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, + spec->num_channel_mode, + &spec->multiout.max_channels); + if (! err && spec->need_dac_fix) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; + return err; } /* @@ -544,6 +553,7 @@ static void setup_preset(struct alc_spec *spec, spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; + spec->need_dac_fix = preset->need_dac_fix; spec->multiout.max_channels = spec->channel_mode[0].channels; @@ -1348,6 +1358,10 @@ static struct hda_verb alc880_pin_clevo_init_verbs[] = { }; static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { + /* change to EAPD mode */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, + /* Headphone output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, /* Front output*/ @@ -1782,25 +1796,9 @@ static int alc_build_pcms(struct hda_codec *codec) } } - /* If the use of more than one ADC is requested for the current - * model, configure a second analog capture-only PCM. - */ - if (spec->num_adc_nids > 1) { - codec->num_pcms++; - info++; - info->name = spec->stream_name_analog; - /* No playback stream for second PCM */ - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; - if (spec->stream_analog_capture) { - snd_assert(spec->adc_nids, return -EINVAL); - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1]; - } - } - + /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { - codec->num_pcms++; + codec->num_pcms = 2; info++; info->name = spec->stream_name_digital; if (spec->multiout.dig_out_nid && @@ -1815,6 +1813,24 @@ static int alc_build_pcms(struct hda_codec *codec) } } + /* If the use of more than one ADC is requested for the current + * model, configure a second analog capture-only PCM. + */ + /* Additional Analaog capture for index #2 */ + if (spec->num_adc_nids > 1 && spec->stream_analog_capture && + spec->adc_nids) { + codec->num_pcms = 3; + info++; + info->name = spec->stream_name_analog; + /* No playback stream for second PCM */ + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; + if (spec->stream_analog_capture) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1]; + } + } + return 0; } @@ -2130,7 +2146,10 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x8086, .pci_subdevice = 0xe20f, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe210, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe211, .config = ALC880_3ST }, + { .pci_subvendor = 0x8086, .pci_subdevice = 0xe212, .config = ALC880_3ST }, + { .pci_subvendor = 0x8086, .pci_subdevice = 0xe213, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe214, .config = ALC880_3ST }, + { .pci_subvendor = 0x8086, .pci_subdevice = 0xe234, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe302, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe303, .config = ALC880_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe304, .config = ALC880_3ST }, @@ -2145,6 +2164,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x107b, .pci_subdevice = 0x4040, .config = ALC880_3ST }, { .pci_subvendor = 0x107b, .pci_subdevice = 0x4041, .config = ALC880_3ST }, /* TCL S700 */ + { .modelname = "tcl", .config = ALC880_TCL_S700 }, { .pci_subvendor = 0x19db, .pci_subdevice = 0x4188, .config = ALC880_TCL_S700 }, /* Back 3 jack, front 2 jack (Internal add Aux-In) */ @@ -2156,8 +2176,13 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .modelname = "3stack-digout", .config = ALC880_3ST_DIG }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xe308, .config = ALC880_3ST_DIG }, { .pci_subvendor = 0x1025, .pci_subdevice = 0x0070, .config = ALC880_3ST_DIG }, - /* Clevo m520G NB */ - { .pci_subvendor = 0x1558, .pci_subdevice = 0x0520, .config = ALC880_CLEVO }, + + /* Clevo laptops */ + { .modelname = "clevo", .config = ALC880_CLEVO }, + { .pci_subvendor = 0x1558, .pci_subdevice = 0x0520, + .config = ALC880_CLEVO }, /* Clevo m520G NB */ + { .pci_subvendor = 0x1558, .pci_subdevice = 0x0660, + .config = ALC880_CLEVO }, /* Clevo m665n */ /* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe305, .config = ALC880_3ST_DIG }, @@ -2222,12 +2247,16 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1043, .pci_subdevice = 0x1113, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1173, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1993, .config = ALC880_ASUS }, + { .pci_subvendor = 0x1043, .pci_subdevice = 0x10c2, .config = ALC880_ASUS_DIG }, /* Asus W6A */ { .pci_subvendor = 0x1043, .pci_subdevice = 0x10c3, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1133, .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1123, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1143, .config = ALC880_ASUS }, + { .modelname = "asus-w1v", .config = ALC880_ASUS_W1V }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x10b3, .config = ALC880_ASUS_W1V }, + { .modelname = "asus-dig", .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x8181, .config = ALC880_ASUS_DIG }, /* ASUS P4GPL-X */ + { .modelname = "asus-dig2", .config = ALC880_ASUS_DIG2 }, { .pci_subvendor = 0x1558, .pci_subdevice = 0x5401, .config = ALC880_ASUS_DIG2 }, { .modelname = "uniwill", .config = ALC880_UNIWILL_DIG }, @@ -2243,6 +2272,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .modelname = "lg-lw", .config = ALC880_LG_LW }, { .pci_subvendor = 0x1854, .pci_subdevice = 0x0018, .config = ALC880_LG_LW }, + { .pci_subvendor = 0x1854, .pci_subdevice = 0x0077, .config = ALC880_LG_LW }, #ifdef CONFIG_SND_DEBUG { .modelname = "test", .config = ALC880_TEST }, @@ -2263,6 +2293,7 @@ static struct alc_config_preset alc880_presets[] = { .dac_nids = alc880_dac_nids, .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_3ST_DIG] = { @@ -2273,6 +2304,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_TCL_S700] = { @@ -2365,6 +2397,7 @@ static struct alc_config_preset alc880_presets[] = { .dac_nids = alc880_asus_dac_nids, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_ASUS_DIG] = { @@ -2376,6 +2409,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_ASUS_DIG2] = { @@ -2387,6 +2421,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_ASUS_W1V] = { @@ -2398,6 +2433,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_UNIWILL_DIG] = { @@ -2408,6 +2444,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), .channel_mode = alc880_asus_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_CLEVO] = { @@ -2419,6 +2456,7 @@ static struct alc_config_preset alc880_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, [ALC880_LG] = { @@ -2430,6 +2468,7 @@ static struct alc_config_preset alc880_presets[] = { .dig_out_nid = ALC880_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes), .channel_mode = alc880_lg_ch_modes, + .need_dac_fix = 1, .input_mux = &alc880_lg_capture_source, .unsol_event = alc880_lg_unsol_event, .init_hook = alc880_lg_automute, @@ -2714,7 +2753,7 @@ static void alc880_auto_init_extra_out(struct hda_codec *codec) pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); } @@ -2755,7 +2794,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pin, + (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0], "Headphone")) < 0 || (err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; @@ -3697,7 +3736,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } - nid = cfg->hp_pin; + nid = cfg->hp_pins[0]; if (nid) { err = alc260_add_playback_controls(spec, nid, "Headphone"); if (err < 0) @@ -3767,7 +3806,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); - nid = spec->autocfg.hp_pin; + nid = spec->autocfg.hp_pins[0]; if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); } @@ -3900,7 +3939,8 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .pci_subvendor = 0x152d, .pci_subdevice = 0x0729, .config = ALC260_BASIC }, /* CTL Travel Master U553W */ { .modelname = "hp", .config = ALC260_HP }, - { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP }, + { .modelname = "hp-3013", .config = ALC260_HP_3013 }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP_3013 }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP_3013 }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3013, .config = ALC260_HP_3013 }, @@ -4266,6 +4306,13 @@ static struct hda_verb alc882_init_verbs[] = { { } }; +static struct hda_verb alc882_eapd_verbs[] = { + /* change to EAPD mode */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, + { } +}; + /* * generic initialization of ADC, input mixers and output mixers */ @@ -4397,6 +4444,9 @@ static struct hda_board_config alc882_cfg_tbl[] = { .config = ALC882_6ST_DIG }, /* Foxconn */ { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* ECS to Intel*/ + { .modelname = "arima", .config = ALC882_ARIMA }, + { .pci_subvendor = 0x161f, .pci_subdevice = 0x2054, + .config = ALC882_ARIMA }, /* Arima W820Di1 */ { .modelname = "auto", .config = ALC882_AUTO }, {} }; @@ -4411,6 +4461,7 @@ static struct alc_config_preset alc882_presets[] = { .dig_in_nid = ALC882_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), .channel_mode = alc882_ch_modes, + .need_dac_fix = 1, .input_mux = &alc882_capture_source, }, [ALC882_6ST_DIG] = { @@ -4424,6 +4475,15 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc882_sixstack_modes, .input_mux = &alc882_capture_source, }, + [ALC882_ARIMA] = { + .mixers = { alc882_base_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_init_verbs, alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), + .channel_mode = alc882_sixstack_modes, + .input_mux = &alc882_capture_source, + }, }; @@ -4466,7 +4526,7 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); /* use dac 0 */ } @@ -4999,16 +5059,23 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = { */ static struct hda_board_config alc883_cfg_tbl[] = { { .modelname = "3stack-dig", .config = ALC883_3ST_2ch_DIG }, + { .modelname = "3stack-6ch-dig", .config = ALC883_3ST_6ch_DIG }, + { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, + .config = ALC883_3ST_6ch_DIG }, /* ECS to Intel*/ + { .modelname = "3stack-6ch", .config = ALC883_3ST_6ch }, + { .pci_subvendor = 0x108e, .pci_subdevice = 0x534d, + .config = ALC883_3ST_6ch }, + { .pci_subvendor = 0x8086, .pci_subdevice = 0xd601, + .config = ALC883_3ST_6ch }, /* D102GGC */ { .modelname = "6stack-dig", .config = ALC883_6ST_DIG }, - { .modelname = "6stack-dig-demo", .config = ALC888_DEMO_BOARD }, { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668, .config = ALC883_6ST_DIG }, /* MSI */ { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668, .config = ALC883_6ST_DIG }, /* Foxconn */ - { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, - .config = ALC883_3ST_6ch_DIG }, /* ECS to Intel*/ - { .pci_subvendor = 0x108e, .pci_subdevice = 0x534d, - .config = ALC883_3ST_6ch }, + { .modelname = "6stack-dig-demo", .config = ALC888_DEMO_BOARD }, + { .modelname = "acer", .config = ALC883_ACER }, + { .pci_subvendor = 0x1025, .pci_subdevice = 0/*0x0102*/, + .config = ALC883_ACER }, { .modelname = "auto", .config = ALC883_AUTO }, {} }; @@ -5038,6 +5105,7 @@ static struct alc_config_preset alc883_presets[] = { .dig_in_nid = ALC883_DIGIN_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, [ALC883_3ST_6ch] = { @@ -5049,6 +5117,7 @@ static struct alc_config_preset alc883_presets[] = { .adc_nids = alc883_adc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, [ALC883_6ST_DIG] = { @@ -5077,6 +5146,23 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, + [ALC883_ACER] = { + .mixers = { alc883_base_mixer, + alc883_chmode_mixer }, + /* On