diff options
Diffstat (limited to 'sound')
45 files changed, 652 insertions, 292 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 656e474dca47..91acc9a243ec 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -863,7 +863,6 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci) struct snd_ac97 *ac97; int ret; - writel(0, aaci->base + AC97_POWERDOWN); /* * Assert AACIRESET for 2us */ @@ -1047,7 +1046,11 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) writel(0x1fff, aaci->base + AACI_INTCLR); writel(aaci->maincr, aaci->base + AACI_MAINCR); - + /* + * Fix: ac97 read back fail errors by reading + * from any arbitrary aaci register. + */ + readl(aaci->base + AACI_CSCH1); ret = aaci_probe_ac97(aaci); if (ret) goto out; diff --git a/sound/core/timer.c b/sound/core/timer.c index 73943651caed..5040c7b862fe 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1160,6 +1160,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, { struct snd_timer_user *tu = timeri->callback_data; struct snd_timer_tread r1; + unsigned long flags; if (event >= SNDRV_TIMER_EVENT_START && event <= SNDRV_TIMER_EVENT_PAUSE) @@ -1169,9 +1170,9 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, r1.event = event; r1.tstamp = *tstamp; r1.val = resolution; - spin_lock(&tu->qlock); + spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); - spin_unlock(&tu->qlock); + spin_unlock_irqrestore(&tu->qlock, flags); kill_fasync(&tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index fff62cc8607c..971a84a4fa77 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -70,7 +70,7 @@ static int snd_ak4113_dev_free(struct snd_device *device) } int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, - ak4113_write_t *write, const unsigned char pgm[5], + ak4113_write_t *write, const unsigned char *pgm, void *private_data, struct ak4113 **r_ak4113) { struct ak4113 *chip; diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c index cafc3a7316a8..ff18286fef9d 100644 --- a/sound/isa/sb/es968.c +++ b/sound/isa/sb/es968.c @@ -93,7 +93,7 @@ static int __devinit snd_card_es968_pnp(int dev, struct snd_card_es968 *acard, return err; } port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 1); + dma8[dev] = pnp_dma(pdev, 0); irq[dev] = pnp_irq(pdev, 0); return 0; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 8dab82d7d19d..668a5ec04499 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2184,10 +2184,9 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, goto ctl_error; #endif - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); + err = snd_card_register(card); + if (err < 0) goto ctl_error; - } snd_printk(KERN_INFO "Card registered: %s\n", card->longname); pci_set_drvdata(pci, chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4bb90675f70f..cec68152dcb1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2272,6 +2272,8 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB), + SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} }; @@ -2362,6 +2364,7 @@ static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ + SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */ {} }; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e6d1bdff1b6e..e9fdfc4b1c57 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -519,14 +519,6 @@ static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) ad198x_power_eapd(codec); return 0; } - -static int ad198x_resume(struct hda_codec *codec) -{ - ad198x_init(codec); - snd_hda_codec_resume_amp(codec); - snd_hda_codec_resume_cache(codec); - return 0; -} #endif static struct hda_codec_ops ad198x_patch_ops = { @@ -539,7 +531,6 @@ static struct hda_codec_ops ad198x_patch_ops = { #endif #ifdef SND_HDA_NEEDS_RESUME .suspend = ad198x_suspend, - .resume = ad198x_resume, #endif .reboot_notify = ad198x_shutup, }; @@ -1896,6 +1887,14 @@ static int patch_ad1981(struct hda_codec *codec) case AD1981_THINKPAD: spec->mixers[0] = ad1981_thinkpad_mixers; spec->input_mux = &ad1981_thinkpad_capture_source; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); break; case AD1981_TOSHIBA: spec->mixers[0] = ad1981_hp_mixers; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 7de782a5b8f4..350ee8ac4153 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -766,7 +766,7 @@ static int build_input(struct hda_codec *codec) for (n = 0; n < AUTO_PIN_LAST; n++) { if (!spec->adc_nid[n]) continue; - err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[i]); + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 61682e1d09da..56e52071c769 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1195,9 +1195,10 @@ static int patch_cxt5045(struct hda_codec *codec) switch (codec->subsystem_id >> 16) { case 0x103c: + case 0x1631: case 0x1734: - /* HP & Fujitsu-Siemens laptops have really bad sound over 0dB - * on NID 0x17. Fix max PCM level to 0 dB + /* HP, Packard Bell, & Fujitsu-Siemens laptops have really bad + * sound over 0dB on NID 0x17. Fix max PCM level to 0 dB * (originally it has 0x2b steps with 0dB offset 0x14) */ snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, @@ -2842,6 +2843,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", 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(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9a23444e9e7a..7404dba16f83 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -230,6 +230,7 @@ enum { ALC888_ACER_ASPIRE_7730G, ALC883_MEDION, ALC883_MEDION_MD2, + ALC883_MEDION_WIM2160, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, @@ -1389,22 +1390,31 @@ struct alc_fixup { static void alc_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, - const struct alc_fixup *fix) + const struct alc_fixup *fix, + int pre_init) { const struct alc_pincfg *cfg; quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); if (!quirk) return; - fix += quirk->value; cfg = fix->pins; - if (cfg) { + if (pre_init && cfg) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n", + codec->chip_name, quirk->name); +#endif for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - if (fix->verbs) + if (!pre_init && fix->verbs) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n", + codec->chip_name, quirk->name); +#endif add_verb(codec->spec, fix->verbs); + } } static int alc_read_coef_idx(struct hda_codec *codec, @@ -1621,6 +1631,11 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { */ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Route to built-in subwoofer as well as speakers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Bias voltage on for external mic port */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, /* Front Mic: set to PIN_IN (empty by default) */ @@ -1632,10 +1647,12 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { /* Enable speaker output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, /* Enable headphone output */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; @@ -4126,7 +4143,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), - SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), + SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG), @@ -4801,6 +4818,25 