diff options
author | Takashi Iwai <tiwai@suse.de> | 2011-12-02 09:43:52 (GMT) |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2011-12-02 09:43:52 (GMT) |
commit | b1ac29620b09f1c224c83d764675dcfaf8dd068b (patch) | |
tree | 72a28f61ffb71d12cc0f0ff6f128d1dcbaab8e7e | |
parent | a09452eeb776d1444effec5fb862c35efb623704 (diff) | |
parent | fc084e0b930d546872ab23667052499f7daf0fed (diff) | |
download | linux-b1ac29620b09f1c224c83d764675dcfaf8dd068b.tar.xz |
Merge branch 'fix/misc' into topic/misc
53 files changed, 725 insertions, 866 deletions
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 4f34432..edad99a 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -349,6 +349,7 @@ STAC92HD83* ref Reference board mic-ref Reference board with power management for ports dell-s14 Dell laptop + dell-vostro-3500 Dell Vostro 3500 laptop hp HP laptops with (inverted) mute-LED hp-dv7-4000 HP dv-7 4000 auto BIOS setup (default) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 03e2771..91fee3b 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -579,7 +579,7 @@ Development Tree ~~~~~~~~~~~~~~~~ The latest development codes for HD-audio are found on sound git tree: -- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git +- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git The master branch or for-next branches can be used as the main development branches in general while the HD-audio specific patches @@ -594,7 +594,7 @@ is, installed via the usual spells: configure, make and make install(-modules). See INSTALL in the package. The snapshot tarballs are found at: -- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ +- ftp://ftp.suse.com/pub/people/tiwai/snapshot/ Sending a Bug Report @@ -696,7 +696,7 @@ via hda-verb won't change the mixer value. The hda-verb program is found in the ftp directory: -- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/ +- ftp://ftp.suse.com/pub/people/tiwai/misc/ Also a git repository is available: @@ -764,7 +764,7 @@ operation, the jack plugging simulation, etc. The package is found in: -- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/ +- ftp://ftp.suse.com/pub/people/tiwai/misc/ A git repository is available: diff --git a/MAINTAINERS b/MAINTAINERS index 4808256..fd7e441 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5648,7 +5648,6 @@ F: drivers/media/video/*7146* F: include/media/*7146* SAMSUNG AUDIO (ASoC) DRIVERS -M: Jassi Brar <jassisinghbrar@gmail.com> M: Sangbeom Kim <sbkim73@samsung.com> L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported @@ -6122,7 +6121,7 @@ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lrg@ti.com> M: Mark Brown <broonie@opensource.wolfsonmicro.com> -T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel@alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC S: Supported diff --git a/drivers/firmware/sigma.c b/drivers/firmware/sigma.c index f10fc52..1eedb6f 100644 --- a/drivers/firmware/sigma.c +++ b/drivers/firmware/sigma.c @@ -14,13 +14,34 @@ #include <linux/module.h> #include <linux/sigma.h> -/* Return: 0==OK, <0==error, =1 ==no more actions */ +static size_t sigma_action_size(struct sigma_action *sa) +{ + size_t payload = 0; + + switch (sa->instr) { + case SIGMA_ACTION_WRITEXBYTES: + case SIGMA_ACTION_WRITESINGLE: + case SIGMA_ACTION_WRITESAFELOAD: + payload = sigma_action_len(sa); + break; + default: + break; + } + + payload = ALIGN(payload, 2); + + return payload + sizeof(struct sigma_action); +} + +/* + * Returns a negative error value in case of an error, 0 if processing of + * the firmware should be stopped after this action, 1 otherwise. + */ static int -process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw) +process_sigma_action(struct i2c_client *client, struct sigma_action *sa) { - struct sigma_action *sa = (void *)(ssfw->fw->data + ssfw->pos); size_t len = sigma_action_len(sa); - int ret = 0; + int ret; pr_debug("%s: instr:%i addr:%#x len:%zu\n", __func__, sa->instr, sa->addr, len); @@ -29,44 +50,50 @@ process_sigma_action(struct i2c_client *client, struct sigma_firmware *ssfw) case SIGMA_ACTION_WRITEXBYTES: case SIGMA_ACTION_WRITESINGLE: case SIGMA_ACTION_WRITESAFELOAD: - if (ssfw->fw->size < ssfw->pos + len) - return -EINVAL; ret = i2c_master_send(client, (void *)&sa->addr, len); if (ret < 0) return -EINVAL; break; - case SIGMA_ACTION_DELAY: - ret = 0; udelay(len); len = 0; break; - case SIGMA_ACTION_END: - return 1; - + return 0; default: return -EINVAL; } - /* when arrive here ret=0 or sent data */ - ssfw->pos += sigma_action_size(sa, len); - return ssfw->pos == ssfw->fw->size; + return 1; } static int process_sigma_actions(struct i2c_client *client, struct sigma_firmware *ssfw) { - pr_debug("%s: processing %p\n", __func__, ssfw); + struct sigma_action *sa; + size_t size; + int ret; + + while (ssfw->pos + sizeof(*sa) <= ssfw->fw->size) { + sa = (struct sigma_action *)(ssfw->fw->data + ssfw->pos); + + size = sigma_action_size(sa); + ssfw->pos += size; + if (ssfw->pos > ssfw->fw->size || size == 0) + break; + + ret = process_sigma_action(client, sa); - while (1) { - int ret = process_sigma_action(client, ssfw); pr_debug("%s: action returned %i\n", __func__, ret); - if (ret == 1) - return 0; - else if (ret) + + if (ret <= 0) return ret; } + + if (ssfw->pos != ssfw->fw->size) + return -EINVAL; + + return 0; } int process_sigma_firmware(struct i2c_client *client, const char *name) @@ -89,16 +116,24 @@ int process_sigma_firmware(struct i2c_client *client, const char *name) /* then verify the header */ ret = -EINVAL; - if (fw->size < sizeof(*ssfw_head)) + + /* + * Reject too small or unreasonable large files. The upper limit has been + * chosen a bit arbitrarily, but it should be enough for all practical + * purposes and having the limit makes it easier to avoid integer + * overflows later in the loading process. + */ + if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) goto done; ssfw_head = (void *)fw->data; if (memcmp(ssfw_head->magic, SIGMA_MAGIC, ARRAY_SIZE(ssfw_head->magic))) goto done; - crc = crc32(0, fw->data, fw->size); + crc = crc32(0, fw->data + sizeof(*ssfw_head), + fw->size - sizeof(*ssfw_head)); pr_debug("%s: crc=%x\n", __func__, crc); - if (crc != ssfw_head->crc) + if (crc != le32_to_cpu(ssfw_head->crc)) goto done; ssfw.pos = sizeof(*ssfw_head); diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index fae2950..83a9cae 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -1963,6 +1963,21 @@ #define WM8958_MICB2_DISCH_WIDTH 1 /* MICB2_DISCH */ /* + * R210 (0xD2) - Mic Detect 3 + */ +#define WM8958_MICD_LVL_MASK 0x07FC /* MICD_LVL - [10:2] */ +#define WM8958_MICD_LVL_SHIFT 2 /* MICD_LVL - [10:2] */ +#define WM8958_MICD_LVL_WIDTH 9 /* MICD_LVL - [10:2] */ +#define WM8958_MICD_VALID 0x0002 /* MICD_VALID */ +#define WM8958_MICD_VALID_MASK 0x0002 /* MICD_VALID */ +#define WM8958_MICD_VALID_SHIFT 1 /* MICD_VALID */ +#define WM8958_MICD_VALID_WIDTH 1 /* MICD_VALID */ +#define WM8958_MICD_STS 0x0001 /* MICD_STS */ +#define WM8958_MICD_STS_MASK 0x0001 /* MICD_STS */ +#define WM8958_MICD_STS_SHIFT 0 /* MICD_STS */ +#define WM8958_MICD_STS_WIDTH 1 /* MICD_STS */ + +/* * R76 (0x4C) - Charge Pump (1) */ #define WM8994_CP_ENA 0x8000 /* CP_ENA */ diff --git a/include/linux/sigma.h b/include/linux/sigma.h index e2accb3..d0de882 100644 --- a/include/linux/sigma.h +++ b/include/linux/sigma.h @@ -24,7 +24,7 @@ struct sigma_firmware { struct sigma_firmware_header { unsigned char magic[7]; u8 version; - u32 crc; + __le32 crc; }; enum { @@ -40,19 +40,14 @@ enum { struct sigma_action { u8 instr; u8 len_hi; - u16 len; - u16 addr; + __le16 len; + __be16 addr; unsigned char payload[]; }; static inline u32 sigma_action_len(struct sigma_action *sa) { - return (sa->len_hi << 16) | sa->len; -} - -static inline size_t sigma_action_size(struct sigma_action *sa, u32 payload_len) -{ - return sizeof(*sa) + payload_len + (payload_len % 2); + return (sa->len_hi << 16) | le16_to_cpu(sa->len); } extern int process_sigma_firmware(struct i2c_client *client, const char *name); diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 5dbab38..130cfe6 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -52,6 +52,7 @@ struct link_slave { struct link_ctl_info info; int vals[2]; /* current values */ unsigned int flags; + struct snd_kcontrol *kctl; /* original kcontrol pointer */ struct snd_kcontrol slave; /* the copy of original control entry */ }; @@ -252,6 +253,7 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, slave->count * sizeof(*slave->vd), GFP_KERNEL); if (!srec) return -ENOMEM; + srec->kctl = slave; srec->slave = *slave; memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd)); srec->master = master_link; @@ -333,10 +335,18 @@ static int master_put(struct snd_kcontrol *kcontrol, static void master_free(struct snd_kcontrol *kcontrol) { struct link_master *master = snd_kcontrol_chip(kcontrol); - struct link_slave *slave; - - list_for_each_entry(slave, &master->slaves, list) - slave->master = NULL; + struct link_slave *slave, *n; + + /* free all slave links and retore the original slave kctls */ + list_for_each_entry_safe(slave, n, &master->slaves, list) { + struct snd_kcontrol *sctl = slave->kctl; + struct list_head olist = sctl->list; + memcpy(sctl, &slave->slave, sizeof(*sctl)); + memcpy(sctl->vd, slave->slave.vd, + sctl->count * sizeof(*sctl->vd)); + sctl->list = olist; /* keep the current linked-list */ + kfree(slave); + } kfree(master); } diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index e083122..dbf94b1 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -148,7 +148,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, struct cs5535audio_dma_desc *desc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i]; desc->addr = cpu_to_le32(addr); - desc->size = cpu_to_le32(period_bytes); + desc->size = cpu_to_le16(period_bytes); desc->ctlreserved = cpu_to_le16(PRD_EOP); desc_addr += sizeof(struct cs5535audio_dma_desc); addr += period_bytes; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 916a186..4562e9d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2331,6 +2331,39 @@ int snd_hda_codec_reset(struct hda_codec *codec) return 0; } +typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *); + +/* apply the function to all matching slave ctls in the mixer list */ +static int map_slaves(struct hda_codec *codec, const char * const *slaves, + map_slave_func_t func, void *data) +{ + struct hda_nid_item *items; + const char * const *s; + int i, err; + + items = codec->mixers.list; + for (i = 0; i < codec->mixers.used; i++) { + struct snd_kcontrol *sctl = items[i].kctl; + if (!sctl || !sctl->id.name || + sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER) + continue; + for (s = slaves; *s; s++) { + if (!strcmp(sctl->id.name, *s)) { + err = func(data, sctl); + if (err) + return err; + break; + } + } + } + return 0; +} + +static int check_slave_present(void *data, struct snd_kcontrol *sctl) +{ + return 1; +} + /** * snd_hda_add_vmaster - create a virtual master control and add slaves * @codec: HD-audio codec @@ -2351,12 +2384,10 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char * const *slaves) { struct snd_kcontrol *kctl; - const char * const *s; int err; - for (s = slaves; *s && !snd_hda_find_mixer_ctl(codec, *s); s++) - ; - if (!*s) { + err = map_slaves(codec, slaves, check_slave_present, NULL); + if (err != 1) { snd_printdd("No slave found for %s\n", name); return 0; } @@ -2367,23 +2398,10 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, if (err < 0) return err; - for (s = slaves; *s; s++) { - struct snd_kcontrol *sctl; - int i = 0; - for (;;) { - sctl = _snd_hda_find_mixer_ctl(codec, *s, i); - if (!sctl) { - if (!i) - snd_printdd("Cannot find slave %s, " - "skipped\n", *s); - break; - } - err = snd_ctl_add_slave(kctl, sctl); - if (err < 0) - return err; - i++; - } - } + err = map_slaves(codec, slaves, (map_slave_func_t)snd_ctl_add_slave, + kctl); + if (err < 0) + return err; return 0; } EXPORT_SYMBOL_HDA(snd_hda_add_vmaster); @@ -4028,9 +4046,9 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, /* Search for codec ID */ for (q = tbl; q->subvendor; q++) { - unsigned long vendorid = (q->subdevice) | (q->subvendor << 16); - - if (vendorid == codec->subsystem_id) + unsigned int mask = 0xffff0000 | q->subdevice_mask; + unsigned int id = (q->subdevice | (q->subvendor << 16)) & mask; + if ((codec->subsystem_id & mask) == id) break; } @@ -4752,6 +4770,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, memset(sequences_hp, 0, sizeof(sequences_hp)); assoc_line_out = 0; + codec->ignore_misc_bit = true; end_nid = codec->start_nid + codec->num_nodes; for (nid = codec->start_nid; nid < end_nid; nid++) { unsigned int wid_caps = get_wcaps(codec, nid); @@ -4767,6 +4786,9 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE)) + codec->ignore_misc_bit = false; conn = get_defcfg_connect(def_conf); if (conn == AC_JACK_PORT_NONE) continue; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 755f2b0..5644711 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -854,6 +854,7 @@ struct hda_codec { unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */ unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ + unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ unsigned int power_transition :1; /* power-state in transition */ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 1c8ddf5..c1da422 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -297,10 +297,18 @@ static int hdmi_update_eld(struct hdmi_eld *e, buf + ELD_FIXED_BYTES + mnl + 3 * i); } + /* + * HDMI sink's ELD info cannot always be retrieved for now, e.g. + * in console or for audio devices. Assume the highest speakers + * configuration, to _not_ prohibit multi-channel audio playback. + */ + if (!e->spk_alloc) + e->spk_alloc = 0xffff; + + e->eld_valid = true; return 0; out_fail: - e->eld_ver = 0; return -EINVAL; } @@ -323,9 +331,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, * ELD is valid, actual eld_size is assigned in hdmi_update_eld() */ - if (!eld->eld_valid) - return -ENOENT; - size = snd_hdmi_get_eld_size(codec, nid); if (size == 0) { /* wfg: workaround for ASUS P5E-VM HDMI board */ @@ -342,18 +347,28 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, for (i = 0; i < size; i++) { unsigned int val = hdmi_get_eld_data(codec, nid, i); + /* + * Graphics driver might be writing to ELD buffer right now. + * Just abort. The caller will repoll after a while. + */ if (!(val & AC_ELDD_ELD_VALID)) { - if (!i) { - snd_printd(KERN_INFO - "HDMI: invalid ELD data\n"); - ret = -EINVAL; - goto error; - } snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n", i); - val = 0; - } else - val &= AC_ELDD_ELD_DATA; + ret = -EINVAL; + goto error; + } + val &= AC_ELDD_ELD_DATA; + /* + * The first byte cannot be zero. This can happen on some DVI + * connections. Some Intel chips may also need some 250ms delay + * to return non-zero ELD data, even when the graphics driver + * correctly writes ELD content before setting ELD_valid bit. + */ + if (!val && !i) { + snd_printdd(KERN_INFO "HDMI: 0 ELD data\n"); + ret = -EINVAL; + goto error; + } buf[i] = val; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 096507d..7d98240 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index dcbea0d..618ddad 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -510,13 +510,15 @@ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) { - return (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT) && - /* disable MISC_NO_PRESENCE check because it may break too - * many devices - */ - /*(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid) & - AC_DEFCFG_MISC_NO_PRESENCE)) &&*/ - (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP); + if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) + return false; + if (!codec->ignore_misc_bit && + (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE)) + return false; + if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) + return false; + return true; } /* flags for hda_nid_item */ @@ -651,6 +653,9 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + /* + * all fields above eld_buffer will be cleared before updating ELD + */ char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 2a2d864..70a7abd 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -58,6 +58,8 @@ struct cs_spec { unsigned int gpio_mask; unsigned int gpio_dir; unsigned int gpio_data; + unsigned int gpio_eapd_hp; /* EAPD GPIO bit for headphones */ + unsigned int gpio_eapd_speaker; /* EAPD GPIO bit for speakers */ struct hda_pcm pcm_rec[2]; /* PCM information */ @@ -76,6 +78,7 @@ enum { CS420X_MBP53, CS420X_MBP55, CS420X_IMAC27, + CS420X_APPLE, CS420X_AUTO, CS420X_MODELS }; @@ -237,6 +240,15 @@ static int cs_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); } +static void cs_update_input_select(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + if (spec->cur_adc) + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); +} + /* * Analog capture */ @@ -250,6 +262,7 @@ static int cs_capture_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_adc = spec->adc_nid[spec->cur_input]; spec->cur_adc_stream_tag = stream_tag; spec->cur_adc_format = format; + cs_update_input_select(codec); snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); return 0; } @@ -689,10 +702,8 @@ static int change_cur_input(struct hda_codec *codec, unsigned int idx, spec->cur_adc_stream_tag, 0, spec->cur_adc_format); } - snd_hda_codec_write(codec, spec->cur_adc, 0, - AC_VERB_SET_CONNECT_SEL, - spec->adc_idx[idx]); spec->cur_input = idx; + cs_update_input_select(codec); return 1; } @@ -920,10 +931,9 @@ static void cs_automute(struct hda_codec *codec) spdif_present ? 0 : PIN_OUT); } } - if (spec->board_config == CS420X_MBP53 || - spec->board_config == CS420X_MBP55 || - spec->board_config == CS420X_IMAC27) { - unsigned int gpio = hp_present ? 0x02 : 0x08; + if (spec->gpio_eapd_hp) { + unsigned int gpio = hp_present ? + spec->gpio_eapd_hp : spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, gpio); } @@ -973,10 +983,7 @@ static void cs_automic(struct hda_codec *codec) } else { spec->cur_input = spec->last_input; } - - snd_hda_codec_write_cache(codec, spec->cur_adc, 0, - AC_VERB_SET_CONNECT_SEL, - spec->adc_idx[spec->cur_input]); + cs_update_input_select(codec); } else { if (present) change_cur_input(codec, spec->automic_idx, 0); @@ -1073,9 +1080,7 @@ static void init_input(struct hda_codec *codec) cs_automic(codec); else { spec->cur_adc = spec->adc_nid[spec->cur_input]; - snd_hda_codec_write(codec, spec->cur_adc, 0, - AC_VERB_SET_CONNECT_SEL, - spec->adc_idx[spec->cur_input]); + cs_update_input_select(codec); } } else { change_cur_input(codec, spec->cur_input, 1); @@ -1273,6 +1278,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { [CS420X_MBP53] = "mbp53", [CS420X_MBP55] = "mbp55", [CS420X_IMAC27] = "imac27", + [CS420X_APPLE] = "apple", [CS420X_AUTO] = "auto", }; @@ -1282,7 +1288,13 @@ static const struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55), - SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27), + /* this conflicts with too many other models */ + /*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/ + {} /* terminator */ +}; + +static const struct snd_pci_quirk cs420x_codec_cfg_tbl[] = { + SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ }; @@ -1364,6 +1376,10 @@ static int patch_cs420x(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); + if (spec->board_config < 0) + spec->board_config = + snd_hda_check_board_codec_sid_config(codec, + CS420X_MODELS, NULL, cs420x_codec_cfg_tbl); if (spec->board_config >= 0) fix_pincfg(codec, spec->board_config, cs_pincfgs); @@ -1371,10 +1387,11 @@ static int patch_cs420x(struct hda_codec *codec) case CS420X_IMAC27: case CS420X_MBP53: case CS420X_MBP55: - /* GPIO1 = headphones */ - /* GPIO3 = speakers */ - spec->gpio_mask = 0x0a; - spec->gpio_dir = 0x0a; + case CS420X_APPLE: + spec->gpio_eapd_hp = 2; /* GPIO1 = headphones */ + spec->gpio_eapd_speaker = 8; /* GPIO3 = speakers */ + spec->gpio_mask = spec->gpio_dir = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; break; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e706e4..0de2119 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3062,7 +3062,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), - SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5066_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 81b7b79..c505fd5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -65,7 +65,11 @@ struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; + + struct hda_codec *codec; struct hdmi_eld sink_eld; + struct delayed_work work; + int repoll_count; }; struct hdmi_spec { @@ -745,8 +749,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, * Unsolicited events */ -static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld); +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) { @@ -755,7 +758,6 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) int pd = !!