TravelMate laptops, GPIO 0 enables the internal speaker + * and the headphone jack. Turn this on and rely on the + * standard mute methods whenever the user wants to turn + * these outputs off. + */ + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + }, }; @@ -5121,7 +5207,7 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ /* use dac 0 */ alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); @@ -5217,8 +5303,10 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - spec->adc_nids = alc883_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + if (! spec->adc_nids && spec->input_mux) { + spec->adc_nids = alc883_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + } codec->patch_ops = alc_patch_ops; if (board_config == ALC883_AUTO) @@ -5481,6 +5569,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = alc262_fujitsu_master_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), }, { @@ -5499,6 +5588,13 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { } /* end */ }; +/* additional init verbs for Benq laptops */ +static struct hda_verb alc262_EAPD_verbs[] = { + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, + {} +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { @@ -5534,7 +5630,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct return err; } } - nid = cfg->hp_pin; + nid = cfg->hp_pins[0]; if (nid) { /* spec->multiout.hp_nid = 2; */ if (nid == 0x16) { @@ -5769,6 +5865,7 @@ static struct hda_board_config alc262_cfg_tbl[] = { { .modelname = "fujitsu", .config = ALC262_FUJITSU }, { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397, .config = ALC262_FUJITSU }, + { .modelname = "hp-bpc", .config = ALC262_HP_BPC }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x208c, .config = ALC262_HP_BPC }, /* xw4400 */ { .pci_subvendor = 0x103c, .pci_subdevice = 0x3014, @@ -5777,6 +5874,9 @@ static struct hda_board_config alc262_cfg_tbl[] = { .config = ALC262_HP_BPC }, /* xw8400 */ { .pci_subvendor = 0x103c, .pci_subdevice = 0x12fe, .config = ALC262_HP_BPC }, /* xw9400 */ + { .modelname = "benq", .config = ALC262_BENQ_ED8 }, + { .pci_subvendor = 0x17ff, .pci_subdevice = 0x0560, + .config = ALC262_BENQ_ED8 }, { .modelname = "auto", .config = ALC262_AUTO }, {} }; @@ -5814,6 +5914,16 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_HP_capture_source, }, + [ALC262_BENQ_ED8] = { + .mixers = { alc262_base_mixer }, + .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + }, }; static int patch_alc262(struct hda_codec *codec) @@ -5942,6 +6052,23 @@ static struct hda_channel_mode alc861_threestack_modes[2] = { { 2, alc861_threestack_ch2_init }, { 6, alc861_threestack_ch6_init }, }; +/* Set mic1 as input and unmute the mixer */ +static struct hda_verb alc861_uniwill_m31_ch2_init[] = { + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/ + { } /* end */ +}; +/* Set mic1 as output and mute mixer */ +static struct hda_verb alc861_uniwill_m31_ch4_init[] = { + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/ + { } /* end */ +}; + +static struct hda_channel_mode alc861_uniwill_m31_modes[2] = { + { 2, alc861_uniwill_m31_ch2_init }, + { 4, alc861_uniwill_m31_ch4_init }, +}; /* patch-ALC861 */ @@ -6020,6 +6147,47 @@ static struct snd_kcontrol_new alc861_3ST_mixer[] = { }, { } /* end */ }; +static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = { + /* output mixer control */ + HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT), + /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */ + + /* Input mixer control */ + /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */ + HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT), + + /* Capture mixer control */ + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .count = 1, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + .private_value = ARRAY_SIZE(alc861_uniwill_m31_modes), + }, + { } /* end */ +}; /* * generic initialization of ADC, input mixers and output mixers @@ -6148,6 +6316,67 @@ static struct hda_verb alc861_threestack_init_verbs[] = { {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, { } }; + +static struct hda_verb alc861_uniwill_m31_init_verbs[] = { + /* + * Unmute ADC0 and set the default input to mic-in + */ + /* port-A for surround (rear panel) */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + /* port-B for mic-in (rear panel) with vref */ + { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-C for line-in (rear panel) */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* port-D for Front */ + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 }, + /* port-E for HP out (front panel) */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, // this has to be set to VREF80 + /* route front PCM to HP */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01 }, + /* port-F for mic-in (front panel) with vref */ + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + /* port-G for CLFE (rear panel) */ + { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + /* port-H for side (rear panel) */ + { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + /* CD-in */ + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, + /* route front mic to ADC1*/ + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* Unmute DAC0~3 & spdif out*/ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Unmute Mixer 14 (mic) 1c (Line in)*/ + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Unmute Stereo Mixer 15 */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, //Output 0~12 step + + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, // hp used DAC 3 (Front) + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + { } +}; + /* * generic initialization of ADC, input mixers and output mixers */ @@ -6401,7 +6630,7 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.hp_pin; + pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, spec->multiout.dac_nids[0]); } @@ -6436,7 +6665,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 || (err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || - (err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pin)) < 0 || + (err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0])) < 0 || (err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; @@ -6477,10 +6706,14 @@ static struct hda_board_config alc861_cfg_tbl[] = { { .modelname = "3stack", .config = ALC861_3ST }, { .pci_subvendor = 0x8086, .pci_subdevice = 0xd600, .config = ALC861_3ST }, + { .modelname = "3stack-660", .config = ALC660_3ST }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x81e7, .config = ALC660_3ST }, { .modelname = "3stack-dig", .config = ALC861_3ST_DIG }, { .modelname = "6stack-dig", .config = ALC861_6ST_DIG }, + { .modelname = "uniwill-m31", .config = ALC861_UNIWILL_M31}, + { .pci_subvendor = 0x1584, .pci_subdevice = 0x9072, + .config = ALC861_UNIWILL_M31 }, { .modelname = "auto", .config = ALC861_AUTO }, {} }; @@ -6493,6 +6726,7 @@ static struct alc_config_preset alc861_presets[] = { .dac_nids = alc861_dac_nids, .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), .channel_mode = alc861_threestack_modes, + .need_dac_fix = 1, .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), .adc_nids = alc861_adc_nids, .input_mux = &alc861_capture_source, @@ -6505,6 +6739,7 @@ static struct alc_config_preset alc861_presets[] = { .dig_out_nid = ALC861_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), .channel_mode = alc861_threestack_modes, + .need_dac_fix = 1, .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), .adc_nids = alc861_adc_nids, .input_mux = &alc861_capture_source, @@ -6528,10 +6763,25 @@ static struct alc_config_preset alc861_presets[] = { .dac_nids = alc660_dac_nids, .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes), .channel_mode = alc861_threestack_modes, + .need_dac_fix = 1, + .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), + .adc_nids = alc861_adc_nids, + .input_mux = &alc861_capture_source, + }, + [ALC861_UNIWILL_M31] = { + .mixers = { alc861_uniwill_m31_mixer }, + .init_verbs = { alc861_uniwill_m31_init_verbs }, + .num_dacs = ARRAY_SIZE(alc861_dac_nids), + .dac_nids = alc861_dac_nids, + .dig_out_nid = ALC861_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes), + .channel_mode = alc861_uniwill_m31_modes, + .need_dac_fix = 1, .num_adc_nids = ARRAY_SIZE(alc861_adc_nids), .adc_nids = alc861_adc_nids, .input_mux = &alc861_capture_source, }, + }; diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 250242cd6c70..76ec3d75fa9e 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -298,6 +298,7 @@ struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 }, + { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 }, {} }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea99083a1024..731b7b97ee71 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -36,15 +36,15 @@ #define NUM_CONTROL_ALLOC 32 #define STAC_HP_EVENT 0x37 -#define STAC_UNSOL_ENABLE (AC_USRSP_EN | STAC_HP_EVENT) #define STAC_REF 0 #define STAC_D945GTP3 1 #define STAC_D945GTP5 2 #define STAC_MACMINI 3 -#define STAC_D965_2112 4 -#define STAC_D965_284B 5 -#define STAC_922X_MODELS 6 /* number of 922x models */ +#define STAC_922X_MODELS 4 /* number of 922x models */ +#define STAC_D965_3ST 4 +#define STAC_D965_5ST 5 +#define STAC_927X_MODELS 6 /* number of 922x models */ struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; @@ -73,6 +73,7 @@ struct sigmatel_spec { hda_nid_t *pin_nids; unsigned int num_pins; unsigned int *pin_configs; + unsigned int *bios_pin_configs; /* codec specific stuff */ struct hda_verb *init; @@ -110,24 +111,10 @@ static hda_nid_t stac922x_adc_nids[2] = { 0x06, 0x07, }; -static hda_nid_t stac9227_adc_nids[2] = { - 0x07, 0x08, -}; - -#if 0 -static hda_nid_t d965_2112_dac_nids[3] = { - 0x02, 0x03, 0x05, -}; -#endif - static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; -static hda_nid_t stac9227_mux_nids[2] = { - 0x15, 0x16, -}; - static hda_nid_t stac927x_adc_nids[3] = { 0x07, 0x08, 0x09 }; @@ -136,8 +123,17 @@ static hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 }; +static hda_nid_t stac9205_adc_nids[2] = { + 0x12, 0x13 +}; + +static hda_nid_t stac9205_mux_nids[2] = { + 0x19, 0x1a +}; + static hda_nid_t stac9200_pin_nids[8] = { - 0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, + 0x08, 0x09, 0x0d, 0x0e, + 0x0f, 0x10, 0x11, 0x12, }; static hda_nid_t stac922x_pin_nids[10] = { @@ -151,6 +147,13 @@ static hda_nid_t stac927x_pin_nids[14] = { 0x14, 0x21, 0x22, 0x23, }; +static hda_nid_t stac9205_pin_nids[12] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x14, 0x16, 0x17, 0x18, + 0x21, 0x22, + +}; + static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -190,25 +193,23 @@ static struct hda_verb stac922x_core_init[] = { {} }; -static struct hda_verb stac9227_core_init[] = { +static struct hda_verb d965_core_init[] = { /* set master volume and direct control */ - { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* unmute node 0x1b */ { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* select node 0x03 as DAC */ + { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, {} }; -static struct hda_verb d965_2112_core_init[] = { +static struct hda_verb stac927x_core_init[] = { /* set master volume and direct control */ - { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute node 0x1b */ - { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* select node 0x03 as DAC */ - { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, {} }; -static struct hda_verb stac927x_core_init[] = { +static struct hda_verb stac9205_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, {} @@ -277,6 +278,21 @@ static snd_kcontrol_new_t stac927x_mixer[] = { { } /* end */ }; +static snd_kcontrol_new_t stac9205_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .count = 1, + .info = stac92xx_mux_enum_info, + .get = stac92xx_mux_enum_get, + .put = stac92xx_mux_enum_put, + }, + HDA_CODEC_VOLUME("InMux Capture Volume", 0x19, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("InVol Capture Volume", 0x1b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("ADCMux Capture Switch", 0x1d, 0x0, HDA_OUTPUT), + { } /* end */ +}; + static int stac92xx_build_controls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -341,38 +357,67 @@ static unsigned int d945gtp5_pin_configs[10] = { 0x02a19320, 0x40000100, }; -static unsigned int d965_2112_pin_configs[10] = { - 0x0221401f, 0x40000100, 0x40000100, 0x01014011, - 0x01a19021, 0x01813024, 0x01452130, 0x40000100, - 0x02a19320, 0x40000100, -}; - static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_REF] = ref922x_pin_configs, [STAC_D945GTP3] = d945gtp3_pin_configs, [STAC_D945GTP5] = d945gtp5_pin_configs, [STAC_MACMINI] = d945gtp5_pin_configs, - [STAC_D965_2112] = d965_2112_pin_configs, }; static struct hda_board_config stac922x_cfg_tbl[] = { + { .modelname = "5stack", .config = STAC_D945GTP5 }, + { .modelname = "3stack", .config = STAC_D945GTP3 }, { .modelname = "ref", .