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) } } +static void alc880_auto_init_input_src(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int c; + + for (c = 0; c < spec->num_adc_nids; c++) { + unsigned int mux_idx; + const struct hda_input_mux *imux; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; + if (!imux->num_items && mux_idx > 0) + imux = &spec->input_mux[0]; + if (imux) + snd_hda_codec_write(codec, spec->adc_nids[c], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[0].index); + } +} + /* parse the BIOS configuration and set up the alc_spec */ /* return 1 if successful, 0 if the proper config is not found, * or a negative error code @@ -4879,6 +4915,7 @@ static void alc880_auto_init(struct hda_codec *codec) alc880_auto_init_multi_out(codec); alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); + alc880_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -4984,6 +5021,70 @@ static void set_capture_mixer(struct hda_codec *codec) } } +/* fill adc_nids (and capsrc_nids) containing all active input pins */ +static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, + int num_nids) +{ + struct alc_spec *spec = codec->spec; + int n; + hda_nid_t fallback_adc = 0, fallback_cap = 0; + + for (n = 0; n < num_nids; n++) { + hda_nid_t adc, cap; + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int nconns, i, j; + + adc = nids[n]; + if (get_wcaps_type(get_wcaps(codec, adc)) != AC_WID_AUD_IN) + continue; + cap = adc; + nconns = snd_hda_get_connections(codec, cap, conn, + ARRAY_SIZE(conn)); + if (nconns == 1) { + cap = conn[0]; + nconns = snd_hda_get_connections(codec, cap, conn, + ARRAY_SIZE(conn)); + } + if (nconns <= 0) + continue; + if (!fallback_adc) { + fallback_adc = adc; + fallback_cap = cap; + } + for (i = 0; i < AUTO_PIN_LAST; i++) { + hda_nid_t nid = spec->autocfg.input_pins[i]; + if (!nid) + continue; + for (j = 0; j < nconns; j++) { + if (conn[j] == nid) + break; + } + if (j >= nconns) + break; + } + if (i >= AUTO_PIN_LAST) { + int num_adcs = spec->num_adc_nids; + spec->private_adc_nids[num_adcs] = adc; + spec->private_capsrc_nids[num_adcs] = cap; + spec->num_adc_nids++; + spec->adc_nids = spec->private_adc_nids; + if (adc != cap) + spec->capsrc_nids = spec->private_capsrc_nids; + } + } + if (!spec->num_adc_nids) { + printk(KERN_WARNING "hda_codec: %s: no valid ADC found;" + " using fallback 0x%x\n", + codec->chip_name, fallback_adc); + spec->private_adc_nids[0] = fallback_adc; + spec->adc_nids = spec->private_adc_nids; + if (fallback_adc != fallback_cap) { + spec->private_capsrc_nids[0] = fallback_cap; + spec->capsrc_nids = spec->private_adc_nids; + } + } +} + #ifdef CONFIG_SND_HDA_INPUT_BEEP #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) @@ -6326,6 +6427,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) } } +#define alc260_auto_init_input_src alc880_auto_init_input_src + /* * generic initialization of ADC, input mixers and output mixers */ @@ -6412,6 +6515,7 @@ static void alc260_auto_init(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); + alc260_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -8384,6 +8488,42 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc883_medion_wim2160_verbs[] = { + /* Unmute front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Set speaker pin to front mixer */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Init headphone pin */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc883_medion_wim2160_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1a; + spec->autocfg.speaker_pins[0] = 0x15; +} + static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -8398,9 +8538,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), @@ -9095,6 +9233,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", + [ALC883_MEDION_WIM2160] = "medion-wim2160", [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", @@ -9211,6 +9350,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R), @@ -9749,6 +9889,21 @@ static struct alc_config_preset alc882_presets[] = { .setup = alc883_medion_md2_setup, .init_hook = alc_automute_amp, }, + [ALC883_MEDION_WIM2160] = { + .mixers = { alc883_medion_wim2160_mixer }, + .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc883_medion_wim2160_setup, + .init_hook = alc_automute_amp, + }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, @@ -10041,13 +10196,12 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, int idx; alc_set_pin_output(codec, nid, pin_type); + if (dac_idx >= spec->multiout.num_dacs) + return; if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; - else { - if (spec->multiout.num_dacs >= dac_idx) - return; + else idx = spec->multiout.dac_nids[dac_idx] - 2; - } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } @@ -10295,7 +10449,8 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC882_AUTO; } - alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups); + if (board_config == ALC882_AUTO) + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1); if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ @@ -10368,6 +10523,9 @@ static int patch_alc882(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (board_config == ALC882_AUTO) + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0); + spec->vmaster_nid = 0x0c; codec->patch_ops = alc_patch_ops; @@ -12459,11 +12617,11 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, 0x15); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc268_acer_lc_unsol_event(struct hda_codec *codec, @@ -12748,6 +12906,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x21: /* ALC269vb has this pin, too */ dac = 0x03; break; default: @@ -13333,9 +13492,9 @@ static hda_nid_t alc269vb_capsrc_nids[1] = { 0x22, }; -/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24), - * not a mux! - */ +static hda_nid_t alc269_adc_candidates[] = { + 0x08, 0x09, 0x07, +}; #define alc269_modes alc260_modes #define alc269_capture_source alc880_lg_lw_capture_source @@ -13482,11 +13641,11 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, 0x15); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); @@ -13511,11 +13670,11 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec) /* Check port replicator headphone socket */ present |= snd_hda_jack_detect(codec, 0x1a); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); @@ -13646,11 +13805,11 @@ static void alc269_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, nid); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } /* unsolicited event for HP jack sensing */ @@ -13667,19 +13826,19 @@ static void alc269_laptop_unsol_event(struct hda_codec *codec, } } -static void alc269_laptop_dmic_setup(struct hda_codec *codec) +static void alc269_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; - spec->int_mic.pin = 0x12; - spec->int_mic.mux_idx = 5; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; spec->auto_mic = 1; } -static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +static void alc269_laptop_dmic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; @@ -13687,14 +13846,14 