diff options
Diffstat (limited to 'sound')
46 files changed, 543 insertions, 297 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5b3274b465e..199cca3366d 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -26,10 +26,10 @@ #include <asm/irq.h> #include <linux/mutex.h> -#include <asm/hardware.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/pxa2xx-gpio.h> -#include <asm/arch/audio.h> +#include <mach/hardware.h> +#include <mach/pxa-regs.h> +#include <mach/pxa2xx-gpio.h> +#include <mach/audio.h> #include "pxa2xx-pcm.h" diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 0ede9e4656a..381094aab23 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -21,8 +21,8 @@ #include <sound/pcm_params.h> #include <asm/dma.h> -#include <asm/hardware.h> -#include <asm/arch/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/pxa-regs.h> #include "pxa2xx-pcm.h" diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index faeddf3eced..b9c51bf8cd7 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -71,8 +71,8 @@ #include <linux/pm.h> #endif -#include <asm/hardware.h> -#include <asm/arch/h3600.h> +#include <mach/hardware.h> +#include <mach/h3600.h> #include <asm/mach-types.h> #include <asm/dma.h> diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 558dadbf45f..e024e4588b8 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -604,6 +604,9 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in { struct seq_oss_synth *rec; + if (dev < 0 || dev >= dp->max_synthdev) + return -ENXIO; + if (dp->synths[dev].is_midi) { struct midi_info minf; snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf); diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 87e3aefeddc..83e90057270 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -189,9 +189,7 @@ void snd_tea575x_init(struct snd_tea575x *tea) } memset(&tea->vd, 0, sizeof(tea->vd)); - tea->vd.owner = tea->card->module; strcpy(tea->vd.name, tea->tea5759 ? "TEA5759 radio" : "TEA5757 radio"); - tea->vd.type = VID_TYPE_TUNER; tea->vd.release = snd_tea575x_release; video_set_drvdata(&tea->vd, tea); tea->vd.fops = &tea->fops; diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index dbe63db4bfd..4d4b8ddc26b 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -325,6 +325,7 @@ static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, struct pnp_dev *pdev) { + acard->wss = pdev; if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; cport[dev] = -1; diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 41c047e665e..0797ca441a3 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -68,7 +68,9 @@ MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (AD1848)}," static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ //static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +#ifdef CONFIG_PNP static int isapnp = 1; /* Enable ISA PnP detection */ +#endif static long port = SNDRV_DEFAULT_PORT1; /* 0x530,0xe80,0xf40,0x604 */ static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x300,0x310,0x320,0x330 */ static long fm_port = SNDRV_DEFAULT_PORT1; /* 0x388 */ @@ -85,8 +87,10 @@ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for opti9xx based soundcard."); //module_param(enable, bool, 0444); //MODULE_PARM_DESC(enable, "Enable opti9xx soundcard."); +#ifdef CONFIG_PNP module_param(isapnp, bool, 0444); MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); +#endif module_param(port, long, 0444); MODULE_PARM_DESC(port, "WSS port # for opti9xx driver."); module_param(mpu_port, long, 0444); @@ -688,7 +692,7 @@ static void snd_card_opti9xx_free(struct snd_card *card) if (chip) { #ifdef OPTi93X struct snd_cs4231 *codec = chip->codec; - if (codec->irq > 0) { + if (codec && codec->irq > 0) { disable_irq(codec->irq); free_irq(codec->irq, codec); } diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index bb4a0969f46..725fef0f59a 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -22,7 +22,7 @@ #include <linux/kernel.h> #include <linux/interrupt.h> -#include <asm/hardware.h> +#include <mach/hardware.h> #include <asm/dma.h> #include <asm/io.h> #include <asm/hardware/iomd.h> diff --git a/sound/oss/vidc_fill.S b/sound/oss/vidc_fill.S index 01ccc074cc1..bed34921d17 100644 --- a/sound/oss/vidc_fill.S +++ b/sound/oss/vidc_fill.S @@ -11,7 +11,7 @@ */ #include <linux/linkage.h> #include <asm/assembler.h> -#include <asm/hardware.h> +#include <mach/hardware.h> #include <asm/hardware/iomd.h> .text diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index 88490418f93..c47842fad65 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -47,7 +47,7 @@ #include "waveartist.h" #ifdef CONFIG_ARM -#include <asm/hardware.h> +#include <mach/hardware.h> #include <asm/mach-types.h> #endif diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 07364c00768..8c49a00a5e3 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -161,6 +161,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, +{ 0x54524103, 0xffffffff, "TR28023", NULL, NULL }, { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] @@ -169,7 +170,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF { 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, -{ 0x574d4c00, 0xffffffff, "WM9701A", NULL, NULL }, +{ 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, { 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, { 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 0746e9ccc20..f4fbc795ee8 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3381,8 +3381,8 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, } /* create a virtual master control and add slaves */ -int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, - const unsigned int *tlv, const char **slaves) +static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, + const unsigned int *tlv, const char **slaves) { struct snd_kcontrol *kctl; const char **s; diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 7e3e8942d07..974e05122f0 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -94,7 +94,7 @@ enum azf_freq_t { AZF_FREQ(48000), AZF_FREQ(66200), #undef AZF_FREQ -} AZF_FREQUENCIES; +}; /** recording area (see also: playback bit flag definitions) **/ #define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */ @@ -210,7 +210,7 @@ enum azf_freq_t { enum { AZF_GAME_LEGACY_IO_PORT = 0x200 -} AZF_GAME_CONFIGS; +}; #define IDX_GAME_LEGACY_COMPATIBLE 0x00 /* in some operation mode, writing anything to this port diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index fbf1124f7c7..9bf95367c88 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -522,7 +522,7 @@ static unsigned int snd_es1371_wait_src_ready(struct ensoniq * ensoniq) return r; cond_resched(); } - snd_printk(KERN_ERR "wait source ready timeout 0x%lx [0x%x]\n", + snd_printk(KERN_ERR "wait src ready timeout 0x%lx [0x%x]\n", ES_REG(ensoniq, 1371_SMPRATE), r); return 0; } @@ -1629,6 +1629,7 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq, memset(&ac97, 0, sizeof(ac97)); ac97.private_data = ensoniq; ac97.private_free = snd_ensoniq_mixer_free_ac97; + ac97.pci = ensoniq->pci; ac97.scaps = AC97_SCAP_AUDIO; if ((err = snd_ac97_mixer(pbus, &ac97, &ensoniq->u.es1371.ac97)) < 0) return err; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 16715a68ba5..ef9f072b47f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1047,9 +1047,13 @@ static int azx_setup_periods(struct azx *chip, pos_adj = bdl_pos_adj[chip->dev_index]; if (pos_adj > 0) { struct snd_pcm_runtime *runtime = substream->runtime; + int pos_align = pos_adj; pos_adj = (pos_adj * runtime->rate + 47999) / 48000; if (!pos_adj) - pos_adj = 1; + pos_adj = pos_align; + else + pos_adj = ((pos_adj + pos_align - 1) / pos_align) * + pos_align; pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { snd_printk(KERN_WARNING "Too big adjustment %d\n", diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2807bc840d2..add4e87e0b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -122,6 +122,8 @@ enum { /* ALC269 models */ enum { ALC269_BASIC, + ALC269_ASUS_EEEPC_P703, + ALC269_ASUS_EEEPC_P901, ALC269_AUTO, ALC269_MODEL_LAST /* last tag */ }; @@ -7905,6 +7907,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), 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), @@ -10946,7 +10949,23 @@ static int patch_alc268(struct hda_codec *codec) static hda_nid_t alc269_adc_nids[1] = { /* ADC1 */ - 0x07, + 0x08, +}; + +static struct hda_input_mux alc269_eeepc_dmic_capture_source = { + .num_items = 2, + .items = { + { "i-Mic", 0x5 }, + { "e-Mic", 0x0 }, + }, +}; + +static struct hda_input_mux alc269_eeepc_amic_capture_source = { + .num_items = 2, + .items = { + { "i-Mic", 0x1 }, + { "e-Mic", 0x0 }, + }, }; #define alc269_modes alc260_modes @@ -10968,10 +10987,27 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { { } /* end */ }; +/* bind volumes of both NID 0x0c and 0x0d */ +static struct hda_bind_ctls alc269_epc_bind_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc269_eeepc_mixer[] = { + HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol), + HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT), + { } /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc269_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -10987,6 +11023,13 @@ static struct snd_kcontrol_new alc269_capture_mixer[] = { { } /* end */ }; +/* capture mixer elements */ +static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + { } /* end */ +}; + /* * generic initialization of ADC, input mixers and output mixers */ @@ -10994,7 +11037,7 @@ static struct hda_verb alc269_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the * analog-loopback mixer widget @@ -11057,6 +11100,98 @@ static struct hda_verb alc269_init_verbs[] = { { } }; +static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x23, AC_VERB_SET_CONNECT_SEL, 0x05}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc269_eeepc_amic_init_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x23, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc269_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned int bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? AMP_IN_MUTE(0) : 0; + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); +} + +static void alc269_eeepc_dmic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL, + present ? 0 : 5); +} + +static void alc269_eeepc_amic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE, + present ? AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0)); + snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE, + present ? AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1)); +} + +/* unsolicited event for HP jack sensing */ +static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc269_speaker_automute(codec); + + if ((res >> 26) == ALC880_MIC_EVENT) + alc269_eeepc_dmic_automute(codec); +} + +static void alc269_eeepc_dmic_inithook(struct hda_codec *codec) +{ + alc269_speaker_automute(codec); + alc269_eeepc_dmic_automute(codec); +} + +/* unsolicited event for HP jack sensing */ +static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc269_speaker_automute(codec); + + if ((res >> 26) == ALC880_MIC_EVENT) + alc269_eeepc_amic_automute(codec); +} + +static void alc269_eeepc_amic_inithook(struct hda_codec *codec) +{ + alc269_speaker_automute(codec); + alc269_eeepc_amic_automute(codec); +} + /* add playback controls from the parsed DAC table */ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) @@ -11188,6 +11323,9 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + spec->mixers[spec->num_mixers] = alc269_capture_mixer; + spec->num_mixers++; + return 1; } @@ -11215,12 +11353,16 @@ static const char *alc269_models[ALC269_MODEL_LAST] = { }; static struct snd_pci_quirk alc269_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", + ALC269_ASUS_EEEPC_P703), + SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901", + ALC269_ASUS_EEEPC_P901), {} }; static struct alc_config_preset alc269_presets[] = { [ALC269_BASIC] = { - .mixers = { alc269_base_mixer }, + .mixers = { alc269_base_mixer, alc269_capture_mixer }, .init_verbs = { alc269_init_verbs }, .num_dacs = ARRAY_SIZE(alc269_dac_nids), .dac_nids = alc269_dac_nids, @@ -11229,6 +11371,32 @@ static struct alc_config_preset alc269_presets[] = { .channel_mode = alc269_modes, .input_mux = &alc269_capture_source, }, + [ALC269_ASUS_EEEPC_P703] = { + .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer }, + .init_verbs = { alc269_init_verbs, + alc269_eeepc_amic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_eeepc_amic_capture_source, + .unsol_event = alc269_eeepc_amic_unsol_event, + .init_hook = alc269_eeepc_amic_inithook, + }, + [ALC269_ASUS_EEEPC_P901] = { + .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer}, + .init_verbs = { alc269_init_verbs, + alc269_eeepc_dmic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_eeepc_dmic_capture_source, + .unsol_event = alc269_eeepc_dmic_unsol_event, + .init_hook = alc269_eeepc_dmic_inithook, + }, }; static int patch_alc269(struct hda_codec *codec) @@ -11282,8 +11450,6 @@ static int patch_alc269(struct hda_codec *codec) spec->adc_nids = alc269_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); - spec->mixers[spec->num_mixers] = alc269_capture_mixer; - spec->num_mixers++; codec->patch_ops = alc_patch_ops; if (board_config == ALC269_AUTO) @@ -12994,6 +13160,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 08cb77f5188..7fdafcb0015 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -94,6 +94,9 @@ enum { STAC_INTEL_MAC_V3, STAC_INTEL_MAC_V4, STAC_INTEL_MAC_V5, + STAC_INTEL_MAC_AUTO, /* This model is selected if no module parameter + * is given, one of the above models will be + * chosen according to the subsystem id. */ /* for backward compatibility */ STAC_MACMINI, STAC_MACBOOK, @@ -1483,6 +1486,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs, [STAC_INTEL_MAC_V4] = intel_mac_v4_pin_configs, [STAC_INTEL_MAC_V5] = intel_mac_v5_pin_configs, + [STAC_INTEL_MAC_AUTO] = intel_mac_v3_pin_configs, /* for backward compatibility */ [STAC_MACMINI] = intel_mac_v3_pin_configs, [STAC_MACBOOK] = intel_mac_v5_pin_configs, @@ -1505,6 +1509,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_INTEL_MAC_V3] = "intel-mac-v3", [STAC_INTEL_MAC_V4] = "intel-mac-v4", [STAC_INTEL_MAC_V5] = "intel-mac-v5", + [STAC_INTEL_MAC_AUTO] = "intel-mac-auto", /* for backward compatibility */ [STAC_MACMINI] = "macmini", [STAC_MACBOOK] = "macbook", @@ -1576,9 +1581,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707, "Intel D945P", STAC_D945GTP5), /* other systems */ - /* Apple Mac Mini (early 2006) */ + /* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */ SND_PCI_QUIRK(0x8384, 0x7680, - "Mac Mini", STAC_INTEL_MAC_V3), + "Mac", STAC_INTEL_MAC_AUTO), /* Dell systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a7, "unknown Dell", STAC_922X_DELL_D81), @@ -3725,7 +3730,7 @@ static int patch_stac922x(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS, stac922x_models, stac922x_cfg_tbl); - if (spec->board_config == STAC_INTEL_MAC_V3) { + if (spec->board_config == STAC_INTEL_MAC_AUTO) { spec->gpio_mask = spec->gpio_dir = 0x03; spec->gpio_data = 0x03; /* Intel Macs have all same PCI SSID, so we need to check @@ -3757,6 +3762,9 @@ static int patch_stac922x(struct hda_codec *codec) case 0x106b2200: spec->board_config = STAC_INTEL_MAC_V5; break; + default: + spec->board_config = STAC_INTEL_MAC_V3; + break; } } diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 9ca11332614..54df8baf916 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -42,7 +42,7 @@ #include <sound/info.h> #include <asm/io.h> #include <asm/dma.h> -#include <asm/dreamcast/sysasic.h> +#include <mach/sysasic.h> #include "aica.h" MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>"); diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index fee5f8e5895..3f326219f1e 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -36,8 +36,8 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <asm/arch/at32ap700x.h> -#include <asm/arch/portmux.h> +#include <mach/at32ap700x.h> +#include <mach/portmux.h> #include "../codecs/wm8510.h" #include "at32-pcm.h" diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index d47492b2b6e..7ab48bd25e4 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -28,8 +28,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/arch/hardware.h> -#include <asm/arch/at91_ssc.h> +#include <mach/hardware.h> +#include <mach/at91_ssc.h> #include "at91-pcm.h" diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h index 58d0f00a07b..e5aada2cb10 100644 --- a/sound/soc/at91/at91-pcm.h +++ b/sound/soc/at91/at91-pcm.h @@ -19,7 +19,7 @@ #ifndef _AT91_PCM_H #define _AT91_PCM_H -#include <asm/arch/hardware.h> +#include <mach/hardware.h> struct at91_ssc_periph { void __iomem *base; diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index 090e607f869..5d44515e62e 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -28,9 +28,9 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <asm/arch/hardware.h> -#include <asm/arch/at91_pmc.h> -#include <asm/arch/at91_ssc.h> +#include <mach/hardware.h> +#include <mach/at91_pmc.h> +#include <mach/at91_ssc.h> #include "at91-pcm.h" #include "at91-ssc.h" diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index d532de95424..b081e83766b 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -33,8 +33,8 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <asm/hardware.h> -#include <asm/arch/gpio.h> +#include <mach/hardware.h> +#include <mach/gpio.h> #include "../codecs/wm8731.h" #include "at91-pcm.h" diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index ba4b5c199f2..9384702c7eb 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -231,7 +231,7 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) /* if both TX and RX are idle, disable PSC */ stat = au_readl(I2S_STAT(pscdata)); - if (!(stat & (PSC_I2SSTAT_RB | PSC_I2SSTAT_RB))) { + if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) { au_writel(0, I2S_CFG(pscdata)); au_sync(); au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 9fc8edd8222..1fb7f9a7aec 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -427,20 +427,20 @@ static const struct snd_soc_dapm_route audio_map[] = { {"HPOUTR", NULL, "Headphone PGA"}, {"Headphone PGA", NULL, "Right HP Mixer"}, - /* mono hp mixer */ - {"Mono HP Mixer", NULL, "Left HP Mixer"}, - {"Mono HP Mixer", NULL, "Right HP Mixer"}, + /* mono mixer */ + {"Mono Mixer", NULL, "Left HP Mixer"}, + {"Mono Mixer", NULL, "Right HP Mixer"}, /* Out3 Mux */ {"Out3 Mux", "Left", "Left HP Mixer"}, {"Out3 Mux", "Mono", "Phone Mixer"}, - {"Out3 Mux", "Left + Right", "Mono HP Mixer"}, + {"Out3 Mux", "Left + Right", "Mono Mixer"}, {"Out 3 PGA", NULL, "Out3 Mux"}, {"OUT3", NULL, "Out 3 PGA"}, /* speaker Mux */ {"Speaker Mux", "Speaker Mix", "Speaker Mixer"}, - {"Speaker Mux", "Headphone Mix", "Mono HP Mixer"}, + {"Speaker Mux", "Headphone Mix", "Mono Mixer"}, {"Speaker PGA", NULL, "Speaker Mux"}, {"LOUT2", NULL, "Speaker PGA"}, {"ROUT2", NULL, "Speaker PGA"}, diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 5e2c306399e..65fdbd81a37 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -19,9 +19,8 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <asm/mach-types.h> #include <asm/dma.h> -#include <asm/arch/hardware.h> +#include <mach/hardware.h> #include "../codecs/tlv320aic3x.h" #include "davinci-pcm.h" diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index da2bc590286..d2d3da9729f 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -132,12 +132,17 @@ struct fsl_dma_private { * Since each link descriptor has a 32-bit byte count field, we set * period_bytes_max to the largest 32-bit number. We also have no maximum * number of periods. + * + * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a + * limitation in the SSI driver requires the sample rates for playback and + * capture to be the same. */ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_JOINT_DUPLEX, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, @@ -322,14 +327,75 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, * fsl_dma_open: open a new substream. * * Each substream has its own DMA buffer. + * + * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link + * descriptors that ping-pong from one period to the next. For example, if + * there are six periods and two link descriptors, this is how they look + * before playback starts: + * + * The last link descriptor + * ____________ points back to the first + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * | | + * V V + * _________________________________________ + * | | | | | | | The DMA buffer is + * | | | | | | | divided into 6 parts + * |______|______|______|______|______|______| + * + * and here's how they look after the first period is finished playing: + * + * ____________ + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * |______________ + * | | + * V V + * _________________________________________ + * | | | | | | | + * | | | | | | | + * |______|______|______|______|______|______| + * + * The first link descriptor now points to the third period. The DMA + * controller is currently playing the second period. When it finishes, it + * will jump back to the first descriptor and play the third period. + * + * There are four reasons we