(res & AC_UNSOL_RES_PD); int eldv = !!(res & AC_UNSOL_RES_ELDV); int pin_idx; - struct hdmi_eld *eld; printk(KERN_INFO "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", @@ -764,17 +766,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) pin_idx = pin_nid_to_pin_index(spec, pin_nid); if (pin_idx < 0) return; - eld = &spec->pins[pin_idx].sink_eld; - - hdmi_present_sense(codec, pin_nid, eld); - /* - * HDMI sink's ELD info cannot always be retrieved for now, e.g. - * in console or for audio devices. Assume the highest speakers - * configuration, to _not_ prohibit multi-channel audio playback. - */ - if (!eld->spk_alloc) - eld->spk_alloc = 0xffff; + hdmi_present_sense(&spec->pins[pin_idx], 1); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -968,9 +961,11 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return 0; } -static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld) +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { + struct hda_codec *codec = per_pin->codec; + struct hdmi_eld *eld = &per_pin->sink_eld; + hda_nid_t pin_nid = per_pin->pin_nid; /* * Always execute a GetPinSense verb here, even when called from * hdmi_intrinsic_event; for some NVIDIA HW, the unsolicited @@ -980,26 +975,42 @@ static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, * the unsolicited response to avoid custom WARs. */ int present = snd_hda_pin_sense(codec, pin_nid); + bool eld_valid = false; - memset(eld, 0, sizeof(*eld)); + memset(eld, 0, offsetof(struct hdmi_eld, eld_buffer)); eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); if (eld->monitor_present) - eld->eld_valid = !!(present & AC_PINSENSE_ELDV); - else - eld->eld_valid = 0; + eld_valid = !!(present & AC_PINSENSE_ELDV); printk(KERN_INFO "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, eld->monitor_present, eld_valid); - if (eld->eld_valid) + if (eld_valid) { if (!snd_hdmi_get_eld(eld, codec, pin_nid)) snd_hdmi_show_eld(eld); + else if (repoll) { + queue_delayed_work(codec->bus->workq, + &per_pin->work, + msecs_to_jiffies(300)); + } + } snd_hda_input_jack_report(codec, pin_nid); } +static void hdmi_repoll_eld(struct work_struct *work) +{ + struct hdmi_spec_per_pin *per_pin = + container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work); + + if (per_pin->repoll_count++ > 6) + per_pin->repoll_count = 0; + + hdmi_present_sense(per_pin, per_pin->repoll_count); +} + static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) { struct hdmi_spec *spec = codec->spec; @@ -1228,7 +1239,7 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) if (err < 0) return err; - hdmi_present_sense(codec, per_pin->pin_nid, &per_pin->sink_eld); + hdmi_present_sense(per_pin, 0); return 0; } @@ -1279,6 +1290,8 @@ static int generic_hdmi_init(struct hda_codec *codec) AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | pin_nid); + per_pin->codec = codec; + INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld); snd_hda_eld_proc_new(codec, eld, pin_idx); } return 0; @@ -1293,10 +1306,12 @@ static void generic_hdmi_free(struct hda_codec *codec) struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; struct hdmi_eld *eld = &per_pin->sink_eld; + cancel_delayed_work(&per_pin->work); snd_hda_eld_proc_free(codec, eld); } snd_hda_input_jack_free(codec); + flush_workqueue(codec->bus->workq); kfree(spec); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a24e068..cbde019 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -277,6 +277,12 @@ static bool alc_dyn_adc_pcm_resetup(struct hda_codec *codec, int cur) return false; } +static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx) +{ + return spec->capsrc_nids ? + spec->capsrc_nids[idx] : spec->adc_nids[idx]; +} + /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, unsigned int idx, bool force) @@ -284,7 +290,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, struct alc_spec *spec = codec->spec; const struct hda_input_mux *imux; unsigned int mux_idx; - int i, type; + int i, type, num_conns; hda_nid_t nid; mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; @@ -303,20 +309,20 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, adc_idx = spec->dyn_adc_idx[idx]; } - nid = spec->capsrc_nids ? - spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; + nid = get_capsrc(spec, adc_idx); /* no selection? */ - if (snd_hda_get_conn_list(codec, nid, NULL) <= 1) + num_conns = snd_hda_get_conn_list(codec, nid, NULL); + if (num_conns <= 1) return 1; type = get_wcaps_type(get_wcaps(codec, nid)); if (type == AC_WID_AUD_MIX) { /* Matrix-mixer style (e.g. ALC882) */ - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, + int active = imux->items[idx].index; + for (i = 0; i < num_conns; i++) { + unsigned int v = (i == active) ? 0 : HDA_AMP_MUTE; + snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, i, HDA_AMP_MUTE, v); } } else { @@ -1053,8 +1059,19 @@ static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec) spec->imux_pins[2] = spec->dock_mic_pin; for (i = 0; i < 3; i++) { strcpy(imux->items[i].label, texts[i]); - if (spec->imux_pins[i]) + if (spec->imux_pins[i]) { + hda_nid_t pin = spec->imux_pins[i]; + int c; + for (c = 0; c < spec->num_adc_nids; c++) { + hda_nid_t cap = get_capsrc(spec, c); + int idx = get_connection_index(codec, cap, pin); + if (idx >= 0) { + imux->items[i].index = idx; + break; + } + } imux->num_items = i + 1; + } } spec->num_mux_defs = 1; spec->input_mux = imux; @@ -1451,7 +1468,7 @@ static void alc_apply_fixup(struct hda_codec *codec, int action) switch (fix->type) { case ALC_FIXUP_SKU: if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku) - break;; + break; snd_printdd(KERN_INFO "hda_codec: %s: " "Apply sku override for %s\n", codec->chip_name, modelname); @@ -1956,10 +1973,8 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - const hda_nid_t *nids = spec->capsrc_nids; - if (!nids) - nids = spec->adc_nids; - err = snd_hda_add_nid(codec, kctl, i, nids[i]); + err = snd_hda_add_nid(codec, kctl, i, + get_capsrc(spec, i)); if (err < 0) return err; } @@ -2746,8 +2761,7 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec) } for (c = 0; c < num_adcs; c++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[c] : spec->adc_nids[c]; + hda_nid_t cap = get_capsrc(spec, c); idx = get_connection_index(codec, cap, pin); if (idx >= 0) { spec->imux_pins[imux->num_items] = pin; @@ -3693,8 +3707,7 @@ static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin) if (!pin) return 0; for (i = 0; i < spec->num_adc_nids; i++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[i] : spec->adc_nids[i]; + hda_nid_t cap = get_capsrc(spec, i); int idx; idx = get_connection_index(codec, cap, pin); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4e715fe..d8d2f9d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -95,6 +95,7 @@ enum { STAC_92HD83XXX_REF, STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, + STAC_DELL_VOSTRO_3500, STAC_92HD83XXX_HP, STAC_92HD83XXX_HP_cNB11_INTQUAD, STAC_HP_DV7_4000, @@ -226,7 +227,6 @@ struct sigmatel_spec { /* power management */ unsigned int num_pwrs; - const unsigned int *pwr_mapping; const hda_nid_t *pwr_nids; const hda_nid_t *dac_list; @@ -373,18 +373,15 @@ static const unsigned long stac92hd73xx_capvols[] = { #define STAC92HD83_DAC_COUNT 3 -static const hda_nid_t stac92hd83xxx_pwr_nids[4] = { - 0xa, 0xb, 0xd, 0xe, +static const hda_nid_t stac92hd83xxx_pwr_nids[7] = { + 0x0a, 0x0b, 0x0c, 0xd, 0x0e, + 0x0f, 0x10 }; static const hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { 0x1e, 0, }; -static const unsigned int stac92hd83xxx_pwr_mapping[4] = { - 0x03, 0x0c, 0x20, 0x40, -}; - static const hda_nid_t stac92hd83xxx_dmic_nids[] = { 0x11, 0x20, }; @@ -1644,6 +1641,8 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a, "Alienware M17x", STAC_ALIENWARE_M17X), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, + "Alienware M17x", STAC_ALIENWARE_M17X), {} /* terminator */ }; @@ -1659,6 +1658,12 @@ static const unsigned int dell_s14_pin_configs[10] = { 0x40f000f0, 0x40f000f0, }; +static const unsigned int dell_vostro_3500_pin_configs[10] = { + 0x02a11020, 0x0221101f, 0x400000f0, 0x90170110, + 0x400000f1, 0x400000f2, 0x400000f3, 0x90a60160, + 0x400000f4, 0x400000f5, +}; + static const unsigned int hp_dv7_4000_pin_configs[10] = { 0x03a12050, 0x0321201f, 0x40f000f0, 0x90170110, 0x40f000f0, 