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x2668, /* DFI LanParty */ .config = STAC_REF }, /* SigmaTel reference board */ + /* Intel 945G based systems */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x0101, .config = STAC_D945GTP3 }, /* Intel D945GTP - 3 Stack */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x0202, - .config = STAC_D945GTP3 }, /* Intel D945GNT - 3 Stack, 9221 A1 */ + .config = STAC_D945GTP3 }, /* Intel D945GNT - 3 Stack */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, - .pci_subdevice = 0x0b0b, - .config = STAC_D945GTP3 }, /* Intel D945PSN - 3 Stack, 9221 A1 */ + .pci_subdevice = 0x0606, + .config = STAC_D945GTP3 }, /* Intel D945GTP - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0601, + .config = STAC_D945GTP3 }, /* Intel D945GTP - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0111, + .config = STAC_D945GTP3 }, /* Intel D945GZP - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x1115, + .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x1116, + .config = STAC_D945GTP3 }, /* Intel D945GBO - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x1117, + .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x1118, + .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x1119, + .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x8826, + .config = STAC_D945GTP3 }, /* Intel D945GPM - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x5049, + .config = STAC_D945GTP3 }, /* Intel D945GCZ - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x5055, + .config = STAC_D945GTP3 }, /* Intel D945GCZ - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x5048, + .config = STAC_D945GTP3 }, /* Intel D945GPB - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0110, + .config = STAC_D945GTP3 }, /* Intel D945GLR - 3 Stack */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, - .pci_subdevice = 0x0707, - .config = STAC_D945GTP5 }, /* Intel D945PSV - 5 Stack */ - { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x0404, .config = STAC_D945GTP5 }, /* Intel D945GTP - 5 Stack */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, @@ -384,44 +429,214 @@ static struct hda_board_config stac922x_cfg_tbl[] = { { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x0417, .config = STAC_D945GTP5 }, /* Intel D975XBK - 5 Stack */ + /* Intel 945P based systems */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0b0b, + .config = STAC_D945GTP3 }, /* Intel D945PSN - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0112, + .config = STAC_D945GTP3 }, /* Intel D945PLN - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0d0d, + .config = STAC_D945GTP3 }, /* Intel D945PLM - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0909, + .config = STAC_D945GTP3 }, /* Intel D945PAW - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0505, + .config = STAC_D945GTP3 }, /* Intel D945PLM - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0707, + .config = STAC_D945GTP5 }, /* Intel D945PSV - 5 Stack */ + /* other systems */ { .pci_subvendor = 0x8384, .pci_subdevice = 0x7680, .config = STAC_MACMINI }, /* Apple Mac Mini (early 2006) */ + {} /* terminator */ +}; + +static unsigned int ref927x_pin_configs[14] = { + 0x02214020, 0x02a19080, 0x0181304e, 0x01014010, + 0x01a19040, 0x01011012, 0x01016011, 0x0101201f, + 0x183301f0, 0x18a001f0, 0x18a001f0, 0x01442070, + 0x01c42190, 0x40000100, +}; + +static unsigned int d965_3st_pin_configs[14] = { + 0x0221401f, 0x02a19120, 0x40000100, 0x01014011, + 0x01a19021, 0x01813024, 0x40000100, 0x40000100, + 0x40000100, 0x40000100, 0x40000100, 0x40000100, + 0x40000100, 0x40000100 +}; + +static unsigned int d965_5st_pin_configs[14] = { + 0x02214020, 0x02a19080, 0x0181304e, 0x01014010, + 0x01a19040, 0x01011012, 0x01016011, 0x40000100, + 0x40000100, 0x40000100, 0x40000100, 0x01442070, + 0x40000100, 0x40000100 +}; + +static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { + [STAC_REF] = ref927x_pin_configs, + [STAC_D965_3ST] = d965_3st_pin_configs, + [STAC_D965_5ST] = d965_5st_pin_configs, +}; + +static struct hda_board_config stac927x_cfg_tbl[] = { + { .modelname = "5stack", .config = STAC_D965_5ST }, + { .modelname = "3stack", .config = STAC_D965_3ST }, + { .modelname = "ref", + .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2668, /* DFI LanParty */ + .config = STAC_REF }, /* SigmaTel reference board */ + /* Intel 946 based systems */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x3d01, + .config = STAC_D965_3ST }, /* D946 configuration */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0xa301, + .config = STAC_D965_3ST }, /* Intel D946GZT - 3 stack */ + /* 965 based 3 stack systems */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2116, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2115, + .config = STAC_D965_3ST }, /* Intel DQ965WC - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2114, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2113, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x2112, - .config = STAC_D965_2112 }, + .config = STAC_D965_3ST }, /* Intel DG965MS - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2111, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2110, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2009, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2008, + .config = STAC_D965_3ST }, /* Intel DQ965GF - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2007, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2006, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2005, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2004, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2003, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2002, + .config = STAC_D965_3ST }, /* Intel D965 3Stack config */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2001, + .config = STAC_D965_3ST }, /* Intel DQ965GF - 3 Stack */ + /* 965 based 5 stack systems */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2301, + .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2302, + .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2303, + .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, - .pci_subdevice = 0x284b, - .config = STAC_D965_284B }, + .pci_subdevice = 0x2304, + .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2305, + .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2501, + .config = STAC_D965_5ST }, /* Intel DG965MQ - 5 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2502, + .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2503, + .config = STAC_D965_5ST }, /* Intel DG965 - 5 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x2504, + .config = STAC_D965_5ST }, /* Intel DQ965GF - 5 Stack */ {} /* terminator */ }; -static unsigned int ref927x_pin_configs[14] = { - 0x01813122, 0x01a19021, 0x01014010, 0x01016011, - 0x01012012, 0x01011014, 0x40000100, 0x40000100, - 0x40000100, 0x40000100, 0x40000100, 0x01441030, - 0x01c41030, 0x40000100, +static unsigned int ref9205_pin_configs[12] = { + 0x40000100, 0x40000100, 0x01016011, 0x01014010, + 0x01813122, 0x01a19021, 0x40000100, 0x40000100, + 0x40000100, 0x40000100, 0x01441030, 0x01c41030 }; -static unsigned int *stac927x_brd_tbl[] = { - ref927x_pin_configs, +static unsigned int *stac9205_brd_tbl[] = { + ref9205_pin_configs, }; -static struct hda_board_config stac927x_cfg_tbl[] = { +static struct hda_board_config stac9205_cfg_tbl[] = { { .modelname = "ref", .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x2668, /* DFI LanParty */ .config = STAC_REF }, /* SigmaTel reference board */ + /* Dell laptops have BIOS problem */ + { .pci_subvendor = PCI_VENDOR_ID_DELL, .pci_subdevice = 0x01b5, + .config = STAC_REF }, /* Dell Inspiron 630m */ + { .pci_subvendor = PCI_VENDOR_ID_DELL, .pci_subdevice = 0x01c2, + .config = STAC_REF }, /* Dell Latitude D620 */ + { .pci_subvendor = PCI_VENDOR_ID_DELL, .pci_subdevice = 0x01cb, + .config = STAC_REF }, /* Dell Latitude 120L */ {} /* terminator */ }; +static int stac92xx_save_bios_config_regs(struct hda_codec *codec) +{ + int i; + struct sigmatel_spec *spec = codec->spec; + + if (! spec->bios_pin_configs) { + spec->bios_pin_configs = kcalloc(spec->num_pins, + sizeof(*spec->bios_pin_configs), GFP_KERNEL); + if (! spec->bios_pin_configs) + return -ENOMEM; + } + + for (i = 0; i < spec->num_pins; i++) { + hda_nid_t nid = spec->pin_nids[i]; + unsigned int pin_cfg; + + pin_cfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0x00); + snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n", + nid, pin_cfg); + spec->bios_pin_configs[i] = pin_cfg; + } + + return 0; +} + static void stac92xx_set_config_regs(struct hda_codec *codec) { int i; struct sigmatel_spec *spec = codec->spec; unsigned int pin_cfg; - for (i=0; i < spec->num_pins; i++) { + if (! spec->pin_nids || ! spec->pin_configs) + return; + + for (i = 0; i < spec->num_pins; i++) { snd_hda_codec_write(codec, spec->pin_nids[i], 0, AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, spec->pin_configs[i] & 0x000000ff); @@ -795,11 +1010,29 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, return 0; } +/* create volume control/switch for the given prefx type */ +static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs) +{ + char name[32]; + int err; + + sprintf(name, "%s Playback Volume", pfx); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", pfx); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); + if (err < 0) + return err; + return 0; +} + /* add playback controls from the parsed DAC table */ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, const struct auto_pin_cfg *cfg) { - char name[32]; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; @@ -814,26 +1047,15 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, if (i == 2) { /* Center/LFE */ - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0) - return err; - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) - return err; - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0) + err = create_controls(spec, "Center", nid, 1); + if (err < 0) return err; - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) + err = create_controls(spec, "LFE", nid, 2); + if (err < 0) return err; } else { - sprintf(name, "%s Playback Volume", chname[i]); - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) - return err; - sprintf(name, "%s Playback Switch", chname[i]); - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) + err = create_controls(spec, chname[i], nid, 3); + if (err < 0) return err; } } @@ -849,39 +1071,85 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, return 0; } -/* add playback controls for HP output */ -static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) +static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) { - struct sigmatel_spec *spec = codec->spec; - hda_nid_t pin = cfg->hp_pin; - hda_nid_t nid; - int i, err; - unsigned int wid_caps; + int i; - if (! pin) - return 0; + for (i = 0; i < spec->multiout.num_dacs; i++) { + if (spec->multiout.dac_nids[i] == nid) + return 1; + } + if (spec->multiout.hp_nid == nid) + return 1; + return 0; +} - wid_caps = get_wcaps(codec, pin); - if (wid_caps & AC_WCAP_UNSOL_CAP) - spec->hp_detect = 1; +static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) +{ + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + else if (spec->multiout.num_dacs > 4) { + printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); + return 1; + } else { + spec->multiout.dac_nids[spec->multiout.num_dacs] = nid; + spec->multiout.num_dacs++; + } + return 0; +} - nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - for (i = 0; i < cfg->line_outs; i++) { - if (! spec->multiout.dac_nids[i]) +/* add playback controls for Speaker and HP outputs */ +static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, + struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid; + int i, old_num_dacs, err; + + old_num_dacs = spec->multiout.num_dacs; + for (i = 0; i < cfg->hp_outs; i++) { + unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]); + if (wid_caps & AC_WCAP_UNSOL_CAP) + spec->hp_detect = 1; + nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) continue; - if (spec->multiout.dac_nids[i] == nid) - return 0; + add_spec_dacs(spec, nid); + } + for (i = 0; i < cfg->speaker_outs; i++) { + nid = snd_hda_codec_read(codec, cfg->speaker_pins[0], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) + continue; + add_spec_dacs(spec, nid); } - spec->multiout.hp_nid = nid; - - /* control HP volume/switch on the output mixer amp */ - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Headphone Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) - return err; - if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Headphone Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) - return err; + for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) { + static const char *pfxs[] = { + "Speaker", "External Speaker", "Speaker2", + }; + err = create_controls(spec, pfxs[i - old_num_dacs], + spec->multiout.dac_nids[i], 3); + if (err < 0) + return err; + } + if (spec->multiout.hp_nid) { + const char *pfx; + if (old_num_dacs == spec->multiout.num_dacs) + pfx = "Master"; + else + pfx = "Headphone"; + err = create_controls(spec, pfx, spec->multiout.hp_nid, 3); + if (err < 0) + return err; + } return 0; } @@ -895,23 +1163,28 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const int i, j, k; for (i = 0; i < AUTO_PIN_LAST; i++) { - int index = -1; - if (cfg->input_pins[i]) { - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; - - for (j=0; j<spec->num_muxes; j++) { - int num_cons = snd_hda_get_connections(codec, spec->mux_nids[j], con_lst, HDA_MAX_NUM_INPUTS); - for (k=0; k<num_cons; k++) - if (con_lst[k] == cfg->input_pins[i]) { - index = k; - break; - } - if (index >= 0) - break; - } - imux->items[imux->num_items].index = index; - imux->num_items++; + int index; + + if (!cfg->input_pins[i]) + continue; + index = -1; + for (j = 0; j < spec->num_muxes; j++) { + int num_cons; + num_cons = snd_hda_get_connections(codec, + spec->mux_nids[j], + con_lst, + HDA_MAX_NUM_INPUTS); + for (k = 0; k < num_cons; k++) + if (con_lst[k] == cfg->input_pins[i]) { + index = k; + goto found; + } } + continue; + found: + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = index; + imux->num_items++; } if (imux->num_items == 1) { @@ -944,11 +1217,20 @@ static void stac92xx_auto_init_multi_out(struct hda_codec *codec) static void stac92xx_auto_init_hp_out(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - hda_nid_t pin; + int i; - pin = spec->autocfg.hp_pin; - if (pin) /* connect to front */ - stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + for (i = 0; i < spec->autocfg.hp_outs; i++) { + hda_nid_t pin; + pin = spec->autocfg.hp_pins[i]; + if (pin) /* connect to front */ + stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + } + for (i = 0; i < spec->autocfg.speaker_outs; i++) { + hda_nid_t pin; + pin = spec->autocfg.speaker_pins[i]; + if (pin) /* connect to front */ + stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN); + } } static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in) @@ -994,7 +1276,7 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; - hda_nid_t pin = cfg->hp_pin; + hda_nid_t pin = cfg->hp_pins[0]; unsigned int wid_caps; if (! pin) @@ -1007,6 +1289,57 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec, return 0; } +/* add playback controls for LFE output */ +static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, + struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + int err; + hda_nid_t lfe_pin = 0x0; + int i; + + /* + * search speaker outs and line outs for a mono speaker pin + * with an amp. If one is found, add LFE controls + * for it. + */ + for (i = 0; i < spec->autocfg.speaker_outs && lfe_pin == 0x0; i++) { + hda_nid_t pin = spec->autocfg.speaker_pins[i]; + unsigned long wcaps = get_wcaps(codec, pin); + wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); + if (wcaps == AC_WCAP_OUT_AMP) + /* found a mono speaker with an amp, must be lfe */ + lfe_pin = pin; + } + + /* if speaker_outs is 0, then speakers may be in line_outs */ + if (lfe_pin == 0 && spec->autocfg.speaker_outs == 0) { + for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { + hda_nid_t pin = spec->autocfg.line_out_pins[i]; + unsigned long cfg; + cfg = snd_hda_codec_read(codec, pin, 0, + AC_VERB_GET_CONFIG_DEFAULT, + 0x00); + if (get_defcfg_device(cfg) == AC_JACK_SPEAKER) { + unsigned long wcaps = get_wcaps(codec, pin); + wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); + if (wcaps == AC_WCAP_OUT_AMP) + /* found a mono speaker with an amp, + must be lfe */ + lfe_pin = pin; + } + } + } + + if (lfe_pin) { + err = create_controls(spec, "LFE", lfe_pin, 1); + if (err < 0) + return err; + } + + return 0; +} + static int stac9200_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -1021,6 +1354,9 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) if ((err = stac9200_auto_create_hp_ctls(codec, &spec->autocfg)) < 0) return err; + if ((err = stac9200_auto_create_lfe_ctls(codec, &spec->autocfg)) < 0) + return err; + if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = 0x05; if (spec->autocfg.dig_in_pin) @@ -1073,6 +1409,15 @@ static void stac922x_gpio_mute(struct hda_codec *codec, int pin, int muted) AC_VERB_SET_GPIO_DATA, gpiostate); } +static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, + unsigned int event) +{ + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | event)); +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -1084,9 +1429,10 @@ static int stac92xx_init(struct hda_codec *codec) /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ - snd_hda_codec_write(codec, cfg->hp_pin, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - STAC_UNSOL_ENABLE); + for (i = 0; i < cfg->hp_outs; i++) + enable_pin_detect(codec, cfg->hp_pins[i], + STAC_HP_EVENT); + stac92xx_auto_init_hp_out(codec); /* fake event to set up pins */ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); } else { @@ -1131,6 +1477,9 @@ static void stac92xx_free(struct hda_codec *codec) kfree(spec->kctl_alloc); } + if (spec->bios_pin_configs) + kfree(spec->bios_pin_configs); + kfree(spec); } @@ -1139,6 +1488,8 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, { unsigned int pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); + if (flag == AC_PINCTL_OUT_EN && (pin_ctl & AC_PINCTL_IN_EN)) + return; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_ctl | flag); @@ -1154,33 +1505,57 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, pin_ctl & ~flag); } -static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) +static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) +{ + if (!nid) + return 0; + if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0x00) + & (1 << 31)) + return 1; + return 0; +} + +static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int i, presence; - if ((res >> 26) != STAC_HP_EVENT) - return; - - presence = snd_hda_codec_read(codec, cfg->hp_pin, 0, - AC_VERB_GET_PIN_SENSE, 0x00) >> 31; + presence = 0; + for (i = 0; i < cfg->hp_outs; i++) { + presence = get_pin_presence(codec, cfg->hp_pins[i]); + if (presence) + break; + } if (presence) { /* disable lineouts, enable hp */ for (i = 0; i < cfg->line_outs; i++) stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN); + for (i = 0; i < cfg->speaker_outs; i++) + stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], + AC_PINCTL_OUT_EN); } else { /* enable lineouts, disable hp */ for (i = 0; i < cfg->line_outs; i++) stac92xx_set_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); - stac92xx_reset_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN); + for (i = 0; i < cfg->speaker_outs; i++) + stac92xx_set_pinctl(codec, cfg->speaker_pins[i], + AC_PINCTL_OUT_EN); } } +static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch (res >> 26) { + case STAC_HP_EVENT: + stac92xx_hp_detect(codec, res); + break; + } +} + #ifdef CONFIG_PM static int stac92xx_resume(struct hda_codec *codec) { @@ -1188,6 +1563,7 @@ static int stac92xx_resume(struct hda_codec *codec) int i; stac92xx_init(codec); + stac92xx_set_config_regs(codec); for (i = 0; i < spec->num_mixers; i++) snd_hda_resume_ctls(codec, spec->mixers[i]); if (spec->multiout.dig_out_nid) @@ -1220,12 +1596,18 @@ static int patch_stac9200(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->num_pins = 8; + spec->pin_nids = stac9200_pin_nids; spec->board_config = snd_hda_check_board_config(codec, stac9200_cfg_tbl); - if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n"); - else { - spec->num_pins = 8; - spec->pin_nids = stac9200_pin_nids; + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else { spec->pin_configs = stac9200_brd_tbl[spec->board_config]; stac92xx_set_config_regs(codec); } @@ -1261,13 +1643,19 @@ static int patch_stac922x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->num_pins = 10; + spec->pin_nids = stac922x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, stac922x_cfg_tbl); - if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, " - "using BIOS defaults\n"); - else if (stac922x_brd_tbl[spec->board_config] != NULL) { - spec->num_pins = 10; - spec->pin_nids = stac922x_pin_nids; + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, " + "using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else