diff options
Diffstat (limited to 'sound')
38 files changed, 243 insertions, 154 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 89096e8..772901e 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--); + } while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout); if (!timeout) dev_err(&aaci->dev->dev, @@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & SLFR_1TXB) && timeout--); + } while ((v & SLFR_1TXB) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n"); @@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) do { cond_resched(); v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV); - } while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--); + } while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on RX valid\n"); diff --git a/sound/core/jack.c b/sound/core/jack.c index dd4a12d..077a852 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -47,7 +47,7 @@ static int snd_jack_dev_register(struct snd_device *device) int err; snprintf(jack->name, sizeof(jack->name), "%s %s", - card->longname, jack->id); + card->shortname, jack->id); jack->input_dev->name = jack->name; /* Default to the sound card device. */ diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e178366..0a1798e 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1767,7 +1767,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) AFMT_S8 | AFMT_U16_LE | AFMT_U16_BE | AFMT_S32_LE | AFMT_S32_BE | - AFMT_S24_LE | AFMT_S24_LE | + AFMT_S24_LE | AFMT_S24_BE | AFMT_S24_PACKED; params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params) diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index a466443..2fa9299 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin, while (dst_frames1 > 0) { S1 = S2; if (src_frames1-- > 0) { - S1 = *src; + S2 = *src; src += src_step; } if (pos & ~R_MASK) { diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 5b89c088..48b64e6 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -706,7 +706,6 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev) mtp_card->card = card; mtp_card->irq = -1; mtp_card->share_irq = 0; - mtp_card->inmidiport = 0xffffffff; mtp_card->inmidistate = 0; mtp_card->outmidihwport = 0xffffffff; init_timer(&mtp_card->timer); @@ -719,6 +718,8 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev) if (err < 0) goto __error; + mtp_card->inmidiport = mtp_card->num_ports + MTPAV_PIDX_BROADCAST; + err = snd_mtpav_get_ISA(mtp_card); if (err < 0) goto __error; diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 57d9f15..38931f2 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -847,23 +847,23 @@ static int __init AtaIrqInit(void) of events. So all we need to keep the music playing is to provide the sound hardware with new data upon an interrupt from timer A. */ - mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ - mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ - mfp.tim_ct_a = 8; /* Turn on event counting. */ + st_mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ + st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ + st_mfp.tim_ct_a = 8; /* Turn on event counting. */ /* Register interrupt handler. */ if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", AtaInterrupt)) return 0; - mfp.int_en_a |= 0x20; /* Turn interrupt on. */ - mfp.int_mk_a |= 0x20; + st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */ + st_mfp.int_mk_a |= 0x20; return 1; } #ifdef MODULE static void AtaIrqCleanUp(void) { - mfp.tim_ct_a = 0; /* stop timer */ - mfp.int_en_a &= ~0x20; /* turn interrupt off */ + st_mfp.tim_ct_a = 0; /* stop timer */ + st_mfp.int_en_a &= ~0x20; /* turn interrupt off */ free_irq(IRQ_MFP_TIMA, AtaInterrupt); } #endif /* MODULE */ @@ -1599,7 +1599,7 @@ static int __init dmasound_atari_init(void) is_falcon = 0; } else return -ENODEV; - if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0) + if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0) return dmasound_init(); else { printk("DMA sound driver: Timer A interrupt already in use\n"); diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 6e3a184..82b9bdd 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -744,8 +744,8 @@ config SND_VIRTUOSO select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the - Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X and - HDAV1.3 (Deluxe). + Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X. + Support for the HDAV1.3 (Deluxe) is very experimental. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 3f00ddf..c7c54e7 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); static struct pci_device_id snd_aw2_ids[] = { - {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, + {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, 0, 0, 0}, {0} }; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 7958006..101a1c1 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102, .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]", diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b7bba7d..d03f992 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -487,7 +487,6 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, { struct hda_bus *bus; int err; - char qname[8]; static struct snd_device_ops dev_ops = { .dev_register = snd_hda_bus_dev_register, .dev_free = snd_hda_bus_dev_free, @@ -517,10 +516,12 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, mutex_init(&bus->cmd_mutex); INIT_LIST_HEAD(&bus->codec_list); - snprintf(qname, sizeof(qname), "hda%d", card->number); - bus->workq = create_workqueue(qname); + snprintf(bus->workq_name, sizeof(bus->workq_name), + "hd-audio%d", card->number); + bus->workq = create_singlethread_workqueue(bus->workq_name); if (!bus->workq) { - snd_printk(KERN_ERR "cannot create workqueue %s\n", qname); + snd_printk(KERN_ERR "cannot create workqueue %s\n", + bus->workq_name); kfree(bus); return -ENOMEM; } @@ -3087,6 +3088,16 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare); +int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, + struct hda_multi_out *mout) +{ + mutex_lock(&codec->spdif_mutex); + cleanup_dig_out_stream(codec, mout->dig_out_nid); + mutex_unlock(&codec->spdif_mutex); + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup); + /* * release the digital out */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5810ef5..09a332a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -614,6 +614,7 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; + char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ /* assigned PCMs */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 300ab40..4ae51dc 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -175,7 +175,7 @@ static int reconfig_codec(struct hda_codec *codec) err = snd_hda_codec_build_controls(codec); if (err < 0) return err; - return 0; + return snd_card_register(codec->bus->card); } /* @@ -277,18 +277,19 @@ static ssize_t init_verbs_store(struct device *dev, { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - char *p; - struct hda_verb verb, *v; + struct hda_verb *v; + int nid, verb, param; - verb.nid = simple_strtoul(buf, &p, 0); - verb.verb = simple_strtoul(p, &p, 0); - verb.param = simple_strtoul(p, &p, 0); - if (!verb.nid || !verb.verb || !verb.param) + if (sscanf(buf, "%i %i %i", &nid, &verb, ¶m) != 3) + return -EINVAL; + if (!nid || !verb) return -EINVAL; v = snd_array_new(&codec->init_verbs); if (!v) return -ENOMEM; - *v = verb; + v->nid = nid; + v->verb = verb; + v->param = param; return count; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 11e791b..5e909e0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1947,16 +1947,13 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) return 0; } -static int azx_resume_early(struct pci_dev *pci) -{ - return pci_restore_state(pci); -} - static int azx_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); if (pci_enable_device(pci) < 0) { printk(KERN_ERR "hda-intel: pci_enable_device failed, " "disabling device\n"); @@ -2098,6 +2095,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01), /* including bogus ALC268 in slot#2 that conflicts with ALC888 */ SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), + /* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */ + SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03), {} }; @@ -2468,7 +2467,6 @@ static struct pci_driver driver = { .remove = __devexit_p(azx_remove), #ifdef CONFIG_PM .suspend = azx_suspend, - .resume_early = azx_resume_early, .resume = azx_resume, #endif }; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 1dd8716..44f189c 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -251,6 +251,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream); +int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, + struct hda_multi_out *mout); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, struct snd_pcm_substream *substream, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7ca66d6..144b852 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -399,7 +399,8 @@ static void print_conn_list(struct snd_info_buffer *buffer, { int c, curr = -1; - if (conn_len > 1 && wid_type != AC_WID_AUD_MIX) + if (conn_len > 1 && wid_type != AC_WID_AUD_MIX && + wid_type != AC_WID_VOL_KNB) curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); snd_iprintf(buffer, " Connection: %d\n", conn_len); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2e7371e..e486123 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -275,6 +275,14 @@ static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, format, substream); } +static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ad198x_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + /* * Analog capture */ @@ -333,7 +341,8 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = { .ops = { .open = ad198x_dig_playback_pcm_open, .close = ad198x_dig_playback_pcm_close, - .prepare = ad198x_dig_playback_pcm_prepare + .prepare = ad198x_dig_playback_pcm_prepare, + .cleanup = ad198x_dig_playback_pcm_cleanup }, }; @@ -1885,8 +1894,8 @@ static hda_nid_t ad1988_capsrc_nids[3] = { #define AD1988_SPDIF_OUT_HDMI 0x0b #define AD1988_SPDIF_IN 0x07 -static hda_nid_t ad1989b_slave_dig_outs[2] = { - AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI +static hda_nid_t ad1989b_slave_dig_outs[] = { + AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0 }; static struct hda_input_mux ad1988_6stack_capture_source = { diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 75de40a..0177ef8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -347,6 +347,7 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, &spec->cur_mux[adc_idx]); } +#ifdef CONFIG_SND_JACK static int conexant_add_jack(struct hda_codec *codec, hda_nid_t nid, int type) { @@ -394,7 +395,6 @@ static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) static int conexant_init_jacks(struct hda_codec *codec) { -#ifdef CONFIG_SND_JACK struct conexant_spec *spec = codec->spec; int i; @@ -422,10 +422,19 @@ static int conexant_init_jacks(struct hda_codec *codec) ++hv; } } -#endif return 0; } +#else +static inline void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ +} + +static inline int conexant_init_jacks(struct hda_codec *codec) +{ + return 0; +} +#endif static int conexant_init(struct hda_codec *codec) { @@ -1566,6 +1575,7 @@ static struct snd_pci_quirk cxt5047_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6700", CXT5047_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), {} }; diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 3564f4e..fcc77fe 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -49,11 +49,6 @@ static struct hda_verb pinout_enable_verb[] = { {} /* terminator */ }; -static struct hda_verb pinout_disable_verb[] = { - {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00}, - {} -}; - static struct hda_verb unsolicited_response_verb[] = { {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | INTEL_HDMI_EVENT_TAG}, @@ -248,10 +243,6 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid, static void hdmi_enable_output(struct hda_codec *codec) { - /* Enable Audio InfoFrame Transmission */ - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); - snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_BEST); /* Unmute */ if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, PIN_NID, 0, @@ -260,17 +251,24 @@ static void hdmi_enable_output(struct hda_codec *codec) snd_hda_sequence_write(codec, pinout_enable_verb); } -static void hdmi_disable_output(struct hda_codec *codec) +/* + * Enable Audio InfoFrame Transmission + */ +static void hdmi_start_infoframe_trans(struct hda_codec *codec) { - snd_hda_sequence_write(codec, pinout_disable_verb); - if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, PIN_NID, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_BEST); +} - /* - * FIXME: noises may arise when playing music after reloading the - * kernel module, until the next X restart or monitor repower. - */ +/* + * Disable Audio InfoFrame Transmission + */ +static void hdmi_stop_infoframe_trans(struct hda_codec *codec) +{ + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_DISABLE); } static int hdmi_get_channel_count(struct hda_codec *codec) @@ -368,11 +366,16 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec, struct hdmi_audio_infoframe *ai) { u8 *params = (u8 *)ai; + u8 sum = 0; int i; hdmi_debug_dip_size(codec); hdmi_clear_dip_buffers(codec); /* be paranoid */ + for (i = 0; i < sizeof(ai); i++) + sum += params[i]; + ai->checksum = - sum; + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); for (i = 0; i < sizeof(ai); i++) hdmi_write_dip_byte(codec, PIN_NID, params[i]); @@ -419,14 +422,18 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, /* * CA defaults to 0 for basic stereo audio */ - if (!eld->eld_ver) - return 0; - if (!eld->spk_alloc) - return 0; if (channels <= 2) return 0; /* + * 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; + + /* * expand ELD's speaker allocation mask * * ELD tells the speaker mask in a compact(paired) form, @@ -485,6 +492,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hdmi_setup_channel_mapping(codec, &ai); hdmi_fill_audio_infoframe(codec, &ai); + hdmi_start_infoframe_trans(codec); } @@ -562,7 +570,7 @@ static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, { struct intel_hdmi_spec *spec = codec->spec; - hdmi_disable_output(codec); + hdmi_stop_infoframe_trans(codec); return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -582,8 +590,6 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, substream); - hdmi_enable_output(codec); - return 0; } @@ -628,8 +634,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec) static int intel_hdmi_init(struct hda_codec *codec) { - /* disable audio output as early as possible */ - hdmi_disable_output(codec); + hdmi_enable_output(codec); snd_hda_sequence_write(codec, unsolicited_response_verb); @@ -679,6 +684,7 @@ static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, + { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ }; @@ -687,6 +693,7 @@ MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_ALIAS("snd-hda-codec-id:80862801"); MODULE_ALIAS("snd-hda-codec-id:80862802"); MODULE_ALIAS("snd-hda-codec-id:80862803"); +MODULE_ALIAS("snd-hda-codec-id:80862804"); MODULE_ALIAS("snd-hda-codec-id:10951392"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 82dd084..6c26afc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1037,6 +1037,7 @@ do_sku: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0272: case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: @@ -1065,6 +1066,7 @@ do_sku: case 0x10ec0882: case 0x10ec0883: case 0x10ec0885: + case 0x10ec0887: case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); @@ -7012,12 +7014,15 @@ static int patch_alc882(struct hda_codec *codec) break; case 0x106b1000: /* iMac 24 */ case 0x106b2800: /* AppleTV */ + case 0x106b3e00: /* iMac 24 Aluminium */ board_config = ALC885_IMAC24; break; + case 0x106b00a0: /* MacBookPro3,1 - Another revision */ case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ + case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; default: @@ -8465,6 +8470,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", + ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), @@ -8474,10 +8481,12 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), @@ -8512,6 +8521,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550", + ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530", ALC888_FUJITSU_XA3530), @@ -8526,6 +8537,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} @@ -10545,6 +10557,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x170b, "HP xw*", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c39deeb..3bc4276 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -81,6 +81,7 @@ enum { enum { STAC_92HD83XXX_REF, + STAC_92HD83XXX_PWR_REF, STAC_92HD83XXX_MODELS }; @@ -334,7 +335,7 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { }; static unsigned int stac92hd83xxx_pwr_mapping[4] = { - 0x03, 0x0c, 0x10, 0x40, + 0x03, 0x0c, 0x20, 0x40, }; static hda_nid_t stac92hd83xxx_amp_nids[1] = { @@ -841,10 +842,6 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { }; static struct hda_verb stac92hd83xxx_core_init[] = { - /* start of config #1 */ - { 0xe, AC_VERB_SET_CONNECT_SEL, 0x3}, - - /* start of config #2 */ { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, @@ -885,8 +882,8 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { static struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* unmute and set max the selector */ - { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f }, + /* mute the master volume */ + { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, {} }; @@ -1138,6 +1135,8 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { }; static struct snd_kcontrol_new stac925x_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT), STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), @@ -1736,10 +1735,12 @@ static unsigned int ref92hd83xxx_pin_configs[14] = { static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, + [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = "ref", + [STAC_92HD83XXX_PWR_REF] = "mic-ref", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1798,9 +1799,13 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2, "HP dv5", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, - "HP dv7", STAC_HP_M4), + "HP dv7", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7, + "HP dv4", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, "HP dv7", STAC_HP_M4), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600, + "HP dv5", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603, "HP dv5", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, @@ -2437,6 +2442,14 @@ static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static int stac92xx_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct sigmatel_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + /* * Analog capture callbacks @@ -2481,7 +2494,8 @@ static struct hda_pcm_stream stac92xx_pcm_digital_playback = { .ops = { .open = stac92xx_dig_playback_pcm_open, .close = stac92xx_dig_playback_pcm_close, - .prepare = stac92xx_dig_playback_pcm_prepare + .prepare = stac92xx_dig_playback_pcm_prepare, + .cleanup = stac92xx_dig_playback_pcm_cleanup }, }; @@ -2536,6 +2550,8 @@ static int stac92xx_build_pcms(struct hda_codec *codec) info->name = "STAC92xx Analog"; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs; @@ -3573,13 +3589,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out err = stac92xx_auto_fill_dac_nids(codec); if (err < 0) return err; + err = stac92xx_auto_create_multi_out_ctls(codec, + &spec->autocfg); + if (err < 0) + return err; } - err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg); - - if (err < 0) - return err; - /* setup analog beep controls */ if (spec->anabeep_nid > 0) { err = stac92xx_auto_create_beep_ctls(codec, @@ -4753,7 +4768,9 @@ static struct hda_input_mux stac92hd83xxx_dmux = { static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; + int num_dacs; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4772,15 +4789,16 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; - spec->init = stac92hd83xxx_core_init; - switch (codec->vendor_id) { - case 0x111d7605: - break; - default: - spec->num_pwrs--; - spec->init++; /* switch to config #2 */ - } + /* set port 0xe to select the last DAC + */ + num_dacs = snd_hda_get_connections(codec, 0x0e, + conn, STAC92HD83_DAC_COUNT + 1) - 1; + + snd_hda_codec_write_cache(codec, 0xe, 0, + AC_VERB_SET_CONNECT_SEL, num_dacs); + + spec->init = stac92hd83xxx_core_init; spec->mixer = stac92hd83xxx_mixer; spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids); @@ -4806,6 +4824,15 @@ again: return err; } + switch (codec->vendor_id) { + case 0x111d7604: + case 0x111d7605: + if (spec->board_config == STAC_92HD83XXX_PWR_REF) + break; + spec->num_pwrs = 0; + break; + } + err = stac92xx_parse_auto_config(codec, 0x1d, 0); if (!err) { if (spec->board_config < 0) { @@ -4962,7 +4989,7 @@ again: case STAC_DELL_M4_3: spec->num_dmics = 1; spec->num_smuxes = 0; - spec->num_dmuxes = 0; + spec->num_dmuxes = 1; break; default: spec->num_dmics = STAC92HD71BXX_NUM_DMICS; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 19d3391..e900cdc 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -617,7 +617,7 @@ static int snd_intel8x0_ali_codec_semaphore(struct intel8x0 *chip) int time = 100; if (chip->buggy_semaphore) return 0; /* just ignore ... */ - while (time-- && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY)) + while (--time && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY)) udelay(1); if (! time && ! chip->in_ac97_init) snd_printk(KERN_WARNING "ali_codec_semaphore timeout\n"); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index e9e829e..6c870c1 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -26,7 +26,7 @@ * SPI 0 -> 1st PCM1796 (front) * SPI 1 -> 2nd PCM1796 (surround) * SPI 2 -> 3rd PCM1796 (center/LFE) - * SPI 4 -> 4th PCM1796 (back) and EEPROM self-destruct (do not use!) + * SPI 4 -> 4th PCM1796 (back) * * GPIO 2 -> M0 of CS5381 * GPIO 3 -> M1 of CS5381 @@ -207,12 +207,6 @@ static void xonar_gpio_changed(struct oxygen *chip); static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec, u8 reg, u8 value) { - /* - * We don't want to do writes on SPI 4 because the EEPROM, which shares - * the same pin, might get confused and broken. We'd better take care - * that the driver works with the default register values ... - */ -#if 0 /* maps ALSA channel pair number to SPI output */ static const u8 codec_map[4] = { 0, 1, 2, 4 @@ -223,7 +217,6 @@ static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec, (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, (reg << 8) | value); -#endif } static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec, @@ -683,7 +676,7 @@ static void xonar_hdav_uart_input(struct oxygen *chip) if (chip->uart_input_count >= 2 && chip->uart_input[chip->uart_input_count - 2] == 'O' && chip->uart_input[chip->uart_input_count - 1] == 'K') { - printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:"); + printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:\n"); print_hex_dump_bytes("", DUMP_PREFIX_OFFSET, chip->uart_input, chip->uart_input_count); chip->uart_input_count = 0; @@ -757,9 +750,6 @@ static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0); static int xonar_d2_control_filter(struct snd_kcontrol_new *template) { - if (!strncmp(template->name, "Master Playback ", 16)) - /* disable volume/mute because they would require SPI writes */ - return 1; if (!strncmp(template->name, "CD Capture ", 11)) /* CD in is actually connected to the video in pin */ template->private_value ^= AC97_CD ^ AC97_VIDEO; @@ -850,8 +840,9 @@ static const struct oxygen_model model_xonar_d2 = { .dac_volume_min = 0x0f, .dac_volume_max = 0xff, .misc_flags = OXYGEN_MISC_MIDI, - .function_flags = OXYGEN_FUNCTION_SPI, - .dac_i2s_format = OXYGEN_I2S_FORMAT_I2S, + .function_flags = OXYGEN_FUNCTION_SPI | + OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; @@ -908,6 +899,7 @@ static const struct oxygen_model model_xonar_hdav = { .dac_channels = 8, .dac_volume_min = 0x0f, .dac_volume_max = 0xff, + .