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; - spec->int_mic.mux_idx = 6; + spec->int_mic.mux_idx = 5; spec->auto_mic = 1; } -static void alc269_laptop_amic_setup(struct hda_codec *codec) +static void alc269vb_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -13703,6 +13862,18 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec) spec->auto_mic = 1; } +static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.speaker_pins[0] = 0x14; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 6; + spec->auto_mic = 1; +} + static void alc269_laptop_inithook(struct hda_codec *codec) { alc269_speaker_automute(codec); @@ -13842,7 +14013,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; - hda_nid_t real_capsrc_nids; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc269_ignore); @@ -13866,18 +14036,19 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) { add_verb(spec, alc269vb_init_verbs); - real_capsrc_nids = alc269vb_capsrc_nids[0]; alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21); } else { add_verb(spec, alc269_init_verbs); - real_capsrc_nids = alc269_capsrc_nids[0]; alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); } spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; + fillup_priv_adc_nids(codec, alc269_adc_candidates, + sizeof(alc269_adc_candidates)); + /* set default input source */ - snd_hda_codec_write_cache(codec, real_capsrc_nids, + snd_hda_codec_write_cache(codec, spec->capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, spec->input_mux->items[0].index); @@ -13907,6 +14078,27 @@ static void alc269_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC269_FIXUP_SONY_VAIO, +}; + +const static struct hda_verb alc269_sony_vaio_fixup_verbs[] = { + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, + {} +}; + +static const struct alc_fixup alc269_fixups[] = { + [ALC269_FIXUP_SONY_VAIO] = { + .verbs = alc269_sony_vaio_fixup_verbs + }, +}; + +static struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + {} +}; + + /* * configuration and preset */ @@ -13966,7 +14158,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), - SND_PCI_QUIRK(0x104d, 0x9071, "SONY XTB", ALC269_DMIC), + SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC), SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), @@ -14040,7 +14232,7 @@ static struct alc_config_preset alc269_presets[] = { .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, .unsol_event = alc269_laptop_unsol_event, - .setup = alc269_laptop_amic_setup, + .setup = alc269vb_laptop_amic_setup, .init_hook = alc269_laptop_inithook, }, [ALC269VB_DMIC] = { @@ -14120,6 +14312,9 @@ static int patch_alc269(struct hda_codec *codec) board_config = ALC269_AUTO; } + if (board_config == ALC269_AUTO) + alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1); + if (board_config == ALC269_AUTO) { /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); @@ -14156,20 +14351,25 @@ static int patch_alc269(struct hda_codec *codec) spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; - if (!is_alc269vb) { - spec->adc_nids = alc269_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); - spec->capsrc_nids = alc269_capsrc_nids; - } else { - spec->adc_nids = alc269vb_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); - spec->capsrc_nids = alc269vb_capsrc_nids; + if (!spec->adc_nids) { /* wasn't filled automatically? use default */ + if (!is_alc269vb) { + spec->adc_nids = alc269_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); + spec->capsrc_nids = alc269_capsrc_nids; + } else { + spec->adc_nids = alc269vb_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); + spec->capsrc_nids = alc269vb_capsrc_nids; + } } if (!spec->cap_mixer) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + if (board_config == ALC269_AUTO) + alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0); + spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; @@ -15258,7 +15458,8 @@ static int patch_alc861(struct hda_codec *codec) board_config = ALC861_AUTO; } - alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups); + if (board_config == ALC861_AUTO) + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1); if (board_config == ALC861_AUTO) { /* automatic parse from the BIOS config */ @@ -15295,6 +15496,9 @@ static int patch_alc861(struct hda_codec *codec) spec->vmaster_nid = 0x03; + if (board_config == ALC861_AUTO) + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0); + codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) { spec->init_hook = alc861_auto_init; @@ -16229,7 +16433,8 @@ static int patch_alc861vd(struct hda_codec *codec) board_config = ALC861VD_AUTO; } - alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups); + if (board_config == ALC861VD_AUTO) + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1); if (board_config == ALC861VD_AUTO) { /* automatic parse from the BIOS config */ @@ -16277,6 +16482,9 @@ static int patch_alc861vd(struct hda_codec *codec) spec->vmaster_nid = 0x02; + if (board_config == ALC861VD_AUTO) + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0); + codec->patch_ops = alc_patch_ops; if (board_config == ALC861VD_AUTO) @@ -17115,9 +17323,9 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) @@ -17128,13 +17336,13 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) @@ -17145,13 +17353,13 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x15); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc662_f5z_speaker_automute(struct hda_codec *codec) @@ -17190,14 +17398,14 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) if (present1 || present2) { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + HDA_AMP_MUTE, HDA_AMP_MUTE); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + HDA_AMP_MUTE, HDA_AMP_MUTE); } else { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), 0); + HDA_AMP_MUTE, 0); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), 0); + HDA_AMP_MUTE, 0); } } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c4be3fab94e5..7fb7d017a347 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1607,6 +1607,10 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "Dell Studio 1555", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd, "Dell Studio 1557", STAC_DELL_M6_DMIC), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe, + "Dell Studio XPS 1645", STAC_DELL_M6_BOTH), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413, + "Dell Studio 1558", STAC_DELL_M6_BOTH), {} /* terminator */ }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9ddc37300f6b..73453814e098 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -476,7 +476,7 @@ static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec, knew->name = kstrdup(tmpl->name, GFP_KERNEL); if (!knew->name) return NULL; - return 0; + return knew; } static void via_free_kctls(struct hda_codec *codec) @@ -1215,14 +1215,13 @@ static struct snd_kcontrol_new via_hp_mixer[2] = { }, }; -static int via_hp_build(struct via_spec *spec) +static int via_hp_build(struct hda_codec *codec) { + struct via_spec *spec = codec->spec; struct snd_kcontrol_new *knew; hda_nid_t nid; - - knew = via_clone_control(spec, &via_hp_mixer[0]); - if (knew == NULL) - return -ENOMEM; + int nums; + hda_nid_t conn[HDA_MAX_CONNECTIONS]; switch (spec->codec_type) { case VT1718S: @@ -1239,6 +1238,14 @@ static int via_hp_build(struct via_spec *spec) break; } + nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); + if (nums <= 1) + return 0; + + knew = via_clone_control(spec, &via_hp_mixer[0]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; knew->private_value = nid; @@ -2561,7 +2568,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -3087,7 +3094,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -3654,7 +3661,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -4140,7 +4147,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -4510,7 +4517,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); return 1; } @@ -4930,7 +4937,7 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); @@ -5425,7 +5432,7 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); @@ -5781,7 +5788,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); return 1; } @@ -6000,12 +6007,12 @@ static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec, /* Line-Out: PortE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, - "Master Front Playback Volume", + "Front Playback Volume", HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, - "Master Front Playback Switch", + "Front Playback Switch", HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -6130,7 +6137,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); return 1; } diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index b64e78139d63..b56e33676780 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -849,6 +849,7 @@ struct snd_m3 { struct snd_kcontrol *master_switch; struct snd_kcontrol *master_volume; struct tasklet_struct hwvol_tq; + unsigned int in_suspend; #ifdef CONFIG_PM u16 *suspend_mem; @@ -884,6 +885,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = { MODULE_DEVICE_TABLE(pci, snd_m3_ids); static struct snd_pci_quirk m3_amp_quirk_list[] __devinitdata = { + SND_PCI_QUIRK(0x0E11, 0x0094, "Compaq Evo N600c", 0x0c), SND_PCI_QUIRK(0x10f7, 0x833e, "Panasonic CF-28", 0x0d), SND_PCI_QUIRK(0x10f7, 0x833d, "Panasonic CF-72", 0x0d), SND_PCI_QUIRK(0x1033, 0x80f1, "NEC LM800J/7", 0x03), @@ -1613,6 +1615,11 @@ static void snd_m3_update_hw_volume(unsigned long private_data) outb(0x88, chip->iobase + SHADOW_MIX_REG_MASTER); outb(0x88, chip->iobase + HW_VOL_COUNTER_MASTER); + /* Ignore spurious HV interrupts during suspend / resume, this avoids + mistaking them for a mute button press. */ + if (chip->in_suspend) + return; + if (!chip->master_switch || !chip->master_volume) return; @@ -2424,6 +2431,7 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state) if (chip->suspend_mem == NULL) return 0; + chip->in_suspend = 1; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); snd_pcm_suspend_all(chip->pcm); snd_ac97_suspend(chip->ac97); @@ -2497,6 +2505,7 @@ static int m3_resume(struct pci_dev *pci) snd_m3_hv_init(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D0); + chip->in_suspend = 0; return 0; } #endif /* CONFIG_PM */ diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 55e9315d4ccd..3be8f97c8bc0 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1162,13 +1162,15 @@ static long snd_mixart_BA0_read(struct snd_info_entry *entry, void *file_private unsigned long count, unsigned long pos) { struct mixart_mgr *mgr = entry->private_data; + unsigned long maxsize; - count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ - if(count <= 0) + if (pos >= MIXART_BA0_SIZE) return 0; - if(pos + count > MIXART_BA0_SIZE) - count = (long)(MIXART_BA0_SIZE - pos); - if(copy_to_user_fromio(buf, MIXART_MEM( mgr, pos ), count)) + maxsize = MIXART_BA0_SIZE - pos; + if (count > maxsize) + count = maxsize; + count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ + if (copy_to_user_fromio(buf, MIXART_MEM(mgr, pos), count)) return -EFAULT; return count; } @@ -1181,13 +1183,15 @@ static long snd_mixart_BA1_read(struct snd_info_entry *entry, void *file_private unsigned long count, unsigned long pos) { struct mixart_mgr *mgr = entry->private_data; + unsigned long maxsize; - count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ - if(count <= 0) + if (pos > MIXART_BA1_SIZE) return 0; - if(pos + count > MIXART_BA1_SIZE) - count = (long)(MIXART_BA1_SIZE - pos); - if(copy_to_user_fromio(buf, MIXART_REG( mgr, pos ), count)) + maxsize = MIXART_BA1_SIZE - pos; + if (count > maxsize) + count = maxsize; + count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ + if (copy_to_user_fromio(buf, MIXART_REG(mgr, pos), count)) return -EFAULT; return count; } diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 9ef6b96373f5..3e6628c8e665 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -180,7 +180,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); - prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); prtd->params->dma_intr_handler = atmel_pcm_dma_irq; prtd->dma_buffer = runtime->dma_addr; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e588e63f18d2..0b59806905d1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -363,12 +363,12 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, ssc_p->dma_params[dir] = dma_params; /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the pcm driver hw_params() + * The snd_soc_pcm_stream->dma_data field is only used to communicate + * the appropriate DMA parameters to the pcm driver hw_params() * function. It should not be used for other purposes * as it is common to all substreams. */ - rtd->dai->cpu_dai->dma_data = dma_params; + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_params); channels = params_channels(params); diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index fd101d450d56..1f5e57a4bb7a 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -81,9 +81,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; struct snd_soc_codec *codec; struct snd_ac97_bus *ac97_bus; struct snd_ac97_template ac97_template; + int i; int ret = 0; printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); @@ -103,12 +105,6 @@ static int ac97_soc_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); - if (ret < 0) { - printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n"); - goto err; - } - /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) @@ -124,6 +120,13 @@ static int ac97_soc_probe(struct platform_device *pdev) if (ret < 0) goto bus_err; + for (i = 0; i < card->num_links; i++) { + if (card->dai_link[i].codec_dai->ac97_control) { + snd_ac97_dev_add_pdata(codec->ac97, + card->dai_link[i].cpu_dai->ac97_pdata); + } + } + return 0; bus_err: diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a34cbcf7904f..002e289d1255 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -23,7 +23,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/firmware.h> diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8d1c63754be4..9da0724cd47a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3008,34 +3008,39 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - /* Switch