if (stac922x_brd_tbl[spec->board_config] != NULL) { spec->pin_configs = stac922x_brd_tbl[spec->board_config]; stac92xx_set_config_regs(codec); } @@ -1281,25 +1669,6 @@ static int patch_stac922x(struct hda_codec *codec) spec->multiout.dac_nids = spec->dac_nids; - switch (spec->board_config) { - case STAC_D965_2112: - spec->adc_nids = stac9227_adc_nids; - spec->mux_nids = stac9227_mux_nids; -#if 0 - spec->multiout.dac_nids = d965_2112_dac_nids; - spec->multiout.num_dacs = ARRAY_SIZE(d965_2112_dac_nids); -#endif - spec->init = d965_2112_core_init; - spec->mixer = stac9227_mixer; - break; - case STAC_D965_284B: - spec->adc_nids = stac9227_adc_nids; - spec->mux_nids = stac9227_mux_nids; - spec->init = stac9227_core_init; - spec->mixer = stac9227_mixer; - break; - } - err = stac92xx_parse_auto_config(codec, 0x08, 0x09); if (err < 0) { stac92xx_free(codec); @@ -1324,26 +1693,94 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->num_pins = 14; + spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, stac927x_cfg_tbl); - if (spec->board_config < 0) + if (spec->board_config < 0) { snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC927x, using BIOS defaults\n"); - else { - spec->num_pins = 14; - spec->pin_nids = stac927x_pin_nids; + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else if (stac927x_brd_tbl[spec->board_config] != NULL) { spec->pin_configs = stac927x_brd_tbl[spec->board_config]; stac92xx_set_config_regs(codec); } - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; + switch (spec->board_config) { + case STAC_D965_3ST: + spec->adc_nids = stac927x_adc_nids; + spec->mux_nids = stac927x_mux_nids; + spec->num_muxes = 3; + spec->init = d965_core_init; + spec->mixer = stac9227_mixer; + break; + case STAC_D965_5ST: + spec->adc_nids = stac927x_adc_nids; + spec->mux_nids = stac927x_mux_nids; + spec->num_muxes = 3; + spec->init = d965_core_init; + spec->mixer = stac9227_mixer; + break; + default: + spec->adc_nids = stac927x_adc_nids; + spec->mux_nids = stac927x_mux_nids; + spec->num_muxes = 3; + spec->init = stac927x_core_init; + spec->mixer = stac927x_mixer; + } + + spec->multiout.dac_nids = spec->dac_nids; + + err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +} + +static int patch_stac9205(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->num_pins = 14; + spec->pin_nids = stac9205_pin_nids; + spec->board_config = snd_hda_check_board_config(codec, stac9205_cfg_tbl); + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else { + spec->pin_configs = stac9205_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + spec->adc_nids = stac9205_adc_nids; + spec->mux_nids = stac9205_mux_nids; spec->num_muxes = 3; - spec->init = stac927x_core_init; - spec->mixer = stac927x_mixer; + spec->init = stac9205_core_init; + spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; - err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); + err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (err < 0) { stac92xx_free(codec); return err; @@ -1355,10 +1792,10 @@ static int patch_stac927x(struct hda_codec *codec) } /* - * STAC 7661(?) hack + * STAC9872 hack */ -/* static config for Sony VAIO FE550G */ +/* static config for Sony VAIO FE550G and Sony VAIO AR */ static hda_nid_t vaio_dacs[] = { 0x2 }; #define VAIO_HP_DAC 0x5 static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ }; @@ -1389,6 +1826,23 @@ static struct hda_verb vaio_init[] = { {} }; +static struct hda_verb vaio_ar_init[] = { + {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ + {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ + {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ +/* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ +/* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ + {} +}; + /* bind volumes of both NID 0x02 and 0x05 */ static int vaio_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1434,6 +1888,38 @@ static struct snd_kcontrol_new vaio_mixer[] = { .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = vaio_master_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, + .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = vaio_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + }, + /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .count = 1, + .info = stac92xx_mux_enum_info, + .get = stac92xx_mux_enum_get, + .put = stac92xx_mux_enum_put, + }, + {} +}; + +static struct snd_kcontrol_new vaio_ar_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = vaio_master_vol_put, .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), }, { @@ -1447,6 +1933,8 @@ static struct snd_kcontrol_new vaio_mixer[] = { /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), + /*HDA_CODEC_MUTE("Optical Out Switch", 0x10, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Optical Out Volume", 0x10, 0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -1458,7 +1946,7 @@ static struct snd_kcontrol_new vaio_mixer[] = { {} }; -static struct hda_codec_ops stac7661_patch_ops = { +static struct hda_codec_ops stac9872_patch_ops = { .build_controls = stac92xx_build_controls, .build_pcms = stac92xx_build_pcms, .init = stac92xx_init, @@ -1468,23 +1956,34 @@ static struct hda_codec_ops stac7661_patch_ops = { #endif }; -enum { STAC7661_VAIO }; - -static struct hda_board_config stac7661_cfg_tbl[] = { - { .modelname = "vaio", .config = STAC7661_VAIO }, +enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */ + CXD9872RD_VAIO, + /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */ + STAC9872AK_VAIO, + /* Unknown. id=0x83847661 and subsys=0x104D1200. */ + STAC9872K_VAIO, + /* AR Series. id=0x83847664 and subsys=104D1300 */ + CXD9872AKD_VAIO + }; + +static struct hda_board_config stac9872_cfg_tbl[] = { + { .modelname = "vaio", .config = CXD9872RD_VAIO }, + { .modelname = "vaio-ar", .config = CXD9872AKD_VAIO }, { .pci_subvendor = 0x104d, .pci_subdevice = 0x81e6, - .config = STAC7661_VAIO }, + .config = CXD9872RD_VAIO }, { .pci_subvendor = 0x104d, .pci_subdevice = 0x81ef, - .config = STAC7661_VAIO }, + .config = CXD9872RD_VAIO }, + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81fd, + .config = CXD9872AKD_VAIO }, {} }; -static int patch_stac7661(struct hda_codec *codec) +static int patch_stac9872(struct hda_codec *codec) { struct sigmatel_spec *spec; int board_config; - board_config = snd_hda_check_board_config(codec, stac7661_cfg_tbl); + board_config = snd_hda_check_board_config(codec, stac9872_cfg_tbl); if (board_config < 0) /* unknown config, let generic-parser do its job... */ return snd_hda_parse_generic_codec(codec); @@ -1495,7 +1994,9 @@ static int patch_stac7661(struct hda_codec *codec) codec->spec = spec; switch (board_config) { - case STAC7661_VAIO: + case CXD9872RD_VAIO: + case STAC9872AK_VAIO: + case STAC9872K_VAIO: spec->mixer = vaio_mixer; spec->init = vaio_init; spec->multiout.max_channels = 2; @@ -1507,9 +2008,22 @@ static int patch_stac7661(struct hda_codec *codec) spec->input_mux = &vaio_mux; spec->mux_nids = vaio_mux_nids; break; + + case CXD9872AKD_VAIO: + spec->mixer = vaio_ar_mixer; + spec->init = vaio_ar_init; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); + spec->multiout.dac_nids = vaio_dacs; + spec->multiout.hp_nid = VAIO_HP_DAC; + spec->num_adcs = ARRAY_SIZE(vaio_adcs); + spec->adc_nids = vaio_adcs; + spec->input_mux = &vaio_mux; + spec->mux_nids = vaio_mux_nids; + break; } - codec->patch_ops = stac7661_patch_ops; + codec->patch_ops = stac9872_patch_ops; return 0; } @@ -1525,12 +2039,12 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x }, { .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x }, { .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x }, - { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac922x }, - { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac922x }, - { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac922x }, - { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac922x }, - { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac922x }, - { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac922x }, + { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac927x }, + { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac927x }, + { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac927x }, + { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac927x }, + { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac927x }, + { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac927x }, { .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x }, { .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x }, { .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x }, @@ -1541,6 +2055,20 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847627, .name = "STAC9271D", .patch = patch_stac927x }, { .id = 0x83847628, .name = "STAC9274X5NH", .patch = patch_stac927x }, { .id = 0x83847629, .name = "STAC9274D5NH", .patch = patch_stac927x }, - { .id = 0x83847661, .name = "STAC7661", .patch = patch_stac7661 }, + /* The following does not take into account .id=0x83847661 when subsys = + * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are + * currently not fully supported. + */ + { .id = 0x83847661, .name = "CXD9872RD/K", .patch = patch_stac9872 }, + { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 }, + { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 }, + { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 }, + { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 }, + { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 }, + { .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 }, + { .id = 0x838476a4, .name = "STAC9255", .patch = patch_stac9205 }, + { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, + { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, + { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, {} /* terminator */ }; diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 9492f3d2455b..9e76cebd2d22 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -60,6 +60,7 @@ #include "ice1712.h" #include "envy24ht.h" #include "aureon.h" +#include <sound/tlv.h> /* WM8770 registers */ #define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */ @@ -660,6 +661,12 @@ static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } +static DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); +static DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1); +static DECLARE_TLV_DB_SCALE(db_scale_wm_adc, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(db_scale_ac97_master, -4650, 150, 0); +static DECLARE_TLV_DB_SCALE(db_scale_ac97_gain, -3450, 150, 0); + /* * Logarithmic volume values for WM8770 * Computed as 20 * Log10(255 / x) @@ -1409,10 +1416,13 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = wm_master_vol_info, .get = wm_master_vol_get, - .put = wm_master_vol_put + .put = wm_master_vol_put, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1424,11 +1434,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Front Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (2 << 8) | 0 + .private_value = (2 << 8) | 0, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1440,11 +1453,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Rear Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (2 << 8) | 2 + .private_value = (2 << 8) | 2, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1456,11 +1472,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Center Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (1 << 8) | 4 + .private_value = (1 << 8) | 4, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1472,11 +1491,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "LFE Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (1 << 8) | 5 + .private_value = (1 << 8) | 5, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1488,11 +1510,14 @@ static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Side Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (2 << 8) | 6 + .private_value = (2 << 8) | 6, + .tlv = { .p = db_scale_wm_dac } } }; @@ -1506,10 +1531,13 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "PCM Playback Volume", .info = wm_pcm_vol_info, .get = wm_pcm_vol_get, - .put = wm_pcm_vol_put + .put = wm_pcm_vol_put, + .tlv = { .p = db_scale_wm_pcm } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1520,10 +1548,13 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Capture Volume", .info = wm_adc_vol_info, .get = wm_adc_vol_get, - .put = wm_adc_vol_put + .put = wm_adc_vol_put, + .tlv = { .p = db_scale_wm_adc } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1567,11 +1598,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "AC97 Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_MASTER|AUREON_AC97_STEREO + .private_value = AC97_MASTER|AUREON_AC97_STEREO, + .tlv = { .p = db_scale_ac97_master } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1583,11 +1617,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "CD Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_CD|AUREON_AC97_STEREO + .private_value = AC97_CD|AUREON_AC97_STEREO, + .tlv = { .p = db_scale_ac97_gain } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1599,11 +1636,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Aux Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_AUX|AUREON_AC97_STEREO + .private_value = AC97_AUX|AUREON_AC97_STEREO, + .tlv = { .p = db_scale_ac97_gain } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1615,11 +1655,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Line Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_LINE|AUREON_AC97_STEREO + .private_value = AC97_LINE|AUREON_AC97_STEREO, + .tlv = { .p = db_scale_ac97_gain } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1631,11 +1674,14 @@ static struct snd_kcontrol_new ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Mic Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_MIC + .private_value = AC97_MIC, + .tlv = { .p = db_scale_ac97_gain } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1657,11 +1703,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "AC97 Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_MASTER|AUREON_AC97_STEREO + .private_value = AC97_MASTER|AUREON_AC97_STEREO, + .tlv = { .p = db_scale_ac97_master } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1673,11 +1722,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "CD Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_AUX|AUREON_AC97_STEREO + .private_value = AC97_AUX|AUREON_AC97_STEREO, + .tlv = { .p = db_scale_ac97_gain } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1685,15 +1737,18 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { .info = aureon_ac97_mute_info, .get = aureon_ac97_mute_get, .put = aureon_ac97_mute_put, - .private_value = AC97_CD, + .private_value = AC97_CD }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Phono Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_CD|AUREON_AC97_STEREO + .private_value = AC97_CD|AUREON_AC97_STEREO, + .tlv = { .p = db_scale_ac97_gain } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1705,11 +1760,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Line Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_LINE|AUREON_AC97_STEREO + .private_value = AC97_LINE|AUREON_AC97_STEREO, + .tlv = { .p = db_scale_ac97_gain } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1721,11 +1779,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Mic Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_MIC + .private_value = AC97_MIC, + .tlv = { .p = db_scale_ac97_gain } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1744,11 +1805,14 @@ static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Aux Playback Volume", .info = aureon_ac97_vol_info, .get = aureon_ac97_vol_get, .put = aureon_ac97_vol_put, - .private_value = AC97_VIDEO|AUREON_AC97_STEREO + .private_value = AC97_VIDEO|AUREON_AC97_STEREO, + .tlv = { .p = db_scale_ac97_gain } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index bf20858d9f19..dc69392eafa3 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -62,6 +62,7 @@ #include <sound/cs8427.h> #include <sound/info.h> #include <sound/initval.h> +#include <sound/tlv.h> #include <sound/asoundef.h> @@ -1377,6 +1378,7 @@ static int snd_ice1712_pro_mixer_volume_put(struct snd_kcontrol *kcontrol, struc return change; } +static DECLARE_TLV_DB_SCALE(db_scale_playback, -14400, 150, 0); static struct snd_kcontrol_new snd_ice1712_multi_playback_ctrls[] __devinitdata = { { @@ -1390,12 +1392,15 @@ static struct snd_kcontrol_new snd_ice1712_multi_playback_ctrls[] __devinitdata }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Multi Playback Volume", .info = snd_ice1712_pro_mixer_volume_info, .get = snd_ice1712_pro_mixer_volume_get, .put = snd_ice1712_pro_mixer_volume_put, .private_value = 0, .count = 10, + .tlv = { .p = db_scale_playback } }, }; @@ -1420,11 +1425,14 @@ static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch __devinitd static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_volume __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "H/W Multi Capture Volume", .info = snd_ice1712_pro_mixer_volume_info, .get = snd_ice1712_pro_mixer_volume_get, .put = snd_ice1712_pro_mixer_volume_put, .private_value = 10, + .tlv = { .p = db_scale_playback } }; static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_volume __devinitdata = { @@ -1857,7 +1865,7 @@ static int snd_ice1712_pro_internal_clock_put(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); static unsigned int xrate[13] = { - 8000, 9600, 11025, 12000, 1600, 22050, 24000, + 8000, 9600, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 }; unsigned char oval; @@ -1924,7 +1932,7 @@ static int snd_ice1712_pro_internal_clock_default_get(struct snd_kcontrol *kcont { int val; static unsigned int xrate[13] = { - 8000, 9600, 11025, 12000, 1600, 22050, 24000, + 8000, 9600, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 }; @@ -1941,7 +1949,7 @@ static int snd_ice1712_pro_internal_clock_default_put(struct snd_kcontrol *kcont struct snd_ctl_elem_value *ucontrol) { static unsigned int xrate[13] = { - 8000, 9600, 11025, 12000, 1600, 22050, 24000, + 8000, 9600, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 }; unsigned char oval; diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index 502da1c8b5f7..e08d73f4ff85 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -46,6 +46,7 @@ #include "ice1712.h" #include "envy24ht.h" #include "phase.h" +#include <sound/tlv.h> /* WM8770 registers */ #define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */ @@ -696,6 +697,9 @@ static int phase28_oversampling_put(struct snd_kcontrol *kcontrol, struct snd_ct return 0; } +static DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); +static DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1); + static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -706,10 +710,13 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = wm_master_vol_info, .get = wm_master_vol_get, - .put = wm_master_vol_put + .put = wm_master_vol_put, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -721,11 +728,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Front Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (2 << 8) | 0 + .private_value = (2 << 8) | 0, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -737,11 +747,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Rear Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (2 << 8) | 2 + .private_value = (2 << 8) | 2, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -753,11 +766,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Center Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (1 << 8) | 4 + .private_value = (1 << 8) | 4, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -769,11 +785,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "LFE Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (1 << 8) | 5 + .private_value = (1 << 8) | 5, + .tlv = { .p = db_scale_wm_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -785,11 +804,14 @@ static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Side Playback Volume", .info = wm_vol_info, .get = wm_vol_get, .put = wm_vol_put, - .private_value = (2 << 8) | 6 + .private_value = (2 << 8) | 6, + .tlv = { .p = db_scale_wm_dac } } }; @@ -803,10 +825,13 @@ static struct snd_kcontrol_new wm_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "PCM Playback Volume", .info = wm_pcm_vol_info, .get = wm_pcm_vol_get, - .put = wm_pcm_vol_put + .put = wm_pcm_vol_put, + .tlv = { .p = db_scale_wm_pcm } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 0efcad9260a5..6c74c2d2e7f3 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -31,6 +31,7 @@ #include <sound/core.h> #include <sound/info.h> +#include <sound/tlv.h> #include "ice1712.h" #include "envy24ht.h" @@ -564,6 +565,8 @@ static int pontis_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el return changed; } +static DECLARE_TLV_DB_SCALE(db_scale_volume, -6400, 50, 1); + /* * mixers */ @@ -571,17 +574,23 @@ static int pontis_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el static struct snd_kcontrol_new pontis_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "PCM Playback Volume", .info = wm_dac_vol_info, .get = wm_dac_vol_get, .put = wm_dac_vol_put, + .tlv = { .p = db_scale_volume }, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Capture Volume", .info = wm_adc_vol_info, .get = wm_adc_vol_get, .put = wm_adc_vol_put, + .tlv = { .p = db_scale_volume }, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index fdb5cb8fac97..41b2605daa3a 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -35,6 +35,7 @@ #include "envy24ht.h" #include "prodigy192.h" #include "stac946x.h" +#include <sound/tlv.h> static inline void stac9460_put(struct snd_ice1712 *ice, int reg, unsigned char val) { @@ -356,6 +357,9 @@ static int aureon_oversampling_put(struct snd_kcontrol *kcontrol, struct snd_ctl } #endif +static DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * mixers */ @@ -368,14 +372,18 @@ static struct snd_kcontrol_new stac_controls[] __devinitdata = { .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Volume", .