0x40f000f0, 0x90170110, 0xd5a30140, @@ -1675,6 +1680,7 @@ static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, [STAC_DELL_S14] = dell_s14_pin_configs, + [STAC_DELL_VOSTRO_3500] = dell_vostro_3500_pin_configs, [STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs, [STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs, }; @@ -1684,6 +1690,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", + [STAC_DELL_VOSTRO_3500] = "dell-vostro-3500", [STAC_92HD83XXX_HP] = "hp", [STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad", [STAC_HP_DV7_4000] = "hp-dv7-4000", @@ -1697,6 +1704,8 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD83XXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, "unknown Dell", STAC_DELL_S14), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x1028, + "Dell Vostro 3500", STAC_DELL_VOSTRO_3500), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600, "HP", STAC_92HD83XXX_HP), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656, @@ -4432,7 +4441,9 @@ static int stac92xx_init(struct hda_codec *codec) int pinctl, def_conf; /* power on when no jack detection is available */ - if (!spec->hp_detect) { + /* or when the VREF is used for controlling LED */ + if (!spec->hp_detect || + (spec->gpio_led > 8 && spec->gpio_led == nid)) { stac_toggle_power_map(codec, nid, 1); continue; } @@ -4459,8 +4470,12 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 1); continue; } - if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) + if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) { stac_issue_unsol_event(codec, nid); + continue; + } + /* none of the above, turn the port OFF */ + stac_toggle_power_map(codec, nid, 0); } /* sync mute LED */ @@ -4716,11 +4731,7 @@ static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, if (idx >= spec->num_pwrs) return; - /* several codecs have two power down bits */ - if (spec->pwr_mapping) - idx = spec->pwr_mapping[idx]; - else - idx = 1 << idx; + idx = 1 << idx; val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0xff; if (enable) @@ -5046,20 +5057,6 @@ static int stac92xx_pre_resume(struct hda_codec *codec) return 0; } -static int stac92xx_post_suspend(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - if (spec->gpio_led > 8) { - /* with vref-out pin used for mute led control - * codec AFG is prevented from D3 state, but on - * system suspend it can (and should) be used - */ - snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } - return 0; -} - static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { @@ -5618,9 +5615,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e); } - /* reset pin power-down; Windows may leave these bits after reboot */ - snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7EC, 0); - snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7ED, 0); codec->no_trigger_sense = 1; codec->spec = spec; @@ -5630,7 +5624,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->digbeep_nid = 0x21; spec->pwr_nids = stac92hd83xxx_pwr_nids; - spec->pwr_mapping = stac92hd83xxx_pwr_mapping; spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; spec->init = stac92hd83xxx_core_init; @@ -5647,9 +5640,6 @@ again: stac92xx_set_config_regs(codec, stac92hd83xxx_brd_tbl[spec->board_config]); - if (spec->board_config != STAC_92HD83XXX_PWR_REF) - spec->num_pwrs = 0; - codec->patch_ops = stac92xx_patch_ops; if (find_mute_led_gpio(codec, 0)) @@ -5666,8 +5656,6 @@ again: } else { codec->patch_ops.set_power_state = stac92xx_set_power_state; - codec->patch_ops.post_suspend = - stac92xx_post_suspend; } codec->patch_ops.pre_resume = stac92xx_pre_resume; codec->patch_ops.check_power_status = @@ -5858,8 +5846,6 @@ again: (codec->revision_id & 0xf) == 1) spec->stream_delay = 40; /* 40 milliseconds */ - /* no output amps */ - spec->num_pwrs = 0; /* disable VSW */ spec->init = stac92hd71bxx_core_init; unmute_init++; @@ -5874,8 +5860,6 @@ again: if ((codec->revision_id & 0xf) == 1) spec->stream_delay = 40; /* 40 milliseconds */ - /* no output amps */ - spec->num_pwrs = 0; /* fallthru */ default: spec->init = stac92hd71bxx_core_init; @@ -5985,8 +5969,6 @@ again: } else { codec->patch_ops.set_power_state = stac92xx_set_power_state; - codec->patch_ops.post_suspend = - stac92xx_post_suspend; } codec->patch_ops.pre_resume = stac92xx_pre_resume; codec->patch_ops.check_power_status = diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 431c0d4..b513762 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -208,6 +208,7 @@ struct via_spec { /* work to check hp jack state */ struct hda_codec *codec; struct delayed_work vt1708_hp_work; + int hp_work_active; int vt1708_jack_detect; int vt1708_hp_present; @@ -305,27 +306,35 @@ enum { static void analog_low_current_mode(struct hda_codec *codec); static bool is_aa_path_mute(struct hda_codec *codec); -static void vt1708_start_hp_work(struct via_spec *spec) +#define hp_detect_with_aa(codec) \ + (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1 && \ + !is_aa_path_mute(codec)) + +static void vt1708_stop_hp_work(struct via_spec *spec) { if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) return; - snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, - !spec->vt1708_jack_detect); - if (!delayed_work_pending(&spec->vt1708_hp_work)) - schedule_delayed_work(&spec->vt1708_hp_work, - msecs_to_jiffies(100)); + if (spec->hp_work_active) { + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 1); + cancel_delayed_work_sync(&spec->vt1708_hp_work); + spec->hp_work_active = 0; + } } -static void vt1708_stop_hp_work(struct via_spec *spec) +static void vt1708_update_hp_work(struct via_spec *spec) { if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) return; - if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1 - && !is_aa_path_mute(spec->codec)) - return; - snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, - !spec->vt1708_jack_detect); - cancel_delayed_work_sync(&spec->vt1708_hp_work); + if (spec->vt1708_jack_detect && + (spec->active_streams || hp_detect_with_aa(spec->codec))) { + if (!spec->hp_work_active) { + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 0); + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); + spec->hp_work_active = 1; + } + } else if (!hp_detect_with_aa(spec->codec)) + vt1708_stop_hp_work(spec); } static void set_widgets_power_state(struct hda_codec *codec) @@ -343,12 +352,7 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol, set_widgets_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol)); - if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { - if (is_aa_path_mute(codec)) - vt1708_start_hp_work(codec->spec); - else - vt1708_stop_hp_work(codec->spec); - } + vt1708_update_hp_work(codec->spec); return change; } @@ -1154,7 +1158,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_dac_stream_tag = stream_tag; spec->cur_dac_format = format; mutex_unlock(&spec->config_mutex); - vt1708_start_hp_work(spec); + vt1708_update_hp_work(spec); return 0; } @@ -1174,7 +1178,7 @@ static int via_playback_hp_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_hp_stream_tag = stream_tag; spec->cur_hp_format = format; mutex_unlock(&spec->config_mutex); - vt1708_start_hp_work(spec); + vt1708_update_hp_work(spec); return 0; } @@ -1188,7 +1192,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo, snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); spec->active_streams &= ~STREAM_MULTI_OUT; mutex_unlock(&spec->config_mutex); - vt1708_stop_hp_work(spec); + vt1708_update_hp_work(spec); return 0; } @@ -1203,7 +1207,7 @@ static int via_playback_hp_pcm_cleanup(struct hda_pcm_stream *hinfo, snd_hda_codec_setup_stream(codec, spec->hp_dac_nid, 0, 0, 0); spec->active_streams &= ~STREAM_INDEP_HP; mutex_unlock(&spec->config_mutex); - vt1708_stop_hp_work(spec); + vt1708_update_hp_work(spec); return 0; } @@ -1645,7 +1649,8 @@ static void via_hp_automute(struct hda_codec *codec) int nums; struct via_spec *spec = codec->spec; - if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0]) + if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0] && + (spec->codec_type != VT1708 || spec->vt1708_jack_detect)) present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); if (spec->smart51_enabled) @@ -2612,8 +2617,6 @@ static int vt1708_jack_detect_get(struct snd_kcontrol *kcontrol, if (spec->codec_type != VT1708) return 0; - spec->vt1708_jack_detect = - !