over to startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (1 << WM8994_VMID_RAMP_SHIFT)); - - /* Disable main biases */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0); + if (codec->bias_level == SND_SOC_BIAS_STANDBY) { + /* Switch over to startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (1 << WM8994_VMID_RAMP_SHIFT)); - /* Discharge line */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_1, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH); + /* Disable main biases */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, 0); - msleep(5); + /* Discharge line */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); - /* Switch off startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, 0); + msleep(5); + /* Switch off startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, 0); + } break; } codec->bias_level = level; @@ -3402,7 +3407,7 @@ struct snd_soc_dai wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, - .playback = { + .capture = { .stream_name = "AIF3 Capture", .channels_min = 2, .channels_max = 2, @@ -3731,11 +3736,12 @@ static int wm8994_codec_probe(struct platform_device *pdev) case 3: wm8994->hubs.dcs_codes = -5; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.dcs_readback_mode = 1; break; default: + wm8994->hubs.dcs_readback_mode = 1; break; } - /* Remember if AIFnLRCLK is configured as a GPIO. This should be * configured on init - if a system wants to do this dynamically diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 486bdd21a98a..e1f225a3ac46 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -62,21 +62,27 @@ static const char *speaker_mode_text[] = { static const struct soc_enum speaker_mode = SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text); -static void wait_for_dc_servo(struct snd_soc_codec *codec) +static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { unsigned int reg; int count = 0; + unsigned int val; + + val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1; + + /* Trigger the command */ + snd_soc_write(codec, WM8993_DC_SERVO_0, val); dev_dbg(codec->dev, "Waiting for DC servo...\n"); do { count++; msleep(1); - reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0); + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400); + } while (reg & op && count < 400); - if (reg & WM8993_DCS_DATAPATH_BUSY) + if (reg & op) dev_err(codec->dev, "Timed out waiting for DC Servo\n"); } @@ -86,51 +92,58 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = codec->private_data; - u16 reg, dcs_cfg; + u16 reg, reg_l, reg_r, dcs_cfg; /* Set for 32 series updates */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, WM8993_DCS_SERIES_NO_01_MASK, 32 << WM8993_DCS_SERIES_NO_01_SHIFT); - - /* Enable the DC servo. Write all bits to avoid triggering startup - * or write calibration. - */ - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - 0xFFFF, - WM8993_DCS_ENA_CHAN_0 | - WM8993_DCS_ENA_CHAN_1 | - WM8993_DCS_TRIG_SERIES_1 | - WM8993_DCS_TRIG_SERIES_0); - - wait_for_dc_servo(codec); + wait_for_dc_servo(codec, + WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1); /* Apply correction to DC servo result */ if (hubs->dcs_codes) { dev_dbg(codec->dev, "Applying %d code DC servo correction\n", hubs->dcs_codes); + /* Different chips in the family support different + * readback methods. + */ + switch (hubs->dcs_readback_mode) { + case 0: + reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) + & WM8993_DCS_INTEG_CHAN_0_MASK;; + reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) + & WM8993_DCS_INTEG_CHAN_1_MASK; + break; + case 1: + reg = snd_soc_read(codec, WM8993_DC_SERVO_3); + reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) + >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; + reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; + break; + default: + WARN(1, "Unknown DCS readback method"); + break; + } + /* HPOUT1L */ - reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) & - WM8993_DCS_INTEG_CHAN_0_MASK;; - reg += hubs->dcs_codes; - dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + if (reg_l + hubs->dcs_codes > 0 && + reg_l + hubs->dcs_codes < 0xff) + reg_l += hubs->dcs_codes; + dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ - reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & - WM8993_DCS_INTEG_CHAN_1_MASK; - reg += hubs->dcs_codes; - dcs_cfg |= reg; + if (reg_r + hubs->dcs_codes > 0 && + reg_r + hubs->dcs_codes < 0xff) + reg_r += hubs->dcs_codes; + dcs_cfg |= reg_r; /* Do it */ snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - WM8993_DCS_TRIG_DAC_WR_0 | - WM8993_DCS_TRIG_DAC_WR_1, - WM8993_DCS_TRIG_DAC_WR_0 | - WM8993_DCS_TRIG_DAC_WR_1); - - wait_for_dc_servo(codec); + wait_for_dc_servo(codec, + WM8993_DCS_TRIG_DAC_WR_0 | + WM8993_DCS_TRIG_DAC_WR_1); } } @@ -141,10 +154,16 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm_hubs_data *hubs = codec->private_data; int ret; ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + /* If we're applying an offset correction then updating the + * callibration would be likely to introduce further offsets. */ + if (hubs->dcs_codes) + return ret; + /* Only need to do this if the outputs are active */ if (snd_soc_read(codec, WM8993_POWER_MANAGEMENT_1) & (WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA)) diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 420104fe9c90..e51c16683589 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -21,6 +21,7 @@ extern const unsigned int wm_hubs_spkmix_tlv[]; /* This *must* be the first element of the codec->private_data struct */ struct wm_hubs_data { int dcs_codes; + int dcs_readback_mode; int hp_startup_mode; }; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 62af7e025e7f..adadcd3aa1b1 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -586,7 +586,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; - davinci_i2s_dai.dma_data = dev->dma_params; + davinci_i2s_dai.capture.dma_data = dev->dma_params; + davinci_i2s_dai.playback.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6c80cc35ecad..79f0f4ad242c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -918,7 +918,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; - davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; + davinci_mcasp_dai[pdata->op_mode].capture.dma_data = dev->dma_params; + davinci_mcasp_dai[pdata->op_mode].playback.dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 80c7fdf2f521..2dc406f42fe7 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -649,8 +649,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_hardware *ppcm; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *pa; struct davinci_pcm_dma_params *params; + + pa = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); if (!pa) return -ENODEV; params = &pa[substream->stream]; diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 86668ab3f4d4..2b31ac673ea4 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -71,7 +71,12 @@ static void imx_ssi_dma_callback(int channel, void *data) static void snd_imx_dma_err_callback(int channel, void *data, int err) { - pr_err("DMA error callback called\n"); + struct snd_pcm_substream *substream = data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int ret; pr_err("DMA timeout on channel %d -%s%s%s%s\n", channel, @@ -79,16 +84,26 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err) err & IMX_DMA_ERR_REQUEST ? " request" : "", err & IMX_DMA_ERR_TRANSFER ? " transfer" : "", err & IMX_DMA_ERR_BUFFER ? " buffer" : ""); + + imx_dma_disable(iprtd->dma); + ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count, + IMX_DMA_LENGTH_LOOP, dma_params->dma_addr, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_MODE_WRITE : DMA_MODE_READ); + if (!ret) + imx_dma_enable(iprtd->dma); } static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_dma_params *dma_params; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; int ret; + dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); if (iprtd->dma < 0) { pr_err("Failed to claim the audio DMA\n"); @@ -193,10 +208,12 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_dma_params *dma_params; struct imx_pcm_runtime_data *iprtd = runtime->private_data; int err; + dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + iprtd->substream = substream; iprtd->buf = (unsigned int *)substream->dma_buffer.area; iprtd->period_cnt = 0; diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index f96a373699cf..6b518e07eea9 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -39,23 +39,24 @@ struct imx_pcm_runtime_data { unsigned long offset; unsigned long last_offset; unsigned long size; - struct timer_list timer; - int poll_time; + struct hrtimer hrt; + int poll_time_ns; + struct snd_pcm_substream *substream; + atomic_t running; }; -static inline void imx_ssi_set_next_poll(struct imx_pcm_runtime_data *iprtd) +static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { - iprtd->timer.expires = jiffies + iprtd->poll_time; -} - -static void imx_ssi_timer_callback(unsigned long data) -{ - struct snd_pcm_substream *substream = (void *)data; + struct imx_pcm_runtime_data *iprtd = + container_of(hrt, struct imx_pcm_runtime_data, hrt); + struct snd_pcm_substream *substream = iprtd->substream; struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; struct pt_regs regs; unsigned long delta; + if (!atomic_read(&iprtd->running)) + return HRTIMER_NORESTART; + get_fiq_regs(®s); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -72,16 +73,14 @@ static void imx_ssi_timer_callback(unsigned long data) /* If we've transferred at least a period then report it and * reset our poll time */ - if (delta >= runtime->period_size) { + if (delta >= iprtd->period) { snd_pcm_period_elapsed(substream); iprtd->last_offset = iprtd->offset; - - imx_ssi_set_next_poll(iprtd); } - /* Restart the timer; if we didn't report we'll run on the next tick */ - add_timer(&iprtd->timer); + hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); + return HRTIMER_RESTART; } static struct fiq_handler fh = { @@ -99,8 +98,8 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, iprtd->period = params_period_bytes(params) ; iprtd->offset = 0; iprtd->last_offset = 0; - iprtd->poll_time = HZ / (params_rate(params) / params_period_size(params)); - + iprtd->poll_time_ns = 1000000000 / params_rate(params) * + params_period_size(params); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; @@ -135,8 +134,9 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - imx_ssi_set_next_poll(iprtd); - add_timer(&iprtd->timer); + atomic_set(&iprtd->running, 1); + hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns), + HRTIMER_MODE_REL); if (++fiq_enable == 1) enable_fiq(imx_pcm_fiq); @@ -145,11 +145,11 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - del_timer(&iprtd->timer); + atomic_set(&iprtd->running, 0); + if (--fiq_enable == 0) disable_fiq(imx_pcm_fiq); - break; default: return -EINVAL; @@ -180,7 +180,7 @@ static struct snd_pcm_hardware snd_imx_hardware = { .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 16 * 1024, - .periods_min = 2, + .periods_min = 4, .periods_max = 255, .fifo_size = 0, }; @@ -194,9 +194,11 @@ static int snd_imx_open(struct snd_pcm_substream *substream) iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); runtime->private_data = iprtd; - init_timer(&iprtd->timer); - iprtd->timer.data = (unsigned long)substream; - iprtd->timer.function = imx_ssi_timer_callback; + iprtd->substream = substream; + + atomic_set(&iprtd->running, 0); + hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + iprtd->hrt.function = snd_hrtimer_callback; ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -212,7 +214,8 @@ static int snd_imx_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - del_timer_sync(&iprtd->timer); + hrtimer_cancel(&iprtd->hrt); + kfree(iprtd); return 0; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 6546b06cbd2a..80b4fee2442b 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -235,17 +235,20 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct imx_ssi *ssi = cpu_dai->private_data; + struct imx_pcm_dma_params *dma_data; u32 reg, sccr; /* Tx/Rx config */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { reg = SSI_STCCR; - cpu_dai->dma_data = &ssi->dma_params_tx; + dma_data = &ssi->dma_params_tx; } else { reg = SSI_SRCCR; - cpu_dai->dma_data = &ssi->dma_params_rx; + dma_data = &ssi->dma_params_rx; } + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; /* DAI data (word) size */ @@ -653,7 +656,8 @@ static int imx_ssi_probe(struct platform_device *pdev) dai->private_data = ssi; if ((cpu_is_mx27() || cpu_is_mx21()) && - !(ssi->flags & IMX_SSI_USE_AC97)) { + !(ssi->flags & IMX_SSI_USE_AC97) && + (ssi->flags & IMX_SSI_DMA)) { ssi->flags |= IMX_SSI_DMA; platform = imx_ssi_dma_mx2_init(pdev, ssi); } else diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index e814a9591f78..8ad9dc901007 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -297,7 +297,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; omap_mcbsp_dai_dma_params[id][substream->stream].data_type = OMAP_DMA_DATA_TYPE_S16; - cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; + + snd_soc_dai_set_dma_data(cpu_dai, substream, + &omap_mcbsp_dai_dma_params[id][substream->stream]); if (mcbsp_data->configured) { /* McBSP already configured by another stream */ diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 25f19e4728bf..b7f4f7e015f3 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -150,7 +150,8 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, int stream = substream->stream; int channels, err, link_mask = 0; - cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + snd_soc_dai_set_dma_data(cpu_dai, substream, + &omap_mcpdm_dai_dma_params[stream]); channels = params_channels(params); switch (channels) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index ba8acbb0a7fa..1e521904ea64 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -61,12 +61,11 @@ static void omap_pcm_dma_irq(int ch, u16 stat, void *data) struct omap_runtime_data *prtd = runtime->private_data; unsigned long flags; - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) { + if ((cpu_is_omap1510())) { /* * OMAP1510 doesn't fully support DMA progress counter * and there is no software emulation implemented yet, - * so have to maintain our own playback progress counter + * so have to maintain our own progress counters * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); @@ -101,9 +100,11 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct omap_runtime_data *prtd = runtime->private_data; - struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data; + struct omap_pcm_dma_data *dma_data; int err = 0; + dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma_data) @@ -190,8 +191,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + if ((cpu_is_omap1510())) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); else @@ -249,14 +249,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (cpu_is_omap1510()) { + offset = prtd->period_index * runtime->period_size; + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else if (!