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -387,11 +395,14 @@ static struct snd_kcontrol_new stac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "DAC Volume", .count = 6, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -404,11 +415,14 @@ static struct snd_kcontrol_new stac_controls[] __devinitdata = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "ADC Volume", .count = 1, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, + .tlv = { .p = db_scale_adc } }, #if 0 { diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index fec9440cb310..bf98ea34feb0 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -87,16 +87,33 @@ static void revo_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) * initialize the chips on M-Audio Revolution cards */ -static unsigned int revo71_num_stereo_front[] = {2}; -static char *revo71_channel_names_front[] = {"PCM Playback Volume"}; +#define AK_DAC(xname,xch) { .name = xname, .num_channels = xch } -static unsigned int revo71_num_stereo_surround[] = {1, 1, 2, 2}; -static char *revo71_channel_names_surround[] = {"PCM Center Playback Volume", "PCM LFE Playback Volume", - "PCM Side Playback Volume", "PCM Rear Playback Volume"}; +static struct snd_akm4xxx_dac_channel revo71_front[] = { + AK_DAC("PCM Playback Volume", 2) +}; + +static struct snd_akm4xxx_dac_channel revo71_surround[] = { + AK_DAC("PCM Center Playback Volume", 1), + AK_DAC("PCM LFE Playback Volume", 1), + AK_DAC("PCM Side Playback Volume", 2), + AK_DAC("PCM Rear Playback Volume", 2), +}; -static unsigned int revo51_num_stereo[] = {2, 1, 1, 2}; -static char *revo51_channel_names[] = {"PCM Playback Volume", "PCM Center Playback Volume", - "PCM LFE Playback Volume", "PCM Rear Playback Volume"}; +static struct snd_akm4xxx_dac_channel revo51_dac[] = { + AK_DAC("PCM Playback Volume", 2), + AK_DAC("PCM Center Playback Volume", 1), + AK_DAC("PCM LFE Playback Volume", 1), + AK_DAC("PCM Rear Playback Volume", 2), +}; + +static struct snd_akm4xxx_adc_channel revo51_adc[] = { + { + .name = "PCM Capture Volume", + .switch_name = "PCM Capture Switch", + .num_channels = 2 + }, +}; static struct snd_akm4xxx akm_revo_front __devinitdata = { .type = SND_AK4381, @@ -104,8 +121,7 @@ static struct snd_akm4xxx akm_revo_front __devinitdata = { .ops = { .set_rate_val = revo_set_rate_val }, - .num_stereo = revo71_num_stereo_front, - .channel_names = revo71_channel_names_front + .dac_info = revo71_front, }; static struct snd_ak4xxx_private akm_revo_front_priv __devinitdata = { @@ -127,8 +143,7 @@ static struct snd_akm4xxx akm_revo_surround __devinitdata = { .ops = { .set_rate_val = revo_set_rate_val }, - .num_stereo = revo71_num_stereo_surround, - .channel_names = revo71_channel_names_surround + .dac_info = revo71_surround, }; static struct snd_ak4xxx_private akm_revo_surround_priv __devinitdata = { @@ -149,8 +164,7 @@ static struct snd_akm4xxx akm_revo51 __devinitdata = { .ops = { .set_rate_val = revo_set_rate_val }, - .num_stereo = revo51_num_stereo, - .channel_names = revo51_channel_names + .dac_info = revo51_dac, }; static struct snd_ak4xxx_private akm_revo51_priv __devinitdata = { @@ -159,7 +173,25 @@ static struct snd_ak4xxx_private akm_revo51_priv __devinitdata = { .data_mask = VT1724_REVO_CDOUT, .clk_mask = VT1724_REVO_CCLK, .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2, - .cs_addr = 0, + .cs_addr = VT1724_REVO_CS1 | VT1724_REVO_CS2, + .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2, + .add_flags = VT1724_REVO_CCLK, /* high at init */ + .mask_flags = 0, +}; + +static struct snd_akm4xxx akm_revo51_adc __devinitdata = { + .type = SND_AK5365, + .num_adcs = 2, + .adc_info = revo51_adc, +}; + +static struct snd_ak4xxx_private akm_revo51_adc_priv __devinitdata = { + .caddr = 2, + .cif = 0, + .data_mask = VT1724_REVO_CDOUT, + .clk_mask = VT1724_REVO_CCLK, + .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2, + .cs_addr = VT1724_REVO_CS0 | VT1724_REVO_CS2, .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2, .add_flags = VT1724_REVO_CCLK, /* high at init */ .mask_flags = 0, @@ -202,9 +234,13 @@ static int __devinit revo_init(struct snd_ice1712 *ice) snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, VT1724_REVO_MUTE); break; case VT1724_SUBDEVICE_REVOLUTION51: - ice->akm_codecs = 1; + ice->akm_codecs = 2; if ((err = snd_ice1712_akm4xxx_init(ak, &akm_revo51, &akm_revo51_priv, ice)) < 0) return err; + err = snd_ice1712_akm4xxx_init(ak + 1, &akm_revo51_adc, + &akm_revo51_adc_priv, ice); + if (err < 0) + return err; /* unmute all codecs - needed! */ snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, VT1724_REVO_MUTE); break; diff --git a/sound/pci/ice1712/revo.h b/sound/pci/ice1712/revo.h index dea52ea219df..efbb86ec3289 100644 --- a/sound/pci/ice1712/revo.h +++ b/sound/pci/ice1712/revo.h @@ -42,7 +42,7 @@ extern struct snd_ice1712_card_info snd_vt1724_revo_cards[]; #define VT1724_REVO_CCLK 0x02 #define VT1724_REVO_CDIN 0x04 /* not used */ #define VT1724_REVO_CDOUT 0x08 -#define VT1724_REVO_CS0 0x10 /* not used */ +#define VT1724_REVO_CS0 0x10 /* AK5365 chipselect for Rev. 5.1 */ #define VT1724_REVO_CS1 0x20 /* front AKM4381 chipselect */ #define VT1724_REVO_CS2 0x40 /* surround AKM4355 chipselect */ #define VT1724_REVO_MUTE (1<<22) /* 0 = all mute, 1 = normal operation */ diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 6874263f1681..72dbaedcbdf5 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2251,6 +2251,16 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) /* ACLink on, 2 channels */ cnt = igetdword(chip, ICHREG(GLOB_CNT)); cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK); +#ifdef CONFIG_SND_AC97_POWER_SAVE + /* do cold reset - the full ac97 powerdown may leave the controller + * in a warm state but actually it cannot communicate with the codec. + */ + iputdword(chip, ICHREG(GLOB_CNT), cnt & ~ICH_AC97COLD); + cnt = igetdword(chip, ICHREG(GLOB_CNT)); + udelay(10); + iputdword(chip, ICHREG(GLOB_CNT), cnt | ICH_AC97COLD); + msleep(1); +#else /* finish cold or do warm reset */ cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM; iputdword(chip, ICHREG(GLOB_CNT), cnt); @@ -2265,6 +2275,7 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) return -EIO; __ok: +#endif if (probing) { /* wait for any codec ready status. * Once it becomes ready it should remain ready @@ -2485,7 +2496,7 @@ static int intel8x0_resume(struct pci_dev *pci) card->shortname, chip); chip->irq = pci->irq; synchronize_irq(chip->irq); - snd_intel8x0_chip_init(chip, 1); + snd_intel8x0_chip_init(chip, 0); /* re-initialize mixer stuff */ if (chip->device_type == DEVICE_INTEL_ICH4) { @@ -2615,6 +2626,7 @@ static void __devinit intel8x0_measure_ac97_clock(struct intel8x0 *chip) /* not 48000Hz, tuning the clock.. */ chip->ac97_bus->clock = (chip->ac97_bus->clock * 48000) / pos; printk(KERN_INFO "intel8x0: clocking to %d\n", chip->ac97_bus->clock); + snd_ac97_update_power(chip->ac97[0], AC97_PCM_FRONT_DAC_RATE, 0); } #ifdef CONFIG_PROC_FS diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 91850281f89b..268e2f7241ea 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1045,6 +1045,8 @@ static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state) for (i = 0; i < chip->pcm_devs; i++) snd_pcm_suspend_all(chip->pcm[i]); snd_ac97_suspend(chip->ac97); + if (chip->irq >= 0) + free_irq(chip->irq, chip); pci_disable_device(pci); pci_save_state(pci); return 0; @@ -1058,6 +1060,9 @@ static int intel8x0m_resume(struct pci_dev *pci) pci_restore_state(pci); pci_enable_device(pci); pci_set_master(pci); + request_irq(pci->irq, snd_intel8x0_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, chip); + chip->irq = pci->irq; snd_intel8x0_chip_init(chip, 0); snd_ac97_resume(chip->ac97); diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index cc43ecd67906..216aee5f93e7 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1109,13 +1109,13 @@ static long long snd_mixart_BA0_llseek(struct snd_info_entry *entry, offset = offset & ~3; /* 4 bytes aligned */ switch(orig) { - case 0: /* SEEK_SET */ + case SEEK_SET: file->f_pos = offset; break; - case 1: /* SEEK_CUR */ + case SEEK_CUR: file->f_pos += offset; break; - case 2: /* SEEK_END, offset is negative */ + case SEEK_END: /* offset is negative */ file->f_pos = MIXART_BA0_SIZE + offset; break; default: @@ -1135,13 +1135,13 @@ static long long snd_mixart_BA1_llseek(struct snd_info_entry *entry, offset = offset & ~3; /* 4 bytes aligned */ switch(orig) { - case 0: /* SEEK_SET */ + case SEEK_SET: file->f_pos = offset; break; - case 1: /* SEEK_CUR */ + case SEEK_CUR: file->f_pos += offset; break; - case 2: /* SEEK_END, offset is negative */ + case SEEK_END: /* offset is negative */ file->f_pos = MIXART_BA1_SIZE + offset; break; default: diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index ed47b732c103..13de0f71d4b7 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -31,6 +31,7 @@ #include "mixart_core.h" #include "mixart_hwdep.h" #include <sound/control.h> +#include <sound/tlv.h> #include "mixart_mixer.h" static u32 mixart_analog_level[256] = { @@ -388,12 +389,17 @@ static int mixart_analog_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e return changed; } +static DECLARE_TLV_DB_SCALE(db_scale_analog, -9600, 50, 0); + static struct snd_kcontrol_new mixart_control_analog_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), /* name will be filled later */ .info = mixart_analog_vol_info, .get = mixart_analog_vol_get, .put = mixart_analog_vol_put, + .tlv = { .p = db_scale_analog }, }; /* shared */ @@ -866,14 +872,19 @@ static int mixart_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem return changed; } +static DECLARE_TLV_DB_SCALE(db_scale_digital, -10950, 50, 0); + static struct snd_kcontrol_new snd_mixart_pcm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), /* name will be filled later */ /* count will be filled later */ .info = mixart_digital_vol_info, /* shared */ .get = mixart_pcm_vol_get, .put = mixart_pcm_vol_put, + .tlv = { .p = db_scale_digital }, }; @@ -984,10 +995,13 @@ static int mixart_monitor_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ static struct snd_kcontrol_new mixart_control_monitor_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Monitoring Volume", .info = mixart_digital_vol_info, /* shared */ .get = mixart_monitor_vol_get, .put = mixart_monitor_vol_put, + .tlv = { .p = db_scale_digital }, }; /* diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index 94e63a1e90d9..b133ad9e095e 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -31,6 +31,7 @@ #include "pcxhr_hwdep.h" #include "pcxhr_core.h" #include <sound/control.h> +#include <sound/tlv.h> #include <sound/asoundef.h> #include "pcxhr_mixer.h" @@ -43,6 +44,9 @@ #define PCXHR_ANALOG_PLAYBACK_LEVEL_MAX 128 /* 0.0 dB */ #define PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL 104 /* -24.0 dB ( 0.0 dB - fix level +24.0 dB ) */ +static DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -9600, 50, 0); +static DECLARE_TLV_DB_SCALE(db_scale_analog_playback, -12800, 100, 0); + static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel) { int err, vol; @@ -130,10 +134,13 @@ static int pcxhr_analog_vol_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new pcxhr_control_analog_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), /* name will be filled later */ .info = pcxhr_analog_vol_info, .get = pcxhr_analog_vol_get, .put = pcxhr_analog_vol_put, + /* tlv will be filled later */ }; /* shared */ @@ -188,6 +195,7 @@ static struct snd_kcontrol_new pcxhr_control_output_switch = { #define PCXHR_DIGITAL_LEVEL_MAX 0x1ff /* +18 dB */ #define PCXHR_DIGITAL_ZERO_LEVEL 0x1b7 /* 0 dB */ +static DECLARE_TLV_DB_SCALE(db_scale_digital, -10950, 50, 0); #define MORE_THAN_ONE_STREAM_LEVEL 0x000001 #define VALID_STREAM_PAN_LEVEL_MASK 0x800000 @@ -343,11 +351,14 @@ static int pcxhr_pcm_vol_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_pcxhr_pcm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), /* name will be filled later */ /* count will be filled later */ .info = pcxhr_digital_vol_info, /* shared */ .get = pcxhr_pcm_vol_get, .put = pcxhr_pcm_vol_put, + .tlv = { .p = db_scale_digital }, }; @@ -433,10 +444,13 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new pcxhr_control_monitor_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Monitoring Volume", .info = pcxhr_digital_vol_info, /* shared */ .get = pcxhr_monitor_vol_get, .put = pcxhr_monitor_vol_put, + .tlv = { .p = db_scale_digital }, }; /* @@ -928,6 +942,7 @@ int pcxhr_create_mixer(struct pcxhr_mgr *mgr) temp = pcxhr_control_analog_level; temp.name = "Master Playback Volume"; temp.private_value = 0; /* playback */ + temp.tlv.p = db_scale_analog_playback; if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) return err; /* output mute controls */ @@ -963,6 +978,7 @@ int pcxhr_create_mixer(struct pcxhr_mgr *mgr) temp = pcxhr_control_analog_level; temp.name = "Master Capture Volume"; temp.private_value = 1; /* capture */ + temp.tlv.p = db_scale_analog_capture; if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) return err; diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index f435fcd6dca9..fe210c853442 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -673,9 +673,13 @@ static struct lbuspath lbus_rec_path = { #define FIRMWARE_VERSIONS 1 static union firmware_version firmware_versions[] = { { - .firmware.ASIC = 3,.firmware.CODEC = 2, - .firmware.AUXDSP = 3,.firmware.PROG = 773, - }, + .firmware = { + .ASIC = 3, + .CODEC = 2, + .AUXDSP = 3, + .PROG = 773, + }, + }, }; static u32 atoh(unsigned char *in, unsigned int len) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index e5a52da77b85..d3e07de433b0 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -726,22 +726,36 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) } -static int hdsp_check_for_firmware (struct hdsp *hdsp, int show_err) +#ifdef HDSP_FW_LOADER +static int __devinit hdsp_request_fw_loader(struct hdsp *hdsp); +#endif + +static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand) { - if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) + return 0; if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - snd_printk(KERN_ERR "Hammerfall-DSP: firmware not present.\n"); hdsp->state &= ~HDSP_FirmwareLoaded; - if (! show_err) + if (! load_on_demand) return -EIO; + snd_printk(KERN_ERR "Hammerfall-DSP: firmware not present.\n"); /* try to load firmware */ - if (hdsp->state & HDSP_FirmwareCached) { - if (snd_hdsp_load_firmware_from_cache(hdsp) != 0) - snd_printk(KERN_ERR "Hammerfall-DSP: Firmware loading from cache failed, please upload manually.\n"); - } else { - snd_printk(KERN_ERR "Hammerfall-DSP: No firmware loaded nor cached, please upload firmware.\n"); + if (! (hdsp->state & HDSP_FirmwareCached)) { +#ifdef HDSP_FW_LOADER + if (! hdsp_request_fw_loader(hdsp)) + return 0; +#endif + snd_printk(KERN_ERR + "Hammerfall-DSP: No firmware loaded nor " + "cached, please upload firmware.\n"); + return -EIO; + } + if (snd_hdsp_load_firmware_from_cache(hdsp) != 0) { + snd_printk(KERN_ERR + "Hammerfall-DSP: Firmware loading from " + "cache failed, please upload manually.\n"); + return -EIO; } - return -EIO; } return 0; } @@ -3181,8 +3195,16 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) return; } } else { - snd_iprintf(buffer, "No firmware loaded nor cached, please upload firmware.\n"); - return; + int err = -EINVAL; +#ifdef HDSP_FW_LOADER + err = hdsp_request_fw_loader(hdsp); +#endif + if (err < 0) { + snd_iprintf(buffer, + "No firmware loaded nor cached, " + "please upload firmware.\n"); + return; + } } } @@ -3851,7 +3873,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd) if (hdsp_check_for_iobox (hdsp)) return -EIO; - if (hdsp_check_for_firmware(hdsp, 1)) + if (hdsp_check_for_firmware(hdsp, 0)) /* no auto-loading in trigger */ return -EIO; spin_lock(&hdsp->lock); diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 4930cc6b054d..ebbe12d78d8c 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -40,6 +40,7 @@ #include <sound/core.h> #include <sound/info.h> #include <sound/control.h> +#include <sound/tlv.h> #include <sound/trident.h> #include <sound/asoundef.h> @@ -2627,6 +2628,8 @@ static int snd_trident_vol_control_get(struct snd_kcontrol *kcontrol, return 0; } +static DECLARE_TLV_DB_SCALE(db_scale_gvol, -6375, 25, 0); + static int snd_trident_vol_control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2653,6 +2656,7 @@ static struct snd_kcontrol_new snd_trident_vol_music_control __devinitdata = .get = snd_trident_vol_control_get, .put = snd_trident_vol_control_put, .private_value = 16, + .tlv = { .p = db_scale_gvol }, }; static struct snd_kcontrol_new snd_trident_vol_wave_control __devinitdata = @@ -2663,6 +2667,7 @@ static struct snd_kcontrol_new snd_trident_vol_wave_control __devinitdata = .get = snd_trident_vol_control_get, .put = snd_trident_vol_control_put, .private_value = 0, + .tlv = { .p = db_scale_gvol }, }; /*--------------------------------------------------------------------------- @@ -2730,6 +2735,7 @@ static struct snd_kcontrol_new snd_trident_pcm_vol_control __devinitdata = .info = snd_trident_pcm_vol_control_info, .get = snd_trident_pcm_vol_control_get, .put = snd_trident_pcm_vol_control_put, + /* FIXME: no tlv yet */ }; /*--------------------------------------------------------------------------- @@ -2839,6 +2845,8 @@ static int snd_trident_pcm_rvol_control_put(struct snd_kcontrol *kcontrol, return change; } +static DECLARE_TLV_DB_SCALE(db_scale_crvol, -3175, 25, 1); + static struct snd_kcontrol_new snd_trident_pcm_rvol_control __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2848,6 +2856,7 @@ static struct snd_kcontrol_new snd_trident_pcm_rvol_control __devinitdata = .info = snd_trident_pcm_rvol_control_info, .get = snd_trident_pcm_rvol_control_get, .put = snd_trident_pcm_rvol_control_put, + .tlv = { .p = db_scale_crvol }, }; /*--------------------------------------------------------------------------- @@ -2903,6 +2912,7 @@ static struct snd_kcontrol_new snd_trident_pcm_cvol_control __devinitdata = .info = snd_trident_pcm_cvol_control_info, .get = snd_trident_pcm_cvol_control_get, .put = snd_trident_pcm_cvol_control_put, + .tlv = { .p = db_scale_crvol }, }; static void snd_trident_notify_pcm_change1(struct snd_card *card, diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 08da9234efb3..6db3d4cc4d8d 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -59,6 +59,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/info.h> +#include <sound/tlv.h> #include <sound/ac97_codec.h> #include <sound/mpu401.h> #include <sound/initval.h> @@ -1277,7 +1278,18 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream) if (! ratep->used) ratep->rate = 0; spin_unlock_irq(&ratep->lock); - + if (! ratep->rate) { + if (! viadev->direction) { + snd_ac97_update_power(chip->ac97, + AC97_PCM_FRONT_DAC_RATE, 0); + snd_ac97_update_power(chip->ac97, + AC97_PCM_SURR_DAC_RATE, 0); + snd_ac97_update_power(chip->ac97, + AC97_PCM_LFE_DAC_RATE, 0); + } else + snd_ac97_update_power(chip->ac97, + AC97_PCM_LR_ADC_RATE, 0); + } viadev->substream = NULL; return 0; } @@ -1687,21 +1699,29 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol, return change; } +static DECLARE_TLV_DB_SCALE(db_scale_dxs, -9450, 150, 1); + static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .info = snd_via8233_dxs_volume_info, .get = snd_via8233_pcmdxs_volume_get, .put = snd_via8233_pcmdxs_volume_put, + .tlv = { .p = db_scale_dxs } }; static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = { .name = "VIA DXS Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .count = 4, .info = snd_via8233_dxs_volume_info, .get = snd_via8233_dxs_volume_get, .put = snd_via8233_dxs_volume_put, + .tlv = { .p = db_scale_dxs } }; /* @@ -2393,6 +2413,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci, int revision) { .subvendor = 0x16f3, .subdevice = 0x6405, .action = VIA_DXS_SRC }, /* Jetway K8M8MS */ { .subvendor = 0x1734, .subdevice = 0x1078, .action = VIA_DXS_SRC }, /* FSC Amilo L7300 */ { .subvendor = 0x1734, .subdevice = 0x1093, .action = VIA_DXS_SRC }, /* FSC */ + { .subvendor = 0x1734, .subdevice = 0x10ab, .action = VIA_DXS_SRC }, /* FSC */ { .subvendor = 0x1849, .subdevice = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */ { .subvendor = 0x1849, .subdevice = 0x9739, .action = VIA_DXS_SRC }, /* ASRock mobo(?) */ { .subvendor = 0x1849, .subdevice = 0x9761, .action = VIA_DXS_SRC }, /* ASRock mobo(?) */ diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index 9c03c6b4e490..e7cd8acab59a 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -26,6 +26,7 @@ #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "vx222.h" #define CARD_NAME "VX222" @@ -72,6 +73,9 @@ MODULE_DEVICE_TABLE(pci, snd_vx222_ids); /* */ +static DECLARE_TLV_DB_SCALE(db_scale_old_vol, -11350, 50, 0); +static DECLARE_TLV_DB_SCALE(db_scale_akm, -7350, 50, 0); + static struct snd_vx_hardware vx222_old_hw = { .name = "VX222/Old", @@ -81,6 +85,7 @@ static struct snd_vx_hardware vx222_old_hw = { .num_ins = 1, .num_outs = 1, .output_level_max = VX_ANALOG_OUT_LEVEL_MAX, + .output_level_db_scale = db_scale_old_vol, }; static struct snd_vx_hardware vx222_v2_hw = { @@ -92,6 +97,7 @@ static struct snd_vx_hardware vx222_v2_hw = { .num_ins = 1, .num_outs = 1, .output_level_max = VX2_AKM_LEVEL_MAX, + .output_level_db_scale = db_scale_akm, }; static struct snd_vx_hardware vx222_mic_hw = { @@ -103,6 +109,7 @@ static struct snd_vx_hardware vx222_mic_hw = { .num_ins = 1, .num_outs = 1, .output_level_max = VX2_AKM_LEVEL_MAX, + .output_level_db_scale = db_scale_akm, }; diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 9b6d345b83a6..5e51950e05f9 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -28,6 +28,7 @@ #include <sound/core.h> #include <sound/control.h> +#include <sound/tlv.h> #include <asm/io.h> #include "vx222.h" @@ -845,6 +846,8 @@ static void vx2_set_input_level(struct snd_vx222 *chip) #define MIC_LEVEL_MAX 0xff +static DECLARE_TLV_DB_SCALE(db_scale_mic, -6450, 50, 0); + /* * controls API for input levels */ @@ -922,18 +925,24 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v static struct snd_kcontrol_new vx_control_input_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Capture Volume", .info = vx_input_level_info, .get = vx_input_level_get, .put = vx_input_level_put, + .tlv = { .p = db_scale_mic }, }; static struct snd_kcontrol_new vx_control_mic_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Mic Capture Volume", .info = vx_mic_level_info, .get = vx_mic_level_get, .put = vx_mic_level_put, + .tlv = { .p = db_scale_mic }, }; /* diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index a55b5fd7da64..24f6fc52f898 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -36,6 +36,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/info.h> +#include <sound/tlv.h> #include <sound/ymfpci.h> #include <sound/asoundef.h> #include <sound/mpu401.h> @@ -1477,11 +1478,15 @@ static int snd_ymfpci_put_single(struct snd_kcontrol *kcontrol, return change; } +static DECLARE_TLV_DB_LINEAR(db_scale_native, TLV_DB_GAIN_MUTE, 0); + #define YMFPCI_DOUBLE(xname, xindex, reg) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_ymfpci_info_double, \ .get = snd_ymfpci_get_double, .put = snd_ymfpci_put_double, \ - .private_value = reg } + .private_value = reg, \ + .tlv = { .p = db_scale_native } } static int snd_ymfpci_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { |