diff options
Diffstat (limited to 'sound')
42 files changed, 369 insertions, 193 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 91acc9a243ec..7c1fc64cb53d 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -30,6 +30,8 @@ #define DRIVER_NAME "aaci-pl041" +#define FRAME_PERIOD_US 21 + /* * PM support is not complete. Turn it off. */ @@ -48,7 +50,11 @@ static void aaci_ac97_select_codec(struct aaci *aaci, struct snd_ac97 *ac97) if (v & SLFR_1RXV) readl(aaci->base + AACI_SL1RX); - writel(maincr, aaci->base + AACI_MAINCR); + if (maincr != readl(aaci->base + AACI_MAINCR)) { + writel(maincr, aaci->base + AACI_MAINCR); + readl(aaci->base + AACI_MAINCR); + udelay(1); + } } /* @@ -64,8 +70,8 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct aaci *aaci = ac97->private_data; + int timeout; u32 v; - int timeout = 5000; if (ac97->num >= 4) return; @@ -81,14 +87,17 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, writel(val << 4, aaci->base + AACI_SL2TX); writel(reg << 12, aaci->base + AACI_SL1TX); - /* - * Wait for the transmission of both slots to complete. - */ + /* Initially, wait one frame period */ + udelay(FRAME_PERIOD_US); + + /* And then wait an additional eight frame periods for it to be sent */ + timeout = FRAME_PERIOD_US * 8; do { + udelay(1); v = readl(aaci->base + AACI_SLFR); } while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout); - if (!timeout) + if (v & (SLFR_1TXB|SLFR_2TXB)) dev_err(&aaci->dev->dev, "timeout waiting for write to complete\n"); @@ -101,9 +110,8 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct aaci *aaci = ac97->private_data; + int timeout, retries = 10; u32 v; - int timeout = 5000; - int retries = 10; if (ac97->num >= 4) return ~0; @@ -117,35 +125,34 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) */ writel((reg << 12) | (1 << 19), aaci->base + AACI_SL1TX); - /* - * Wait for the transmission to complete. - */ + /* Initially, wait one frame period */ + udelay(FRAME_PERIOD_US); + + /* And then wait an additional eight frame periods for it to be sent */ + timeout = FRAME_PERIOD_US * 8; do { + udelay(1); v = readl(aaci->base + AACI_SLFR); } while ((v & SLFR_1TXB) && --timeout); - if (!timeout) { + if (v & SLFR_1TXB) { dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n"); v = ~0; goto out; } - /* - * Give the AC'97 codec more than enough time - * to respond. (42us = ~2 frames at 48kHz.) - */ - udelay(42); + /* Now wait for the response frame */ + udelay(FRAME_PERIOD_US); - /* - * Wait for slot 2 to indicate data. - */ - timeout = 5000; + /* And then wait an additional eight frame periods for data */ + timeout = FRAME_PERIOD_US * 8; do { + udelay(1); cond_resched(); v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV); } while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout); - if (!timeout) { + if (v != (SLFR_1RXV|SLFR_2RXV)) { dev_err(&aaci->dev->dev, "timeout on RX valid\n"); v = ~0; goto out; @@ -179,6 +186,7 @@ aaci_chan_wait_ready(struct aaci_runtime *aacirun, unsigned long mask) int timeout = 5000; do { + udelay(1); val = readl(aacirun->base + AACI_SR); } while (val & mask && timeout--); } @@ -874,7 +882,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci) * Give the AC'97 codec more than enough time * to wake up. (42us = ~2 frames at 48kHz.) */ - udelay(42); + udelay(FRAME_PERIOD_US * 2); ret = snd_ac97_bus(aaci->card, 0, &aaci_bus_ops, aaci, &ac97_bus); if (ret) @@ -989,6 +997,8 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) * disabling the channel doesn't clear the FIFO. */ writel(aaci->maincr & ~MAINCR_IE, aaci->base + AACI_MAINCR); + readl(aaci->base + AACI_MAINCR); + udelay(1); writel(aaci->maincr, aaci->base + AACI_MAINCR); /* diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 10c3a871a12d..b310702c646e 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -33,9 +33,12 @@ #include <linux/dw_dmac.h> #include <mach/cpu.h> -#include <mach/hardware.h> #include <mach/gpio.h> +#ifdef CONFIG_ARCH_AT91 +#include <mach/hardware.h> +#endif + #include "ac97c.h" enum { diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 7730575bfadd..b8b31c433d64 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); struct snd_timer *t = stime->timer; + unsigned long oruns; if (!atomic_read(&stime->running)) return HRTIMER_NORESTART; - hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); - snd_timer_interrupt(stime->timer, t->sticks); + oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); + snd_timer_interrupt(stime->timer, t->sticks * oruns); if (!atomic_read(&stime->running)) return HRTIMER_NORESTART; @@ -104,7 +105,7 @@ static int snd_hrtimer_stop(struct snd_timer *t) } static struct snd_timer_hardware hrtimer_hw = { - .flags = SNDRV_TIMER_HW_AUTO, + .