do this: + * + * 1. The only way to get the DMA controller to automatically restart the + * transfer when it gets to the end of the buffer is to use chaining + * mode. Basic direct mode doesn't offer that feature. + * 2. We need to receive an interrupt at the end of every period. The DMA + * controller can generate an interrupt at the end of every link transfer + * (aka segment). Making each period into a DMA segment will give us the + * interrupts we need. + * 3. By creating only two link descriptors, regardless of the number of + * periods, we do not need to reallocate the link descriptors if the + * number of periods changes. + * 4. All of the audio data is still stored in a single, contiguous DMA + * buffer, which is what ALSA expects. We're just dividing it into + * contiguous parts, and creating a link descriptor for each one. */ static int fsl_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private; + struct ccsr_dma_channel __iomem *dma_channel; dma_addr_t ld_buf_phys; + u64 temp_link; /* Pointer to next link descriptor */ + u32 mr; unsigned int channel; int ret = 0; + unsigned int i; /* * Reject any DMA buffer whose size is not a multiple of the period @@ -390,68 +456,74 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); runtime->private_data = dma_private; + /* Program the fixed DMA controller parameters */ + + dma_channel = dma_private->dma_channel; + + temp_link = dma_private->ld_buf_phys + + sizeof(struct fsl_dma_link_descriptor); + + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->next = cpu_to_be64(temp_link); + + temp_link += sizeof(struct fsl_dma_link_descriptor); + } + /* The last link descriptor points to the first */ + dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + + /* Tell the DMA controller where the first link descriptor is */ + out_be32(&dma_channel->clndar, + CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); + out_be32(&dma_channel->eclndar, + CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); + + /* The manual says the BCR must be clear before enabling EMP */ + out_be32(&dma_channel->bcr, 0); + + /* + * Program the mode register for interrupts, external master control, + * and source/destination hold. Also clear the Channel Abort bit. + */ + mr = in_be32(&dma_channel->mr) & + ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); + + /* + * We want External Master Start and External Master Pause enabled, + * because the SSI is controlling the DMA controller. We want the DMA + * controller to be set up in advance, and then we signal only the SSI + * to start transferring. + * + * We want End-Of-Segment Interrupts enabled, because this will generate + * an interrupt at the end of each segment (each link descriptor + * represents one segment). Each DMA segment is the same thing as an + * ALSA period, so this is how we get an interrupt at the end of every + * period. + * + * We want Error Interrupt enabled, so that we can get an error if + * the DMA controller is mis-programmed somehow. + */ + mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + CCSR_DMA_MR_EMS_EN; + + /* For playback, we want the destination address to be held. For + capture, set the source address to be held. */ + mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; + + out_be32(&dma_channel->mr, mr); + return 0; } /** - * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors. + * fsl_dma_hw_params: continue initializing the DMA links * - * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link - * descriptors that ping-pong from one period to the next. For example, if - * there are six periods and two link descriptors, this is how they look - * before playback starts: - * - * The last link descriptor - * ____________ points back to the first - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * | | - * V V - * _________________________________________ - * | | | | | | | The DMA buffer is - * | | | | | | | divided into 6 parts - * |______|______|______|______|______|______| - * - * and here's how they look after the first period is finished playing: - * - * ____________ - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * |______________ - * | | - * V V - * _________________________________________ - * | | | | | | | - * | | | | | | | - * |______|______|______|______|______|______| - * - * The first link descriptor now points to the third period. The DMA - * controller is currently playing the second period. When it finishes, it - * will jump back to the first descriptor and play the third period. - * - * There are four reasons we do this: - * - * 1. The only way to get the DMA controller to automatically restart the - * transfer when it gets to the end of the buffer is to use chaining - * mode. Basic direct mode doesn't offer that feature. - * 2. We need to receive an interrupt at the end of every period. The DMA - * controller can generate an interrupt at the end of every link transfer - * (aka segment). Making each period into a DMA segment will give us the - * interrupts we need. - * 3. By creating only two link descriptors, regardless of the number of - * periods, we do not need to reallocate the link descriptors if the - * number of periods changes. - * 4. All of the audio data is still stored in a single, contiguous DMA - * buffer, which is what ALSA expects. We're