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); ucontrol->value.integer.value[0] = spec->vt1708_jack_detect; return 0; } @@ -2623,18 +2626,22 @@ static int vt1708_jack_detect_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - int change; + int val; if (spec->codec_type != VT1708) return 0; - spec->vt1708_jack_detect = ucontrol->value.integer.value[0]; - change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) - == !spec->vt1708_jack_detect; - if (spec->vt1708_jack_detect) { + val = !!ucontrol->value.integer.value[0]; + if (spec->vt1708_jack_detect == val) + return 0; + spec->vt1708_jack_detect = val; + if (spec->vt1708_jack_detect && + snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") != 1) { mute_aa_path(codec, 1); notify_aa_path_ctls(codec); } - return change; + via_hp_automute(codec); + vt1708_update_hp_work(spec); + return 1; } static const struct snd_kcontrol_new vt1708_jack_detect_ctl = { @@ -2771,6 +2778,7 @@ static int via_init(struct hda_codec *codec) via_auto_init_unsol_event(codec); via_hp_automute(codec); + vt1708_update_hp_work(spec); return 0; } @@ -2787,7 +2795,9 @@ static void vt1708_update_hp_jack_state(struct work_struct *work) spec->vt1708_hp_present ^= 1; via_hp_automute(spec->codec); } - vt1708_start_hp_work(spec); + if (spec->vt1708_jack_detect) + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); } static int get_mux_nids(struct hda_codec *codec) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 29e3125..11718b49 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1077,6 +1077,13 @@ static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *subs } if (civ != igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV)) continue; + + /* IO read operation is very expensive inside virtual machine + * as it is emulated. The probability that subsequent PICB read + * will return different result is high enough to loop till + * timeout here. + * Same CIV is strict enough condition to be sure that PICB + * is valid inside VM on emulated card. */ if (chip->inside_vm) break; if (ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb)) @@ -2930,6 +2937,45 @@ static unsigned int sis_codec_bits[3] = { ICH_PCR, ICH_SCR, ICH_SIS_TCR }; +static int __devinit snd_intel8x0_inside_vm(struct pci_dev *pci) +{ + int result = inside_vm; + char *msg = NULL; + + /* check module parameter first (override detection) */ + if (result >= 0) { + msg = result ? "enable (forced) VM" : "disable (forced) VM"; + goto fini; + } + + /* detect KVM and Parallels virtual environments */ + result = kvm_para_available(); +#ifdef X86_FEATURE_HYPERVISOR + result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); +#endif + if (!result) + goto fini; + + /* check for known (emulated) devices */ + if (pci->subsystem_vendor == 0x1af4 && + pci->subsystem_device == 0x1100) { + /* KVM emulated sound, PCI SSID: 1af4:1100 */ + msg = "enable KVM"; + } else if (pci->subsystem_vendor == 0x1ab8) { + /* Parallels VM emulated sound, PCI SSID: 1ab8:xxxx */ + msg = "enable Parallels VM"; + } else { + msg = "disable (unknown or VT-d) VM"; + result = 0; + } + +fini: + if (msg != NULL) + printk(KERN_INFO "intel8x0: %s optimization\n", msg); + + return result; +} + static int __devinit snd_intel8x0_create(struct snd_card *card, struct pci_dev *pci, unsigned long device_type, @@ -2997,9 +3043,7 @@ static int __devinit snd_intel8x0_create(struct snd_card *card, if (xbox) chip->xbox = 1; - chip->inside_vm = inside_vm; - if (inside_vm) - printk(KERN_INFO "intel8x0: enable KVM optimization\n"); + chip->inside_vm = snd_intel8x0_inside_vm(pci); if (pci->vendor == PCI_VENDOR_ID_INTEL && pci->device == PCI_DEVICE_ID_INTEL_440MX) @@ -3243,14 +3287,6 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci, buggy_irq = 0; } - if (inside_vm < 0) { - /* detect KVM and Parallels virtual environments */ - inside_vm = kvm_para_available(); -#if defined(__i386__) || defined(__x86_64__) - inside_vm = inside_vm || boot_cpu_has(X86_FEATURE_HYPERVISOR); -#endif - } - if ((err = snd_intel8x0_create(card, pci, pci_id->driver_data, &chip)) < 0) { snd_card_free(card); diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 5c8717e..8c3e7fc 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -78,10 +78,15 @@ unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port) return ioread32(address); } -void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len) +static void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, + u32 len) { - void __iomem *address = lx_dsp_register(chip, port); - memcpy_fromio(data, address, len*sizeof(u32)); + u32 __iomem *address = lx_dsp_register(chip, port); + int i; + + /* we cannot use memcpy_fromio */ + for (i = 0; i != len; ++i) + data[i] = ioread32(address + i); } @@ -91,11 +96,15 @@ void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data) iowrite32(data, address); } -void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data, - u32 len) +static void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, + const u32 *data, u32 len) { - void __iomem *address = lx_dsp_register(chip, port); - memcpy_toio(address, data, len*sizeof(u32)); + u32 __iomem *address = lx_dsp_register(chip, port); + int i; + + /* we cannot use memcpy_to */ + for (i = 0; i != len; ++i) + iowrite32(data[i], address + i); } diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 1dd5629..4d7ff79 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -72,10 +72,7 @@ enum { }; unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port); -void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len); void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data); -void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data, - u32 len); /* plx register access */ enum { diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index e760ada..19ee220 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6518,7 +6518,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, hdspm->io_type = AES32; hdspm->card_name = "RME AES32"; hdspm->midiPorts = 2; - } else if ((hdspm->firmware_rev == 0xd5) || + } else if ((hdspm->firmware_rev == 0xd2) || ((hdspm->firmware_rev >= 0xc8) && (hdspm->firmware_rev <= 0xcf))) { hdspm->io_type = MADI; diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index a391e62..28dfafb 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -41,6 +41,7 @@ MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int enable = 1; +static int codecs = 1; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); @@ -48,6 +49,8 @@ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); module_param(enable, bool, 0444); MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); +module_param(codecs, int, 0444); +MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)"); static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, @@ -140,6 +143,9 @@ struct sis7019 { dma_addr_t silence_dma_addr; }; +/* These values are also used by the module param 'codecs' to indicate + * which codecs should be present. + */ #define SIS_PRIMARY_CODEC_PRESENT 0x0001 #define SIS_SECONDARY_CODEC_PRESENT 0x0002 #define SIS_TERTIARY_CODEC_PRESENT 0x0004 @@ -1078,6 +1084,7 @@ static int sis_chip_init(struct sis7019 *sis) { unsigned long io = sis->ioport; void __iomem *ioaddr = sis->ioaddr; + unsigned long timeout; u16 status; int count; int i; @@ -1104,21 +1111,45 @@ static int sis_chip_init(struct sis7019 *sis) while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) udelay(1); + /* Command complete, we can let go of the semaphore now. + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!count) + return -EIO; + /* Now that we've finished the reset, find out what's attached. + * There are some codec/board combinations that take an extremely + * long time to come up. 350+ ms has been observed in the field, + * so we'll give them up to 500ms. */ - status = inl(io + SIS_AC97_STATUS); - if (status & SIS_AC97_STATUS_CODEC_READY) - sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; - if (status & SIS_AC97_STATUS_CODEC2_READY) - sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; - if (status & SIS_AC97_STATUS_CODEC3_READY) - sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; - - /* All done, let go of the semaphore, and check for errors + sis->codecs_present = 0; + timeout = msecs_to_jiffies(500) + jiffies; + while (time_before_eq(jiffies, timeout)) { + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + if (sis->codecs_present == codecs) + break; + + msleep(1); + } + + /* All done, check for errors. */ - outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); - if (!sis->codecs_present || !count) + if (!sis->codecs_present) { + printk(KERN_ERR "sis7019: could not find any codecs\n"); return -EIO; + } + + if (sis->codecs_present != codecs) { + printk(KERN_WARNING "sis7019: missing codecs, found %0x, expected %0x\n", + sis->codecs_present, codecs); + } /* Let the hardware know that the audio driver is alive, * and enable PCM slots on the AC-link for L/R playback (3 & 4) and @@ -1390,6 +1421,17 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci, if (!enable) goto error_out; + /* The user can specify which codecs should be present so that we + * can wait for them to show up if they are slow to recover from + * the AC97 cold reset. We default to a single codec, the primary. + * + * We assume that SIS_PRIMARY_*_PRESENT matches bits 0-2. + */ + codecs &= SIS_PRIMARY_CODEC_PRESENT | SIS_SECONDARY_CODEC_PRESENT | + SIS_TERTIARY_CODEC_PRESENT; + if (!codecs) + codecs = SIS_PRIMARY_CODEC_PRESENT; + rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card); if (rc < 0) goto error_out; diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index bee3c94..d1fcc81 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -1,6 +1,6 @@ config SND_ATMEL_SOC tristate "SoC Audio for the Atmel System-on-Chip" - depends on ARCH_AT91 || AVR32 + depends on ARCH_AT91 help Say Y or M if you want to add support for codecs attached to the ATMEL SSC interface. You will also need @@ -24,25 +24,6 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. -config SND_AT32_SOC_PLAYPAQ - tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS - select SND_ATMEL_SOC_SSC - select SND_SOC_WM8510 - help - Say Y or M here if you want to add support for SoC audio - on the LRS PlayPaq. - -config SND_AT32_SOC_PLAYPAQ_SLAVE - bool "Run CODEC on PlayPaq in slave mode" - depends on SND_AT32_SOC_PLAYPAQ - default n - help - Say Y if you want to run with the AT32 SSC generating the BCLK - and FRAME signals on the PlayPaq. Unless you want to play - with the AT32 as the SSC master, you probably want to say N here, - as this will give you better sound quality. - config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index e7ea56b..a5c0bf1 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -8,9 +8,5 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o -# AT32 Machine Support -snd-soc-playpaq-objs := playpaq_wm8510.o - obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o -obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c deleted file mode 100644 index 73ae99a..0000000 --- a/sound/soc/atmel/playpaq_wm8510.c +++ /dev/null @@ -1,473 +0,0 @@ -/* sound/soc/at32/playpaq_wm8510.c - * ASoC machine driver for PlayPaq using WM8510 codec - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum@acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c - * - * NOTE: If you don't have the AT32 enhanced portmux configured (which - * isn't currently in the mainline or Atmel patched kernel), you will - * need to set the MCLK pin (PA30) to peripheral A in your board initialization - * code. Something like: - * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); - * - */ - -/* #define DEBUG */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/errno.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/at32ap700x.h> -#include <mach/portmux.h> - -#include "../codecs/wm8510.h" -#include "atmel-pcm.h" -#include "atmel_ssc_dai.h" - - -/*-------------------------------------------------------------------------*\ - * constants -\*-------------------------------------------------------------------------*/ -#define MCLK_PIN GPIO_PIN_PA(30) -#define MCLK_PERIPH GPIO_PERIPH_A - - -/*-------------------------------------------------------------------------*\ - * data types -\*-------------------------------------------------------------------------*/ -/* SSC clocking data */ -struct ssc_clock_data { - /* CMR div */ - unsigned int cmr_div; - - /* Frame period (as needed by xCMR.PERIOD) */ - unsigned int period; - - /* The SSC clock rate these settings where calculated for */ - unsigned long ssc_rate; -}; - - -/*-------------------------------------------------------------------------*\ - * module data -\*-------------------------------------------------------------------------*/ -static struct clk *_gclk0; -static struct clk *_pll0; - -#define CODEC_CLK (_gclk0) - - -/*-------------------------------------------------------------------------*\ - * Sound SOC operations -\*-------------------------------------------------------------------------*/ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE -static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai); - struct ssc_device *ssc = ssc_p->ssc; - struct ssc_clock_data cd; - unsigned int rate, width_bits, channels; - unsigned int bitrate, ssc_div; - unsigned actual_rate; - - - /* - * Figure out required bitrate - */ - rate = params_rate(params); - channels = params_channels(params); - width_bits = snd_pcm_format_physical_width(params_format(params)); - bitrate = rate * width_bits * channels; - - - /* - * Figure out required SSC divider and period for required bitrate - */ - cd.ssc_rate = clk_get_rate(ssc->clk); - ssc_div = cd.ssc_rate / bitrate; - cd.cmr_div = ssc_div / 2; - if (ssc_div & 1) { - /* round cmr_div up */ - cd.cmr_div++; - } - cd.period = width_bits - 1; - - - /* - * Find actual rate, compare to requested rate - */ - actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n", - rate, actual_rate); - - - return cd; -} -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - -static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai); - struct ssc_device *ssc = ssc_p->ssc; - unsigned int pll_out = 0, bclk = 0, mclk_div = 0; - int ret; - - - /* Due to difficulties with getting the correct clocks from the AT32's - * PLL0, we're going to let the CODEC be in charge of all the clocks - */ -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); -#else - struct ssc_clock_data cd; - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); -#endif - - if (ssc == NULL) { - pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); - return -EINVAL; - } - - - /* - * Figure out PLL and BCLK dividers for WM8510 - */ - switch (params_rate(params)) { - case 48000: - pll_out = 24576000; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 44100: - pll_out = 22579200; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 22050: - pll_out = 22579200; - mclk_div = WM8510_MCLKDIV_4; - bclk = WM8510_BCLKDIV_8; - break; - - case 16000: - pll_out = 24576000; - mclk_div = WM8510_MCLKDIV_6; - bclk = WM8510_BCLKDIV_8; - break; - - case 11025: - pll_out = 22579200; - mclk_div = WM8510_MCLKDIV_8; - bclk = WM8510_BCLKDIV_8; - break; - - case 8000: - pll_out = 24576000; - mclk_div = WM8510_MCLKDIV_12; - bclk = WM8510_BCLKDIV_8; - break; - - default: - pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", - params_rate(params)); - return -EINVAL; - } - - - /* - * set CPU and CODEC DAI configuration - */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CODEC DAI format (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU DAI format (%d)\n", - ret); - return ret; - } - - - /* - * Set CPU clock configuration - */ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); - pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", - cd.cmr_div, cd.period); - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, - cd.period); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU transmit period (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - /* - * Set CODEC clock configuration - */ - pr_debug("playpaq_wm8510: " - "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", - clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); - - -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); - if (ret < 0) { - pr_warning - ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - ret = snd_soc_dai_set_pll(codec_dai, 0, 0, - clk_get_rate(CODEC_CLK), pll_out); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", - ret); - return ret; - } - - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", - ret); - return ret; - } - - - return 0; -} - - - -static struct snd_soc_ops playpaq_wm8510_ops = { - .hw_params = playpaq_wm8510_hw_params, -}; - - - -static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - - - -static const struct snd_soc_dapm_route intercon[] = { - /* speaker connected to SPKOUT */ - {"Ext Spk", NULL, "SPKOUTP"}, - {"Ext Spk", NULL, "SPKOUTN"}, - - {"Mic Bias", NULL, "Int Mic"}, - {"MICN", NULL, "Mic Bias"}, - {"MICP", NULL, "Mic Bias"}, -}; - - - -static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int i; - - /* - * Add DAPM widgets - */ - for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) - snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]); - - - - /* - * Setup audio path interconnects - */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - - - /* always connected pins */ - snd_soc_dapm_enable_pin(dapm, "Int Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - - - - /* Make CSB show PLL rate */ - snd_soc_dai_set_clkdiv(rtd->codec_dai, WM8510_OPCLKDIV, - WM8510_OPCLKDIV_1 | 4); - - return 0; -} - - - -static struct snd_soc_dai_link playpaq_wm8510_dai = { - .name = "WM8510", - .stream_name = "WM8510 PCM", - .cpu_dai_name= "atmel-ssc-dai.0", - .platform_name = "atmel-pcm-audio", - .codec_name = "wm8510-codec.0-0x1a", - .codec_dai_name = "wm8510-hifi", - .init = playpaq_wm8510_init, - .ops = &playpaq_wm8510_ops, -}; - - - -static struct snd_soc_card snd_soc_playpaq = { - .name = "LRS_PlayPaq_WM8510", - .dai_link = &playpaq_wm8510_dai, - .num_links = 1, -}; - -static struct platform_device *playpaq_snd_device; - - -static int __init playpaq_asoc_init(void) -{ - int ret = 0; - - /* - * Configure MCLK for WM8510 - */ - _gclk0 = clk_get(NULL, "gclk0"); - if (IS_ERR(_gclk0)) { - _gclk0 = NULL; - ret = PTR_ERR(_gclk0); - goto err_gclk0; - } - _pll0 = clk_get(NULL, "pll0"); - if (IS_ERR(_pll0)) { - _pll0 = NULL; - ret = PTR_ERR(_pll0); - goto err_pll0; - } - ret = clk_set_parent(_gclk0, _pll0); - if (ret) { - pr_warning("snd-soc-playpaq: " - "Failed to set PLL0 as parent for DAC clock\n"); - goto err_set_clk; - } - clk_set_rate(CODEC_CLK, 12000000); - clk_enable(CODEC_CLK); - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); -#endif - - - /* - * Create and register platform device - */ - playpaq_snd_device = platform_device_alloc("soc-audio", 0); - if (playpaq_snd_device == NULL) { - ret = -ENOMEM; - goto err_device_alloc; - } - - platform_set_drvdata(playpaq_snd_device, &snd_soc_playpaq); - - ret = platform_device_add(playpaq_snd_device); - if (ret) { - pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", - ret); - goto err_device_add; - } - - return 0; - - -err_device_add: - if (playpaq_snd_device != NULL) { - platform_device_put(playpaq_snd_device); - playpaq_snd_device = NULL; - } -err_device_alloc: -err_set_clk: - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } -err_pll0: - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - return ret; -} - - -static void __exit playpaq_asoc_exit(void) -{ - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_free_pin(MCLK_PIN); -#endif - - platform_device_unregister(playpaq_snd_device); - playpaq_snd_device = NULL; -} - -module_init(playpaq_asoc_init); -module_exit(playpaq_asoc_exit); - -MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>"); -MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 444747f..dd7be0d 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -34,7 +34,7 @@ #define AD1836_ADC_CTRL2 13 #define AD1836_ADC_WORD_LEN_MASK 0x30 -#define AD1836_ADC_WORD_OFFSET 5 +#define AD1836_ADC_WORD_OFFSET 4 #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1ccf8dd..45c6302 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -245,7 +245,7 @@ static const char *adau1373_bass_hpf_cutoff_text[] = { }; static const unsigned int adau1373_bass_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(3), 0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1), 3, 4, TLV_DB_SCALE_ITEM(950, 250, 0), 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1f237e..73f46eb 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c_client = to_i2c_client(codec->dev); int reg; regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), @@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) ndelay(500); /* first restore the entire register