(cpu_is_omap1510())) { + } else { ptr = omap_get_dma_src_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else - offset = prtd->period_index * runtime->period_size; + } if (offset >= runtime->buffer_size) offset = 0; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d5fc52d0a3c4..544fd9566f4d 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -122,10 +122,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, ssp_disable(ssp); } - if (cpu_dai->dma_data) { - kfree(cpu_dai->dma_data); - cpu_dai->dma_data = NULL; - } + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); + snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); + return ret; } @@ -142,10 +141,8 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable(ssp->clk); } - if (cpu_dai->dma_data) { - kfree(cpu_dai->dma_data); - cpu_dai->dma_data = NULL; - } + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); + snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); } #ifdef CONFIG_PM @@ -570,19 +567,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + struct pxa2xx_pcm_dma_params *dma_data; + + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* generate correct DMA params */ - if (cpu_dai->dma_data) - kfree(cpu_dai->dma_data); + kfree(dma_data); /* Network mode with one active slot (ttsa == 1) can be used * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, + dma_data = ssp_get_dma_params(ssp, ((chn == 2) && (ttsa != 1)) || (width == 32), substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_dma_data(dai, substream, dma_data); + /* we can only change the settings if the port is not in use */ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index e9ae7b3a7e00..d314115e3dd7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -122,11 +122,14 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; + dma_data = &pxa2xx_ac97_pcm_stereo_out; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in; + dma_data = &pxa2xx_ac97_pcm_stereo_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -137,11 +140,14 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; + dma_data = &pxa2xx_ac97_pcm_aux_mono_out; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in; + dma_data = &pxa2xx_ac97_pcm_aux_mono_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -156,7 +162,8 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in; + snd_soc_dai_set_dma_data(cpu_dai, substream, + &pxa2xx_ac97_pcm_mic_mono_in); return 0; } diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6b8f655d1ad8..c1a5275721e4 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -164,6 +164,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); @@ -171,9 +172,11 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out; + dma_data = &pxa2xx_i2s_pcm_stereo_out; else - cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in; + dma_data = &pxa2xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); /* is port used by another stream */ if (!(SACR0 & SACR0_ENB)) { diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index d38e39575f51..adc7e6f15f93 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -25,9 +25,11 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; + struct pxa2xx_pcm_dma_params *dma; int ret; + dma = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma) diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c index ee8ed9d7e703..ecf4fd04ae96 100644 --- a/sound/soc/s3c24xx/s3c-ac97.c +++ b/sound/soc/s3c24xx/s3c-ac97.c @@ -224,11 +224,14 @@ static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &s3c_ac97_pcm_out; + dma_data = &s3c_ac97_pcm_out; else - cpu_dai->dma_data = &s3c_ac97_pcm_in; + dma_data = &s3c_ac97_pcm_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -238,8 +241,8 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -265,7 +268,7 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); return 0; } @@ -280,7 +283,7 @@ static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; else - cpu_dai->dma_data = &s3c_ac97_mic_in; + snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); return 0; } @@ -290,8 +293,8 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; @@ -311,7 +314,7 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); return 0; } diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c index 7725e26d6c91..1b61c23ff300 100644 --- a/sound/soc/s3c24xx/s3c-dma.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -145,10 +145,12 @@ static int s3c_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); int ret = 0; + pr_debug("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index e994d8374fe6..88515946b6c0 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -339,14 +339,17 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai = rtd->dai; struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->cpu_dai->dma_data = i2s->dma_playback; + dma_data = i2s->dma_playback; else - dai->cpu_dai->dma_data = i2s->dma_capture; + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); /* Working copies of register */ iismod = readl(i2s->regs + S3C2412_IISMOD); @@ -394,8 +397,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -431,7 +434,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, * of the auto reload mechanism of S3C24XX. * This call won't bother S3C64XX. */ - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c index a98f40c3cd29..326f0a9e7e30 100644 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -178,6 +178,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai = rtd->dai; struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -187,9 +188,11 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->cpu_dai->dma_data = pcm->dma_playback; + dma_data = pcm->dma_playback; else - dai->cpu_dai->dma_data = pcm->dma_capture; + dma_data = pcm->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); /* Strictly check for sample size */ switch (params_format(params)) { diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0bc5950b9f02..c3ac890a3986 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -242,14 +242,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; + dma_data = &s3c24xx_i2s_pcm_stereo_out; else - rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in; + dma_data = &s3c24xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -258,13 +261,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->dma_size = 1; + dma_data->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->dma_size = 2; + dma_data->dma_size = 2; break; default: return -EINVAL; @@ -280,8 +281,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -300,7 +301,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 0664fac7612a..5b9ac1759bd2 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -519,7 +519,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) s6000_i2s_dai.dev = &pdev->dev; s6000_i2s_dai.private_data = dev; - s6000_i2s_dai.dma_data = &dev->dma_params; + s6000_i2s_dai.capture.dma_data = &dev->dma_params; + s6000_i2s_dai.playback.dma_data = &dev->dma_params; dev->sifbase = sifmem->start; dev->scbbase = mmio; diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 1d61109e09fa..9c7f7f00cebb 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -58,13 +58,15 @@ static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; int channel; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; dma_addr_t src, dst; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; @@ -101,7 +103,8 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data) { struct snd_pcm *pcm = data; struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *params = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); struct s6000_runtime_data *prtd; unsigned int has_xrun; int i, ret = IRQ_NONE; @@ -172,11 +175,13 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream) { struct s6000_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; unsigned long flags; int srcinc; u32 dma; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + spin_lock_irqsave(&prtd->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -212,10 +217,12 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream) { struct s6000_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; unsigned long flags; u32 channel; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) channel = par->dma_out; else @@ -236,9 +243,11 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream) static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; int ret; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + ret = par->trigger(substream, cmd, 0); if (ret < 0) return ret; @@ -275,13 +284,15 @@ static int s6000_pcm_prepare(struct snd_pcm_substream *substream) static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd = runtime->private_data; unsigned long flags; unsigned int offset; dma_addr_t count; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + spin_lock_irqsave(&prtd->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -305,11 +316,12 @@ static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) static int s6000_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd; int ret; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware); ret = snd_pcm_hw_constraint_step(runtime, 0, @@ -364,7 +376,7 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; int ret; ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); @@ -373,6 +385,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, return ret; } + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (par->same_rate) { spin_lock(&par->lock); if (par->rate == -1 || @@ -392,7 +406,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, static int s6000_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); spin_lock(&par->lock); par->in_use &= ~(1 << substream->stream); @@ -417,7 +432,8 @@ static struct snd_pcm_ops s6000_pcm_ops = { static void s6000_pcm_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *params = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); free_irq(params->irq, pcm); snd_pcm_lib_preallocate_free_for_all(pcm); @@ -429,9 +445,11 @@ static int s6000_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *params; int res; + params = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (!card->dev->dma_mask) card->dev->dma_mask = &s6000_pcm_dmamask; if (!card->dev->coherent_dma_mask) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2320153bd923..ad7f9528d751 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1549,7 +1549,8 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) mutex_unlock(&codec->mutex); return ret; } - if (card->dai_link[i].codec_dai->ac97_control) { + /* Check for codec->ac97 to handle the ac97.c fun */ + if (card->dai_link[i].codec_dai->ac97_control && codec->ac97) { snd_ac97_dev_add_pdata(codec->ac97, card->dai_link[i].cpu_dai->ac97_pdata); } diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 612e18b4bf4e..0ec20b68e8cb 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -254,3 +254,4 @@ module_exit(txx9aclc_ac97_exit); MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>"); MODULE_DESCRIPTION("TXx9 ACLC AC97 driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:txx9aclc-ac97"); diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c index 3175de9a92cb..95b17f731aec 100644 --- a/sound/soc/txx9/txx9aclc-generic.c +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -96,3 +96,4 @@ module_exit(txx9aclc_generic_exit); MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>"); MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:txx9aclc-generic"); diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 2c59afd99611..9e28b20cb2ce 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) DEFINE_WAIT(wait); long timeout = msecs_to_jiffies(50); + if (ep->umidi->disconnected) + return; /* * The substream buffer is empty, but some data might still be in the * currently active URBs, so we have to wait for those to complete. @@ -1123,14 +1125,21 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, * Frees an output endpoint. * May be called when ep hasn't been initialized completely. */ -static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_out_endpoint_clear(struct snd_usb_midi_out_endpoint *ep) { unsigned int i; for (i = 0; i < OUTPUT_URBS; ++i) - if (ep->urbs[i].urb) + if (ep->urbs[i].urb) { free_urb_and_buffer(ep->umidi, ep->urbs[i].urb, ep->max_transfer); + ep->urbs[i].urb = NULL; + } +} + +static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep) +{ + snd_usbmidi_out_endpoint_clear(ep); kfree(ep); } @@ -1262,15 +1271,18 @@ void snd_usbmidi_disconnect(struct list_head* p) usb_kill_urb(ep->out->urbs[j].urb); if (umidi->usb_protocol_ops->finish_out_endpoint) umidi->usb_protocol_ops->finish_out_endpoint(ep->out); + ep->out->active_urbs = 0; + if (ep->out->drain_urbs) { + ep->out->drain_urbs = 0; + wake_up(&ep->out->drain_wait); + } } if (ep->in) for (j = 0; j < INPUT_URBS; ++j) usb_kill_urb(ep->in->urbs[j]); /* free endpoints here; later call can result in Oops */ - if (ep->out) { - snd_usbmidi_out_endpoint_delete(ep->out); - ep->out = NULL; - } + if (ep->out) + snd_usbmidi_out_endpoint_clear(ep->out); if (ep->in) { snd_usbmidi_in_endpoint_delete(ep->in); ep->in = NULL; |