just dividing it into - * contiguous parts, and creating a link descriptor for each one. + * This function obtains hardware parameters about the opened stream and + * programs the DMA controller accordingly. * * Note that due to a quirk of the SSI's STX register, the target address * for the DMA operations depends on the sample size. So we don't program @@ -463,11 +535,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; dma_addr_t temp_addr; /* Pointer to next period */ - u64 temp_link; /* Pointer to next link descriptor */ - u32 mr; /* Temporary variable for MR register */ unsigned int i; @@ -485,8 +554,6 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, dma_private->dma_buf_next = dma_private->dma_buf_phys; /* - * Initialize each link descriptor. - * * The actual address in STX0 (destination for playback, source for * capture) is based on the sample size, but we don't know the sample * size in this function, so we'll have to adjust that later. See @@ -502,16 +569,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * buffer itself. */ temp_addr = substream->dma_buffer.addr; - temp_link = dma_private->ld_buf_phys + - sizeof(struct fsl_dma_link_descriptor); for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; link->count = cpu_to_be32(period_size); - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) link->source_addr = cpu_to_be32(temp_addr); @@ -519,51 +581,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, link->dest_addr = cpu_to_be32(temp_addr); temp_addr += period_size; - temp_link += sizeof(struct fsl_dma_link_descriptor); } - /* The last link descriptor points to the first */ - dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); - - /* Tell the DMA controller where the first link descriptor is */ - out_be32(&dma_channel->clndar, - CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); - out_be32(&dma_channel->eclndar, - CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); - - /* The manual says the BCR must be clear before enabling EMP */ - out_be32(&dma_channel->bcr, 0); - - /* - * Program the mode register for interrupts, external master control, - * and source/destination hold. Also clear the Channel Abort bit. - */ - mr = in_be32(&dma_channel->mr) & - ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); - - /* - * We want External Master Start and External Master Pause enabled, - * because the SSI is controlling the DMA controller. We want the DMA - * controller to be set up in advance, and then we signal only the SSI - * to start transfering. - * - * We want End-Of-Segment Interrupts enabled, because this will generate - * an interrupt at the end of each segment (each link descriptor - * represents one segment). Each DMA segment is the same thing as an - * ALSA period, so this is how we get an interrupt at the end of every - * period. - * - * We want Error Interrupt enabled, so that we can get an error if - * the DMA controller is mis-programmed somehow. - */ - mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | - CCSR_DMA_MR_EMS_EN; - - /* For playback, we want the destination address to be held. For - capture, set the source address to be held. */ - mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; - - out_be32(&dma_channel->mr, mr); return 0; } diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 71bff33f552..157a7895ffa 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,6 +67,8 @@ * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI + * @first_stream: pointer to the stream that was opened first + * @second_stream: pointer to second stream * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened @@ -79,6 +81,8 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; struct device *dev; unsigned int playback; unsigned int capture; @@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) */ } + if (!ssi_private->first_stream) + ssi_private->first_stream = substream; + else { + /* This is the second stream open, so we need to impose sample + * rate and maybe sample size constraints. Note that this can + * cause a race condition if the second stream is opened before + * the first stream is fully initialized. + * + * We provide some protection by checking to make sure the first + * stream is initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample rate or + * size. If the second stream is opened before the first stream + * has received its final parameters, then the second stream may + * be constrained to the wrong sample rate or size. + * + * FIXME: This code does not handle opening and closing streams + * repeatedly. If you open two streams and then close the first + * one, you may not be able to open another stream until you + * close the second one as well. + */ + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + + if (!first_runtime->rate || !first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample rate and size in %s stream first\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, first_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); + + ssi_private->second_stream = substream; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ssi_private->playback++; @@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u32 wl; - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + if (substream == ssi_private->first_stream) { + u32 