cache ... */ - for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { - u8 val = snd_soc_read(codec, reg); - - if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { - dev_err(codec->dev, "i2c write failed\n"); - return -EIO; - } - } + snd_soc_cache_sync(codec); /* ... then disable the power-down bits */ reg = snd_soc_read(codec, CS4270_PWRCTL); diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 23d1bd5..69fde15 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -434,7 +434,8 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { int ret; /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; return 0; @@ -501,8 +502,9 @@ static int cs4271_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 8c3c820..1ee66361 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -555,7 +555,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .probe = cs42l51_probe, - .reg_cache_size = CS42L51_NUMREGS, + .reg_cache_size = CS42L51_NUMREGS + 1, .reg_word_size = sizeof(u8), }; diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 9e7e964..dcf6f2a 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 1; + unsigned int change = 0; - if (((max9877_regs[reg] >> shift) & mask) == val) - change = 0; + if (((max9877_regs[reg] >> shift) & mask) != val) + change = 1; - if (((max9877_regs[reg2] >> shift) & mask) == val2) - change = 0; + if (((max9877_regs[reg2] >> shift) & mask) != val2) + change = 1; if (change) { max9877_regs[reg] &= ~(mask << shift); diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 27a078c..4646e80 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -177,7 +177,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */ static unsigned int mic_bst_tlv[] = { - TLV_DB_RANGE_HEAD(6), + TLV_DB_RANGE_HEAD(7), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d15695d..bbcf921 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -365,7 +365,7 @@ static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ static const unsigned int mic_gain_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(2), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), }; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index bb82408..d2f3715 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -76,6 +76,8 @@ struct sta32x_priv { unsigned int mclk; unsigned int format; + + u32 coef_shadow[STA32X_COEF_COUNT]; }; static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); @@ -227,6 +229,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); int numcoef = kcontrol->private_value >> 16; int index = kcontrol->private_value & 0xffff; unsigned int cfud; @@ -239,6 +242,11 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, snd_soc_write(codec, STA32X_CFUD, cfud); snd_soc_write(codec, STA32X_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA32X_COEF_COUNT); i++) + sta32x->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); for (i = 0; i < 3 * numcoef; i++) snd_soc_write(codec, STA32X_B1CF1 + i, ucontrol->value.bytes.data[i]); @@ -252,6 +260,48 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, return 0; } +int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA32X_CFUD */ + cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0; + + for (i = 0; i < STA32X_COEF_COUNT; i++) { + snd_soc_write(codec, STA32X_CFADDR2, i); + snd_soc_write(codec, STA32X_B1CF1, + (sta32x->coef_shadow[i] >> 16) & 0xff); + snd_soc_write(codec, STA32X_B1CF2, + (sta32x->coef_shadow[i] >> 8) & 0xff); + snd_soc_write(codec, STA32X_B1CF3, + (sta32x->coef_shadow[i]) & 0xff); + /* chip documentation does not say if the bits are + * self-clearing, so do it explicitly */ + snd_soc_write(codec, STA32X_CFUD, cfud); + snd_soc_write(codec, STA32X_CFUD, cfud | 0x01); + } + return 0; +} + +int sta32x_cache_sync(struct snd_soc_codec *codec) +{ + unsigned int mute; + int rc; + + if (!codec->cache_sync) + return 0; + + /* mute during register sync */ + mute = snd_soc_read(codec, STA32X_MMUTE); + snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE); + sta32x_sync_coef_shadow(codec); + rc = snd_soc_cache_sync(codec); + snd_soc_write(codec, STA32X_MMUTE, mute); + return rc; +} + #define SINGLE_COEF(xname, index) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = sta32x_coefficient_info, \ @@ -661,7 +711,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec, return ret; } - snd_soc_cache_sync(codec); + sta32x_cache_sync(codec); } /* Power up to mute */ @@ -790,6 +840,17 @@ static int sta32x_probe(struct snd_soc_codec *codec) STA32X_CxCFG_OM_MASK, 2 << STA32X_CxCFG_OM_SHIFT); + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta32x->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta32x->coef_shadow[i] = 0x7fffff; + sta32x->coef_shadow[55] = 0x5a9df7; + sta32x->coef_shadow[56] = 0x7fffff; + sta32x->coef_shadow[59] = 0x7fffff; + sta32x->coef_shadow[60] = 0x400000; + sta32x->coef_shadow[61] = 0x400000; + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h index b97ee5a..d8e32a6 100644 --- a/sound/soc/codecs/sta32x.h +++ b/sound/soc/codecs/sta32x.h @@ -19,6 +19,7 @@ /* STA326 register addresses */ #define STA32X_REGISTER_COUNT 0x2d +#define STA32X_COEF_COUNT 62 #define STA32X_CONFA 0x00 #define STA32X_CONFB 0x01 diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7e5ec03..a7c9ae1 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + codec->cache_sync = 1; break; } codec->dapm.bias_level = level; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a950471..3a629d0 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -190,6 +190,9 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 ioctl; + if (wm8753->dai_func == ucontrol->value.integer.value[0]) + return 0; + if (codec->active) return -EBUSY; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 91d3c6d..53edd9a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1973,7 +1973,7 @@ static int wm8962_reset(struct snd_soc_codec *codec) static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0); static const DECLARE_TLV_DB_SCALE(mixin_tlv, -1500, 300, 0); static const unsigned int mixinpga_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(5), 0, 1, TLV_DB_SCALE_ITEM(0, 600, 0), 2, 2, TLV_DB_SCALE_ITEM(1300, 1300, 0), 3, 4, TLV_DB_SCALE_ITEM(1800, 200, 0), @@ -1988,7 +1988,7 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); static const DECLARE_TLV_DB_SCALE(hp_tlv, -700, 100, 0); static const unsigned int classd_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(2), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index eec8e14..d1a142f4 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -512,7 +512,7 @@ static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0); static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0); static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0); static const unsigned int drc_max_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0), 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0), }; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6b73efd..6c29885 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -56,7 +56,7 @@ static int wm8994_retune_mobile_base[] = { static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->control_data; + struct wm8994 *control = codec->control_data; switch (reg) { case WM8994_GPIO_1: @@ -2357,6 +2357,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; lrclk = bclk_rate / params_rate(params); + if (!lrclk) { + dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n", + bclk_rate); + return -EINVAL; + } dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", lrclk, bclk_rate / lrclk); @@ -3030,19 +3035,34 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; struct snd_soc_codec *codec = wm8994->codec; - int reg; + int reg, count; - reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); - if (reg < 0) { - dev_err(codec->dev, "Failed to read mic detect status: %d\n", - reg); - return IRQ_NONE; - } + /* We may occasionally read a detection without an impedence + * range being provided - if that happens loop again. + */ + count = 10; + do { + reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); + if (reg < 0) { + dev_err(codec->dev, + "Failed to read mic detect status: %d\n", + reg); + return IRQ_NONE; + } - if (!(reg & WM8958_MICD_VALID)) { - dev_dbg(codec->dev, "Mic detect data not valid\n"); - goto out; - } + if (!(reg & WM8958_MICD_VALID)) { + dev_dbg(codec->dev, "Mic detect data not valid\n"); + goto out; + } + + if (!