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_2WIRE, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 84131a9..69d87de 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -97,12 +97,12 @@ struct pcxhr_mgr { int capture_chips; int fw_file_set; int firmware_num; - int is_hr_stereo:1; - int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ - int board_has_analog:1; /* if 0 the board is digital only */ - int board_has_mic:1; /* if 1 the board has microphone input */ - int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ - int mono_capture:1; /* if 1 the board does mono capture */ + unsigned int is_hr_stereo:1; + unsigned int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ + unsigned int board_has_analog:1; /* if 0 the board is digital only */ + unsigned int board_has_mic:1; /* if 1 the board has microphone input */ + unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ + unsigned int mono_capture:1; /* if 1 the board does mono capture */ struct snd_dma_buffer hostport; diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 8f9e385..ff32111 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -477,7 +477,7 @@ static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream) card->dma_start_bus_addr[SND_PS3_CH_R] = runtime->dma_addr + (runtime->dma_bytes / 2); - pr_debug("%s: vaddr=%p bus=%#lx\n", __func__, + pr_debug("%s: vaddr=%p bus=%#llx\n", __func__, card->dma_start_vaddr[SND_PS3_CH_L], card->dma_start_bus_addr[SND_PS3_CH_L]); @@ -1030,7 +1030,7 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) pr_info("%s: nullbuffer alloc failed\n", __func__); goto clean_preallocate; } - pr_debug("%s: null vaddr=%p dma=%#lx\n", __func__, + pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__, the_card.null_buffer_start_vaddr, the_card.null_buffer_start_dma_addr); /* set default sample rate/word width */ diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 1fac5ef..3dcdc4e 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -44,8 +44,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <mach/hardware.h> - #include "atmel-pcm.h" diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index c5d6790..ff0054b 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino@endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index a828746..391135f 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino@endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749..aea0cb7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -165,10 +165,13 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0x01; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; unsigned short val, val_mask; int ret; struct snd_soc_dapm_path *path; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e3989d4..35d9975 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -3,7 +3,7 @@ * * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com> + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6c21b50..77620ab 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1451,7 +1451,14 @@ static const struct snd_soc_dai wm8753_all_dai[] = { }, }; -struct snd_soc_dai wm8753_dai[2]; +struct snd_soc_dai wm8753_dai[] = { + { + .name = "WM8753 DAI 0", + }, + { + .name = "WM8753 DAI 1", + }, +}; EXPORT_SYMBOL_GPL(wm8753_dai); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5b5afc1..a5731fa 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -2,8 +2,7 @@ * wm8990.c -- WM8990 ALSA Soc Audio driver * * Copyright 2008 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -177,7 +176,9 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; int ret; u16 val; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index bcec3f6..acf39a6 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -183,16 +183,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = { }; /** - * mpc8610_hpcd_machine: ASoC machine data - */ -static struct snd_soc_card mpc8610_hpcd_machine = { - .probe = mpc8610_hpcd_machine_probe, - .remove = mpc8610_hpcd_machine_remove, - .name = "MPC8610 HPCD", - .num_links = 1, -}; - -/** * mpc8610_hpcd_probe: OF probe function for the fabric driver * * This function gets called when an SSI node is found in the device tree. @@ -455,7 +445,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */ machine_data->dai.ops = &mpc8610_hpcd_ops; - mpc8610_hpcd_machine.dai_link = &machine_data->dai; + machine_data->machine.probe = mpc8610_hpcd_machine_probe; + machine_data->machine.remove = mpc8610_hpcd_machine_remove; + machine_data->machine.name = "MPC8610 HPCD"; + machine_data->machine.num_links = 1; + machine_data->machine.dai_link = &machine_data->dai; /* Allocate a new audio platform device structure */ sound_device = platform_device_alloc("soc-audio", -1); @@ -465,7 +459,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, goto error; } - machine_data->sound_devdata.card = &mpc8610_hpcd_machine; + machine_data->sound_devdata.card = &machine_data->machine; machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270; machine_data->machine.platform = &fsl_soc_platform; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index ec5e18a..05dd5ab 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -302,6 +302,10 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->spcr1 |= RINTM(3); regs->rcr2 |= RFIG; regs->xcr2 |= XFIG; + if (cpu_is_omap2430() || cpu_is_omap34xx()) { + regs->xccr = DXENDLY(1) | XDMAEN; + regs->rccr = RFULL_CYCLE | RDMAEN; + } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b0362df..dd3bb29 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -175,9 +175,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_pcm_runtime *runtime = substream->runtime; struct omap_runtime_data *prtd = runtime->private_data; + unsigned long flags; int ret = 0; - spin_lock_irq(&prtd->lock); + spin_lock_irqsave(&prtd->lock, flags); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -195,7 +196,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) default: ret = -EINVAL; } - spin_unlock_irq(&prtd->lock); + spin_unlock_irqrestore(&prtd->lock, flags); return ret; } diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index ad97836..e226fa7 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -91,7 +91,7 @@ static struct snd_soc_dai_link sdp3430_dai = { }; /* Audio machine driver */ -static struct snd_soc_machine snd_soc_machine_sdp3430 = { +static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", .platform = &omap_soc_platform, .dai_link = &sdp3430_dai, @@ -100,7 +100,7 @@ static struct snd_soc_machine snd_soc_machine_sdp3430 = { /* Audio subsystem */ static struct snd_soc_device sdp3430_snd_devdata = { - .machine = &snd_soc_machine_sdp3430, + .card = &snd_soc_sdp3430, .codec_dev = &soc_codec_dev_twl4030, }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55fdb4a..ec3f8bb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1385,7 +1385,10 @@ int snd_soc_init_card(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS - if (ac97) { + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. + */ + if (ac97 && strcmp(codec->name, "AC97") != 0) { ret = soc_ac97_dev_register(codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c709b95..19e3745 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2524,7 +2524,6 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform * build the rate table and bitmap flags */ int r, idx; - unsigned int nonzero_rates = 0; fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); if (fp->rate_table == NULL) { @@ -2532,24 +2531,27 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform return -1; } - fp->nr_rates = nr_rates; - fp->rate_min = fp->rate_max = combine_triple(&fmt[8]); + fp->nr_rates = 0; + fp->rate_min = fp->rate_max = 0; for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) { unsigned int rate = combine_triple(&fmt[idx]); + if (!rate) + continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ if (rate == 48000 && nr_rates == 1 && - chip->usb_id == USB_ID(0x0d8c, 0x0201) && + (chip->usb_id == USB_ID(0x0d8c, 0x0201) || + chip->usb_id == USB_ID(0x0d8c, 0x0102)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; - fp->rate_table[r] = rate; - nonzero_rates |= rate; - if (rate < fp->rate_min) + fp->rate_table[fp->nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) fp->rate_min = rate; - else if (rate > fp->rate_max) + if (!fp->rate_max || rate > fp->rate_max) fp->rate_max = rate; fp->rates |= snd_pcm_rate_to_rate_bit(rate); + fp->nr_rates++; } - if (!nonzero_rates) { + if (!fp->nr_rates) { hwc_debug("All rates were zero. Skipping format!\n"); return -1; } @@ -2966,6 +2968,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); usb_set_interface(chip->dev, fp->iface, 0); init_usb_pitch(chip->dev, fp->iface, alts, fp); init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max); diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 320641a..26bad37 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1625,6 +1625,7 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi, } ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK; + ep_info.out_interval = 0; ep_info.out_cables = endpoint->out_cables & 0x5555; err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]); if (err < 0) |