wl; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + /* The SSI should always be disabled at this points (SSIEN=0) */ + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); - - setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + } return 0; } @@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* * I think we need this delay to allow time for the SSI @@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ssi_private->capture--; + if (ssi_private->first_stream == substream) + ssi_private->first_stream = ssi_private->second_stream; + + ssi_private->second_stream = NULL; + /* * If this is the last active substream, disable the SSI and release * the IRQ. diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 02cec96859b..7694621ec40 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -29,9 +29,9 @@ #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <asm/arch/hardware.h> +#include <mach/hardware.h> #include <linux/gpio.h> -#include <asm/arch/mcbsp.h> +#include <mach/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 00b0c9d73cd..35310e16d7f 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -30,9 +30,9 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <asm/arch/control.h> -#include <asm/arch/dma.h> -#include <asm/arch/mcbsp.h> +#include <mach/control.h> +#include <mach/dma.h> +#include <mach/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index e092f3d836d..690bfeaec4a 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -27,7 +27,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/arch/dma.h> +#include <mach/dma.h> #include "omap-pcm.h" static const struct snd_pcm_hardware omap_pcm_hardware = { diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index c0294464a23..0a53f72077f 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -25,10 +25,10 @@ #include <asm/mach-types.h> #include <asm/hardware/scoop.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/hardware.h> -#include <asm/arch/corgi.h> -#include <asm/arch/audio.h> +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/corgi.h> +#include <mach/audio.h> #include "../codecs/wm8731.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 06e8afb2527..6781c5be242 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -21,9 +21,9 @@ #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/hardware.h> -#include <asm/arch/audio.h> +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/audio.h> #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index 02dcac39cdf..d9c3f7b28be 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -30,9 +30,9 @@ #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/hardware.h> -#include <asm/arch/audio.h> +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/audio.h> #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 65a4e9a8c39..a4697f7e292 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -26,10 +26,10 @@ #include <asm/mach-types.h> #include <asm/hardware/locomo.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/hardware.h> -#include <asm/arch/poodle.h> -#include <asm/arch/audio.h> +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/poodle.h> +#include <mach/audio.h> #include "../codecs/wm8731.h" #include "pxa2xx-pcm.h" @@ -85,17 +85,13 @@ static int poodle_startup(struct snd_pcm_substream *substream) } /* we need to unmute the HP at shutdown as the mute burns power on poodle */ -static int poodle_shutdown(struct snd_pcm_substream *substream) +static void poodle_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; - /* set = unmute headphone */ locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - return 0; } static int poodle_hw_params(struct snd_pcm_substream *substream, @@ -232,7 +228,7 @@ static const struct soc_enum poodle_enum[] = { SOC_ENUM_SINGLE_EXT(2, spk_function), }; -static const snd_kcontrol_new_t wm8731_poodle_controls[] = { +static const struct snd_kcontrol_new wm8731_poodle_controls[] = { SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack, poodle_set_jack), SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk, diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 059af815ea0..d94a495bd6b 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -26,10 +26,10 @@ #include <asm/irq.h> #include <linux/mutex.h> -#include <asm/hardware.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/pxa2xx-gpio.h> -#include <asm/arch/audio.h> +#include <mach/hardware.h> +#include <mach/pxa-regs.h> +#include <mach/pxa2xx-gpio.h> +#include <mach/audio.h> #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 8f96d87f7b4..8548818eea0 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -21,10 +21,10 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <asm/hardware.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/pxa2xx-gpio.h> -#include <asm/arch/audio.h> +#include <mach/hardware.h> +#include <mach/pxa-regs.h> +#include <mach/pxa2xx-gpio.h> +#include <mach/audio.h> #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 2df03ee5819..4345f387fe4 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -22,9 +22,9 @@ #include <sound/soc.h> #include <asm/dma.h> -#include <asm/hardware.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/audio.h> +#include <mach/hardware.h> +#include <mach/pxa-regs.h> +#include <mach/audio.h> #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 64385797da5..eefc25b8351 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -26,10 +26,10 @@ #include <asm/mach-types.h> #include <asm/hardware/scoop.