(reg & WM8958_MICD_STS) || (reg & WM8958_MICD_LVL_MASK)) + break; + + msleep(1); + } while (count--); + + if (count == 0) + dev_warn(codec->dev, "No impedence range reported for jack\n"); #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); @@ -3163,6 +3183,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (wm8994->revision) { case 0: case 1: + case 2: + case 3: wm8994->hubs.dcs_codes_l = -9; wm8994->hubs.dcs_codes_r = -5; break; @@ -3180,9 +3202,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, wm8994_fifo_error, "FIFO error", codec); - wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN, + wm8994_request_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, wm8994_temp_warn, "Thermal warning", codec); - wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT, + wm8994_request_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, wm8994_temp_shut, "Thermal shutdown", codec); ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 3cd35a0..4a398c3 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -807,7 +807,6 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, mdelay(100); /* Normal bias enable & soft start off */ - reg |= WM9081_BIAS_ENA; reg &= ~WM9081_VMID_RAMP; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); @@ -818,7 +817,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_VMID_CONTROL); reg &= ~WM9081_VMID_SEL_MASK; reg |= 0x04; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); @@ -830,14 +829,15 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - /* Startup bias source */ + /* Startup bias source and disable bias */ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg |= WM9081_BIAS_SRC; + reg &= ~WM9081_BIAS_ENA; snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); - /* Disable VMID and biases with soft ramping */ + /* Disable VMID with soft ramping */ reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); + reg &= ~WM9081_VMID_SEL_MASK; reg |= WM9081_VMID_RAMP; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 2b5252c..f94c060 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -177,19 +177,19 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) } static const unsigned int in_tlv[] = { - TLV_DB_RANGE_HEAD(6), + TLV_DB_RANGE_HEAD(3), 0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0), 1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0), 4, 6, TLV_DB_SCALE_ITEM(600, 600, 0), }; static const unsigned int mix_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0), 3, 3, TLV_DB_SCALE_ITEM(0, 0, 0), }; static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); static const unsigned int spkboost_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(2), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84f33d4..48e61e9 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -40,7 +40,7 @@ static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0); static const unsigned int spkboost_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(2), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0268cf9..83c4bd5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -694,6 +694,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; + sysfs_attr_init(&dev_attr->attr); dev_attr->attr.name = "statistics"; dev_attr->attr.mode = S_IRUGO; dev_attr->show = fsl_sysfs_ssi_show; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 31af405..ae49f1c 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } if (strcasecmp(sprop, "i2s-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->dai_format = + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; @@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } machine_data->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->dai_format = + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "lj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "lj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "rj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "ac97-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->dai_format = + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "ac97-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->dai_format = + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else { diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index f75e439..ad9ac42 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -9,6 +9,7 @@ #include "../codecs/wm8994.h" #include <sound/pcm_params.h> +#include <linux/module.h> /* * Default CFG switch settings to use this driver: diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 85bf541..4b8e354 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -191,7 +191,7 @@ static int speyside_late_probe(struct snd_soc_card *card) snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5d3685..a25fa63 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -709,6 +709,12 @@ int snd_soc_resume(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); int i, ac97_control = 0; + /* If the initialization of this soc device failed, there is no codec + * associated with it. Just bail out in this case. + */ + if (list_empty(&card->codec_dev_list)) + return 0; + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 60f65ac..ab23869 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -765,10 +765,61 @@ static void usb_mixer_elem_free(struct snd_kcontrol *kctl) * interface to ALSA control for feature/mixer units */ +/* volume control quirks */ +static void volume_control_quirks(struct usb_mixer_elem_info *cval, + struct snd_kcontrol *kctl) +{ + switch (cval->mixer->chip->usb_id) { + case USB_ID(0x0471, 0x0101): + case USB_ID(0x0471, 0x0104): + case USB_ID(0x0471, 0x0105): + case USB_ID(0x0672, 0x1041): + /* quirk for UDA1321/N101. + * note that detection between firmware 2.1.1.7 (N101) + * and later 2.1.1.21 is not very clear from datasheets. + * I hope that the min value is -15360 for newer firmware --jk + */ + if (!strcmp(kctl->id.name, "PCM Playback Volume") && + cval->min == -15616) { + snd_printk(KERN_INFO + "set volume quirk for UDA1321/N101 chip\n"); + cval->max = -256; + } + break; + + case USB_ID(0x046d, 0x09a4): + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO + "set volume quirk for QuickCam E3500\n"); + cval->min = 6080; + cval->max = 8768; + cval->res = 192; + } + break; + + case USB_ID(0x046d, 0x0808): + case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ + case USB_ID(0x046d, 0x0991): + /* Most audio usb devices lie about volume resolution. + * Most Logitech webcams have res = 384. + * Proboly there is some logitech magic behind this number --fishor + */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO + "set resolution quirk: cval->res = 384\n"); + cval->res = 384; + } + break; + + } +} + /* * retrieve the minimum and maximum values for the specified control */ -static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) +static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, + int default_min, struct snd_kcontrol *kctl) { /* for failsafe */ cval->min = default_min; @@ -844,6 +895,9 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) cval->initialized = 1; } + if (kctl) + volume_control_quirks(cval, kctl); + /* USB descriptions contain the dB scale in 1/256 dB unit * while ALSA TLV contains in 1/100 dB unit */ @@ -864,6 +918,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) return 0; } +#define get_min_max(cval, def) get_min_max_with_quirks(cval, def, NULL) /* get a feature/mixer unit info */ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -882,7 +937,7 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ uinfo->value.integer.max = 1; } else { if (!cval->initialized) { - get_min_max(cval, 0); + get_min_max_with_quirks(cval, 0, kcontrol); if (cval->initialized && cval->dBmin >= cval->dBmax) { kcontrol->vd[0].access &= ~(SNDRV_CTL_ELEM_ACCESS_TLV_READ | @@ -1045,9 +1100,6 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, cval->ch_readonly = readonly_mask; } - /* get min/max values */ - get_min_max(cval, 0); - /* if all channels in the mask are marked read-only, make the control * read-only. set_cur_mix_value() will check the mask again and won't * issue write commands to read-only channels. */ @@ -1069,6 +1121,9 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); + /* get min/max values */ + get_min_max_with_quirks(cval, 0, kctl); + switch (control) { case UAC_FU_MUTE: case UAC_FU_VOLUME: @@ -1118,51 +1173,6 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, break; } - /* volume control quirks */ - switch (state->chip->usb_id) { - case USB_ID(0x0471, 0x0101): - case USB_ID(0x0471, 0x0104): - case USB_ID(0x0471, 0x0105): - case USB_ID(0x0672, 0x1041): - /* quirk for UDA1321/N101. - * note that detection between firmware 2.1.1.7 (N101) - * and later 2.1.1.21 is not very clear from datasheets. - * I hope that the min value is -15360 for newer firmware --jk - */ - if (!strcmp(kctl->id.name, "PCM Playback Volume") && - cval->min == -15616) { - snd_printk(KERN_INFO - "set volume quirk for UDA1321/N101 chip\n"); - cval->max = -256; - } - break; - - case USB_ID(0x046d, 0x09a4): - if (!strcmp(kctl->id.name, "Mic Capture Volume")) { - snd_printk(KERN_INFO - "set volume quirk for QuickCam E3500\n"); - cval->min = 6080; - cval->max = 8768; - cval->res = 192; - } - break; - - case USB_ID(0x046d, 0x0808): - case USB_ID(0x046d, 0x0809): - case USB_ID(0x046d, 0x0991): - /* Most audio usb devices lie about volume resolution. - * Most Logitech webcams have res = 384. - * Proboly there is some logitech magic behind this number --fishor - */ - if (!strcmp(kctl->id.name, "Mic Capture Volume")) { - snd_printk(KERN_INFO - "set resolution quirk: cval->res = 384\n"); - cval->res = 384; - } - break; - - } - range = (cval->max - cval->min) / cval->res; /* Are there devices with volume range more than 255? I use a bit more * to be sure. 384 is a resolution magic number found on Logitech diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b61945f..32d2a21 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1633,6 +1633,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Roland GAIA SH-01 */ + USB_DEVICE(0x0582, 0x0111), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "GAIA", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ USB_DEVICE(0x0582, 0x0113), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "BOSS", */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2e5bc73..a3ddac0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -137,12 +137,12 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -ENOMEM; } if (fp->nr_rates > 0) { - rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL); + rate_table = kmemdup(fp->rate_table, + sizeof(int) * fp->nr_rates, GFP_KERNEL); if (!rate_table) { kfree(fp); return -ENOMEM; } - memcpy(rate_table, fp->rate_table, sizeof(int) * fp->nr_rates); fp->rate_table = rate_table; } @@ -224,10 +224,9 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, if (altsd->bNumEndpoints != 1) return -ENXIO; - fp = kmalloc(sizeof(*fp), GFP_KERNEL); + fp = kmemdup(&ua_format, sizeof(*fp), GFP_KERNEL); if (!fp) return -ENOMEM; - memcpy(fp, &ua_format, sizeof(*fp)); fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; |