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/hardware.h> -#include <asm/arch/akita.h> -#include <asm/arch/spitz.h> +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/akita.h> +#include <mach/spitz.h> #include "../codecs/wm8750.h" #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index fe6cca9c9e7..2baaa750f12 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -29,11 +29,10 @@ #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <asm/arch/tosa.h> -#include <asm/arch/pxa-regs.h> -#include <asm/arch/hardware.h> -#include <asm/arch/audio.h> -#include <asm/arch/tosa.h> +#include <mach/tosa.h> +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/audio.h> #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 4d7a9aa15f1..8089f8ee05c 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -24,14 +24,13 @@ #include <sound/soc-dapm.h> #include <sound/tlv.h> -#include <asm/mach-types.h> #include <asm/hardware/scoop.h> -#include <asm/arch/regs-clock.h> -#include <asm/arch/regs-gpio.h> -#include <asm/hardware.h> -#include <asm/arch/audio.h> +#include <mach/regs-clock.h> +#include <mach/regs-gpio.h> +#include <mach/hardware.h> +#include <mach/audio.h> #include <linux/io.h> -#include <asm/arch/spi-gpio.h> +#include <mach/spi-gpio.h> #include <asm/plat-s3c24xx/regs-iis.h> diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index ee4676ed128..ded7d995a92 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -28,16 +28,16 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> -#include <asm/hardware.h> +#include <mach/hardware.h> #include <linux/io.h> #include <asm/dma.h> #include <asm/plat-s3c24xx/regs-s3c2412-iis.h> -#include <asm/arch/regs-gpio.h> -#include <asm/arch/audio.h> -#include <asm/arch/dma.h> +#include <mach/regs-gpio.h> +#include <mach/audio.h> +#include <mach/dma.h> #include "s3c24xx-pcm.h" #include "s3c2412-i2s.h" diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 783349b7fed..19c5c3cf5d8 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -27,13 +27,13 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <asm/hardware.h> +#include <mach/hardware.h> #include <asm/plat-s3c/regs-ac97.h> -#include <asm/arch/regs-gpio.h> -#include <asm/arch/regs-clock.h> -#include <asm/arch/audio.h> +#include <mach/regs-gpio.h> +#include <mach/regs-clock.h> +#include <mach/audio.h> #include <asm/dma.h> -#include <asm/arch/dma.h> +#include <mach/dma.h> #include "s3c24xx-pcm.h" #include "s3c24xx-ac97.h" diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 397524282b5..ba4476b55fb 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -27,12 +27,12 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <asm/hardware.h> -#include <asm/arch/regs-gpio.h> -#include <asm/arch/regs-clock.h> -#include <asm/arch/audio.h> +#include <mach/hardware.h> +#include <mach/regs-gpio.h> +#include <mach/regs-clock.h> +#include <mach/audio.h> #include <asm/dma.h> -#include <asm/arch/dma.h> +#include <mach/dma.h> #include <asm/plat-s3c24xx/regs-iis.h> diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index cef79b34dc6..e13e614bada 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -27,9 +27,9 @@ #include <sound/soc.h> #include <asm/dma.h> -#include <asm/hardware.h> -#include <asm/arch/dma.h> -#include <asm/arch/audio.h> +#include <mach/hardware.h> +#include <mach/dma.h> +#include <mach/audio.h> #include "s3c24xx-pcm.h" diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061c2a6..f9d100bc847 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -470,6 +470,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, return 0; } +EXPORT_SYMBOL_GPL(dapm_reg_event); /* * Scan each dapm widget for complete audio path. @@ -523,24 +524,6 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; } - /* programmable gain/attenuation */ - if (w->id == snd_soc_dapm_pga) { - int on; - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - w->power = on = (out != 0 && in != 0) ? 1 : 0; - - if (!on) - dapm_set_pga(w, on); /* lower volume to reduce pops */ - dapm_update_bits(w); - if (on) - dapm_set_pga(w, on); /* restore volume from zero */ - - continue; - } - /* pre and post event widgets */ if (w->id == snd_soc_dapm_pre) { if (!w->event) @@ -586,45 +569,56 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) power_change = (w->power == power) ? 0: 1; w->power = power; + if (!power_change) + continue; + /* call any power change event handlers */ - if (power_change) { - if (w->event) { - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); - if (power) { - /* power up event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMU){ - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - } else { - /* power down event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - } - } else - /* no event handler */ - dapm_update_bits(w); + if (w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + /* Lower PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && !power) + dapm_set_pga(w, power); + + dapm_update_bits(w); + + /* Raise PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && power) + dapm_set_pga(w, power); + + /* power up post event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; } } } |