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET, .open = snd_hrtimer_open, .close = snd_hrtimer_close, .start = snd_hrtimer_start, diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index da03597fc893..5c426df87678 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -55,14 +55,13 @@ #include <linux/err.h> #include <linux/platform_device.h> #include <linux/ioport.h> +#include <linux/io.h> #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/rawmidi.h> #include <linux/delay.h> -#include <asm/io.h> - /* * globals */ diff --git a/sound/oss/Makefile b/sound/oss/Makefile index 96f14dcd0cd1..90ffb99c6b17 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y) $(obj)/bin2hex pss_synth < $< > $@ else $(obj)/pss_boot.h: - ( \ + $(Q)( \ echo 'static unsigned char * pss_synth = NULL;'; \ echo 'static int pss_synthLen = 0;'; \ ) > $@ @@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y) $(obj)/hex2hex -i trix_boot < $< > $@ else $(obj)/trix_boot.h: - ( \ + $(Q)( \ echo 'static unsigned char * trix_boot = NULL;'; \ echo 'static int trix_boot_len = 0;'; \ ) > $@ diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 23f49f356e0f..16c0bdfbb164 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) { static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) { stream_t *dma = &vortex->dma_adb[adbdma]; - int temp; + int temp, page, delta; temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2)); - temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1)); - return temp; + page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT; + if (dma->nr_periods >= 4) + delta = (page - dma->period_real) & 3; + else { + delta = (page - dma->period_real); + if (delta < 0) + delta += dma->nr_periods; + } + return (dma->period_virt + delta) * dma->period_bytes + + (temp & (dma->period_bytes - 1)); } static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 6117595fc075..573594bf3225 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -979,31 +979,25 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, snd_azf3328_dbgcallenter(); switch (bitrate) { -#define AZF_FMT_XLATE(in_freq, out_bits) \ - do { \ - case AZF_FREQ_ ## in_freq: \ - freq = SOUNDFORMAT_FREQ_ ## out_bits; \ - break; \ - } while (0); - AZF_FMT_XLATE(4000, SUSPECTED_4000) - AZF_FMT_XLATE(4800, SUSPECTED_4800) - /* the AZF3328 names it "5510" for some strange reason: */ - AZF_FMT_XLATE(5512, 5510) - AZF_FMT_XLATE(6620, 6620) - AZF_FMT_XLATE(8000, 8000) - AZF_FMT_XLATE(9600, 9600) - AZF_FMT_XLATE(11025, 11025) - AZF_FMT_XLATE(13240, SUSPECTED_13240) - AZF_FMT_XLATE(16000, 16000) - AZF_FMT_XLATE(22050, 22050) - AZF_FMT_XLATE(32000, 32000) + case AZF_FREQ_4000: freq = SOUNDFORMAT_FREQ_SUSPECTED_4000; break; + case AZF_FREQ_4800: freq = SOUNDFORMAT_FREQ_SUSPECTED_4800; break; + case AZF_FREQ_5512: + /* the AZF3328 names it "5510" for some strange reason */ + freq = SOUNDFORMAT_FREQ_5510; break; + case AZF_FREQ_6620: freq = SOUNDFORMAT_FREQ_6620; break; + case AZF_FREQ_8000: freq = SOUNDFORMAT_FREQ_8000; break; + case AZF_FREQ_9600: freq = SOUNDFORMAT_FREQ_9600; break; + case AZF_FREQ_11025: freq = SOUNDFORMAT_FREQ_11025; break; + case AZF_FREQ_13240: freq = SOUNDFORMAT_FREQ_SUSPECTED_13240; break; + case AZF_FREQ_16000: freq = SOUNDFORMAT_FREQ_16000; break; + case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break; + case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); /* fall-through */ - AZF_FMT_XLATE(44100, 44100) - AZF_FMT_XLATE(48000, 48000) - AZF_FMT_XLATE(66200, SUSPECTED_66200) -#undef AZF_FMT_XLATE + case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break; + case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break; + case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break; } /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */ /* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 4a663471dadc..74b0560289c0 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) snd_print_pcm_rates(a->rates, buf, sizeof(buf)); if (a->format == AUDIO_CODING_TYPE_LPCM) - snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8)); + snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8); else if (a->max_bitrate) snprintf(buf2, sizeof(buf2), ", max bitrate = %d", a->max_bitrate); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2e91a991eb15..fcedad9a5fef 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), @@ -2703,7 +2704,7 @@ static int __devinit azx_probe(struct pci_dev *pci, if (err < 0) goto out_free; #ifdef CONFIG_SND_HDA_PATCH_LOADER - if (patch[dev]) { + if (patch[dev] && *patch[dev]) { snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", patch[dev]); err = snd_hda_load_patch(chip->bus, patch[dev]); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9bb030a469cd..dd7c5c12225d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -85,6 +85,7 @@ struct conexant_spec { unsigned int auto_mic; int auto_mic_ext; /* autocfg.inputs[] index for ext mic */ unsigned int need_dac_fix; + hda_nid_t slave_dig_outs[2]; /* capture */ unsigned int num_adc_nids; @@ -127,6 +128,7 @@ struct conexant_spec { unsigned int ideapad:1; unsigned int thinkpad:1; unsigned int hp_laptop:1; + unsigned int asus:1; unsigned int ext_mic_present; unsigned int recording; @@ -352,6 +354,8 @@ static int conexant_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; } + if (spec->slave_dig_outs[0]) + codec->slave_dig_outs = spec->slave_dig_outs; } return 0; @@ -403,10 +407,16 @@ static int conexant_add_jack(struct hda_codec *codec, struct conexant_spec *spec; struct conexant_jack *jack; const char *name; - int err; + int i, err; spec = codec->spec; snd_array_init(&spec->jacks, sizeof(*jack), 32); + + jack = spec->jacks.list; + for (i = 0; i < spec->jacks.used; i++, jack++) + if (jack->nid == nid) + return 0 ; /* already present */ + jack = snd_array_new(&spec->jacks); name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ; @@ -2100,7 +2110,7 @@ static int patch_cxt5051(struct hda_codec *codec) static hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; -#define CXT5066_SPDIF_OUT 0x21 +static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 }; /* OLPC's microphone port is DC coupled for use with external sensors, * therefore we use a 50% mic bias in order to center the input signal with @@ -2312,6 +2322,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } + +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_asus_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 0); +} + + /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_hp_laptop_automic(struct hda_codec *codec) { @@ -2387,79 +2410,55 @@ static void cxt5066_hp_automute(struct hda_codec *codec) cxt5066_update_speaker(codec); } -/* unsolicited event for jack sensing */ -static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) +/* Dispatch the right mic autoswitch function */ +static void cxt5066_automic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: - /* ignore mic events in DC mode; we're always using the jack */ - if (!spec->dc_enable) - cxt5066_olpc_automic(codec); - break; - } -} -/* unsolicited event for jack sensing */ -static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res) -{ - snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: + if (spec->dell_vostro) cxt5066_vostro_automic(codec); - break; - } -} - -/* unsolicited event for jack sensing */ -static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) -{ - snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26); - switch (res >> 26) { - case CONEXANT_HP_EVENT: - cxt5066_hp_automute(codec); - break; - case CONEXANT_MIC_EVENT: + else if (spec->ideapad) cxt5066_ideapad_automic(codec); - break; - } + else if (spec->thinkpad) + cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); + else if (spec->asus) + cxt5066_asus_automic(codec); } /* unsolicited event for jack sensing */ -static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + struct conexant_spec *spec = codec->spec; + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_hp_laptop_automic(codec); + /* ignore mic events in DC mode; we're always using the jack */ + if (!spec->dc_enable) + cxt5066_olpc_automic(codec); break; } } /* unsolicited event for jack sensing */ -static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26); + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_thinkpad_automic(codec); + cxt5066_automic(codec); break; } } + static const struct hda_input_mux cxt5066_analog_mic_boost = { .num_items = 5, .items = { @@ -2633,6 +2632,27 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) spec->recording = 0; } +static void conexant_check_dig_outs(struct hda_codec *codec, + hda_nid_t *dig_pins, + int num_pins) +{ + struct conexant_spec *spec = codec->spec; + hda_nid_t *nid_loc = &spec->multiout.dig_out_nid; + int i; + + for (i = 0; i < num_pins; i++, dig_pins++) { + unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins); + if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE) + continue; + if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1) + continue; + if (spec->slave_dig_outs[0]) + nid_loc++; + else + nid_loc = spec->slave_dig_outs; + } +} + static struct hda_input_mux cxt5066_capture_source = { .num_items = 4, .items = { @@ -3039,20 +3059,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; - snd_printdd("CXT5066: init\n"); conexant_init(codec); if (codec->patch_ops.unsol_event) { cxt5066_hp_automute(codec); - if (spec->dell_vostro) - cxt5066_vostro_automic(codec); - else if (spec->ideapad) - cxt5066_ideapad_automic(codec); - else if (spec->thinkpad) - cxt5066_thinkpad_automic(codec); - else if (spec->hp_laptop) - cxt5066_hp_laptop_automic(codec); + cxt5066_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3080,6 +3091,7 @@ enum { CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3091,6 +3103,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTRO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", }; @@ -3102,7 +3115,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), + SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), + SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), @@ -3111,7 +3126,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} }; @@ -3133,7 +3150,8 @@ static int patch_cxt5066(struct hda_codec *codec) spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids); spec->multiout.dac_nids = cxt5066_dac_nids; - spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT; + conexant_check_dig_outs(codec, cxt5066_digout_pin_nids, + ARRAY_SIZE(cxt5066_digout_pin_nids)); spec->num_adc_nids = 1; spec->adc_nids = cxt5066_adc_nids; spec->capsrc_nids = cxt5066_capsrc_nids; @@ -3167,17 +3185,20 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_ASUS: case CXT5066_HP_LAPTOP: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; spec->num_init_verbs++; - spec->hp_laptop = 1; + spec->hp_laptop = board_config == CXT5066_HP_LAPTOP; + spec->asus = board_config == CXT5066_ASUS; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; /* no S/PDIF out */ - spec->multiout.dig_out_nid = 0; + if (board_config == CXT5066_HP_LAPTOP) + spec->multiout.dig_out_nid = 0; /* input source automatically selected */ spec->input_mux = NULL; spec->port_d_mode = 0; @@ -3207,7 +3228,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_DELL_VOSTRO: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_vostro_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; @@ -3224,7 +3245,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_IDEAPAD: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_ideapad_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->init_verbs[0] = cxt5066_init_verbs_ideapad; @@ -3240,7 +3261,7 @@ static int patch_cxt5066(struct hda_codec *codec) break; case CXT5066_THINKPAD: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_thinkpad_event; + codec->patch_ops.unsol_event = cxt5066_unsol_event; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->init_verbs[0] = cxt5066_init_verbs_thinkpad; @@ -3389,7 +3410,7 @@ static void cx_auto_parse_output(struct hda_codec *codec) } } spec->multiout.dac_nids = spec->private_dac_nids; - spec->multiout.max_channels = nums * 2; + spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (cfg->hp_outs > 0) spec->auto_mute = 1; @@ -3708,9 +3729,9 @@ static int cx_auto_init(struct hda_codec *codec) return 0; } -static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, +static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir) + hda_nid_t nid, int hda_dir, int amp_idx) { static char name[32]; static struct snd_kcontrol_new knew[] = { @@ -3722,7 +3743,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir); + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]); @@ -3738,6 +3760,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, return 0; } +#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -3787,29 +3812,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; static const char *prev_label; - int i, err, cidx; + int i, err, cidx, conn_len; + hda_nid_t conn[HDA_MAX_CONNECTIONS]; + + int multi_adc_volume = 0; /* If the ADC nid has several input volumes */ + int adc_nid = spec->adc_nids[0]; + + conn_len = snd_hda_get_connections(codec, adc_nid, conn, + HDA_MAX_CONNECTIONS); + if (conn_len < 0) + return conn_len; + + multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1; + if (!multi_adc_volume) { + err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid, + HDA_INPUT); + if (err < 0) + return err; + } - err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0], - HDA_INPUT); - if (err < 0) - return err; prev_label = NULL; cidx = 0; for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; const char *label; - if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) + int j; + int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP; + if (!pin_amp && !multi_adc_volume) continue; + label = hda_get_autocfg_input_label(codec, cfg, i); if (label == prev_label) cidx++; else cidx = 0; prev_label = label; - err = cx_auto_add_volume(codec, label, " Capture", cidx, - nid, HDA_INPUT); - if (err < 0) - return err; + + if (pin_amp) { + err = cx_auto_add_volume(codec, label, " Boost", cidx, + nid, HDA_INPUT); + if (err < 0) + return err; + } + + if (!multi_adc_volume) + continue; + for (j = 0; j < conn_len; j++) { + if (conn[j] == nid) { + err = cx_auto_add_volume_idx(codec, label, + " Capture", cidx, adc_nid, HDA_INPUT, j); + if (err < 0) + return err; + break; + } + } } return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2d5b83fa8d24..a58767736727 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -642,6 +642,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, hdmi_ai->ver = 0x01; hdmi_ai->len = 0x0a; hdmi_ai->CC02_CT47 = channels - 1; + hdmi_ai->CA = ca; hdmi_checksum_audio_infoframe(hdmi_ai); } else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */ struct dp_audio_infoframe *dp_ai; @@ -651,6 +652,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, dp_ai->len = 0x1b; dp_ai->ver = 0x11 << 2; dp_ai->CC02_CT47 = channels - 1; + dp_ai->CA = ca; } else { snd_printd("HDMI: unknown connection type at pin %d\n", pin_nid); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index be4df4c6fd56..3328a259a242 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2290,6 +2290,29 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0f, 2, 0x0, + HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + + static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -10359,7 +10382,7 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc_automute_amp, }, [ALC888_ACER_ASPIRE_4930G] = { - .mixers = { alc888_base_mixer, + .mixers = { alc888_acer_aspire_4930g_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, alc888_acer_aspire_4930g_verbs }, @@ -14954,9 +14977,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), - SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), - SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), {} @@ -18800,6 +18825,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E), {} }; @@ -19492,6 +19518,7 @@ static const struct alc_fixup alc662_fixups[] = { }; static struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index c2ae63d17cd2..f53897a708b4 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -92,6 +92,8 @@ struct oxygen_model { void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed); + unsigned int (*adjust_dac_routing)(struct oxygen *chip, + unsigned int play_routing); void (*gpio_changed)(struct oxygen *chip); void (*uart_input)(struct oxygen *chip); void (*ac97_switch)(struct oxygen *chip, diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 9bff14d5895d..26c7e8bcb229 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -180,6 +180,8 @@ void oxygen_update_dac_routing(struct oxygen *chip) (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); + if (chip->model.adjust_dac_routing) + reg_value = chip->model.adjust_dac_routing(chip, reg_value); oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value, OXYGEN_PLAY_DAC0_SOURCE_MASK | OXYGEN_PLAY_DAC1_SOURCE_MASK | diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 9f72d424969c..252719101c42 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -392,7 +392,7 @@ static void dump_d1_registers(struct oxygen *chip, unsigned int i; snd_iprintf(buffer, "\nCS4398: 7?"); - for (i = 2; i <= 8; ++i) + for (i = 2; i < 8; ++i) snd_iprintf(buffer, " %02x", data->cs4398_regs[i]); snd_iprintf(buffer, "\n"); dump_cs4362a_registers(data, buffer); diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index e1fa602eba79..bc6eb58be380 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -24,6 +24,11 @@ * * SPI 0 -> CS4245 * + * I²S 1 -> CS4245 + * I²S 2 -> CS4361 (center/LFE) + * I²S 3 -> CS4361 (surround) + * I²S 4 -> CS4361 (front) + * * GPIO 3 <- ? * GPIO 4 <- headphone detect * GPIO 5 -> route input jack to line-in (0) or mic-in (1) @@ -36,6 +41,7 @@ * input 1 <- aux * input 2 <- front mic * input 4 <- line/mic + * DAC out -> headphones * aux out -> front panel headphones */ @@ -207,6 +213,35 @@ static void set_cs4245_adc_params(struct oxygen *chip, cs4245_write_cached(chip, CS4245_ADC_CTRL, value); } +static inline unsigned int shift_bits(unsigned int value, + unsigned int shift_from, + unsigned int shift_to, + unsigned int mask) +{ + if (shift_from < shift_to) + return (value << (shift_to - shift_from)) & mask; + else + return (value >> (shift_from - shift_to)) & mask; +} + +static unsigned int adjust_dg_dac_routing(struct oxygen *chip, + unsigned int play_routing) +{ + return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC1_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC1_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_SHIFT, + OXYGEN_PLAY_DAC2_SOURCE_MASK) | + shift_bits(play_routing, + OXYGEN_PLAY_DAC0_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_SHIFT, + OXYGEN_PLAY_DAC3_SOURCE_MASK); +} + static int output_switch_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -557,6 +592,7 @@ struct oxygen_model model_xonar_dg = { .resume = dg_resume, .set_dac_params = set_cs4245_dac_params, .set_adc_params = set_cs4245_adc_params, + .adjust_dac_routing = adjust_dg_dac_routing, .dump_registers = dump_cs4245_registers, .model_data_size = sizeof(struct dg), .device_config = PLAYBACK_0_TO_I2S | diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h index bd26e092aead..6ce9ad700290 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.h +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h @@ -22,7 +22,7 @@ #define __PDAUDIOCF_H #include <sound/pcm.h> -#include <asm/io.h> +#include <linux/io.h> #include <linux/interrupt.h> #include <pcmcia/cistpl.h> #include <pcmcia/ds.h> diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 989e04abb520..fe33e122e372 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -23,8 +23,8 @@ #include <linux/delay.h> #include <linux/device.h> #include <linux/firmware.h> +#include <linux/io.h> #include <sound/core.h> -#include <asm/io.h> #include "vxpocket.h" diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index da2208e06b0d..5e4d499d8434 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = { .cpu_dai_name = "atmel-ssc-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-0x1a", + .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, .ops = &afeb9260_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index e902b24c1856..ad28663f5bbd 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-0x1b", + .codec_name = "ssm2602-codec.0-001b", .ops = &bf5xx_ssm2602_ops, }; diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd067f79..347a567b01e1 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; + struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec); davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 03d1e860d229..bb4bf65b9e7e 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) return 0; } +static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC; + static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), .read = cx20442_read_reg_cache, diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247a6a99feb8..37b8aa8a680f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, @@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 11, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, @@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), @@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1R" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2R" }, + { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ @@ -1578,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Headphone Mux", "DAC", "DAC1R" }, }; +static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { + { "AIF1DACDAT", NULL, "AIF2DACDAT" }, + { "AIF2DACDAT", NULL, "AIF1DACDAT" }, + { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, + { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, +}; + static const struct snd_soc_dapm_route wm8994_intercon[] = { { "AIF2DACL", NULL, "AIF2DAC Mux" }, { "AIF2DACR", NULL, "AIF2DAC Mux" }, @@ -2386,7 +2399,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 @@ -3129,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); + + if (wm8994->revision < 4) + snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, + ARRAY_SIZE(wm8994_revd_intercon)); + break; case WM8958: snd_soc_dapm_add_routes(dapm, wm8958_intercon, diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 6045cbde492b..608c84c5aa8e 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate) else val = 0; - return snd_soc_update_bits(codec, reg, mask, reg); + return snd_soc_update_bits(codec, reg, mask, val); } /* The size in bits of the FLL divide multiplied by 10 diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c466982eed23..613df5db0b32 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + s8 offset; u16 reg, reg_l, reg_r, dcs_cfg; /* If we're using a digital only path and have a previously @@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) hubs->dcs_codes); /* HPOUT1L */ - 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; + offset = reg_l; + offset += hubs->dcs_codes; + dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ - if (reg_r + hubs->dcs_codes > 0 && - reg_r + hubs->dcs_codes < 0xff) - reg_r += hubs->dcs_codes; - dcs_cfg |= reg_r; + offset = reg_r; + offset += hubs->dcs_codes; + dcs_cfg |= (u8)offset; dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 0c2d6bacc681..fe7984221eb9 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -218,12 +218,24 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_spdif_ops, }, }; -static struct snd_soc_dai_link da8xx_evm_dai = { + +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcasp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, @@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da850_evm_dai, .num_links = 1, }; diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 2101bdcee21f..3167be689621 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Set up digital mute if not provided by the codec */ if (!codec_dai->driver->ops) { codec_dai->driver->ops = &ams_delta_dai_ops; - } else if (!codec_dai->driver->ops->digital_mute) { - codec_dai->driver->ops->digital_mute = ams_delta_digital_mute; } else { ams_delta_ops.startup = ams_delta_startup; ams_delta_ops.shutdown = ams_delta_shutdown; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index fc592f0d5fc7..784cff5f67e8 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", .stream_name = "WM8731", - .cpu_dai_name = "pxa-is2-dai", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec-0.001a", + .codec_name = "wm8731-codec-0.001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 6298ee115e27..a7d4999f9b24 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .codec_name = "wm8731-codec.0-001b", .init = poodle_wm8731_init, .ops = &poodle_ops, }; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index c2acb69b957a..8e1571350630 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link spitz_dai = { .name = "wm8750", .stream_name = "WM8750", - .cpu_dai_name = "pxa-is2", + .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8750-codec.0-001a", + .codec_name = "wm8750-codec.0-001b", .init = spitz_wm8750_init, .ops = &spitz_ops, }; diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 3eec610c10f9..0d0ae2b9eef6 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -397,11 +397,11 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", .stream_name = "WM8753 HiFi", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", .init = neo1973_gta02_wm8753_init, .platform_name = "samsung-audio", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_gta02_hifi_ops, }, { /* Voice via BT */ @@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = { .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", .ops = &neo1973_gta02_voice_ops, - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .platform_name = "samsung-audio", }, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c7a24514beb5..d20815d5ab2e 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -559,9 +559,9 @@ static struct snd_soc_dai_link neo1973_dai[] = { .name = "WM8753", .stream_name = "WM8753 HiFi", .platform_name = "samsung-audio", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .init = neo1973_wm8753_init, .ops = &neo1973_hifi_ops, }, @@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .platform_name = "samsung-audio", .cpu_dai_name = "bluetooth-dai", .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-0x1a", + .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, }, }; diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index bb4292e3596c..08fcaaa66907 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -94,8 +94,8 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic33 = { .name = "tlv320aic33", .stream_name = "TLV320AIC33", - .codec_name = "tlv320aic3x-codec.0-0x1a", - .cpu_dai_name = "s3c24xx-i2s", + .codec_name = "tlv320aic3x-codec.0-001a", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", .init = simtec_hermes_init, diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index fbba4e367729..116e3e670167 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -85,8 +85,8 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link simtec_dai_aic23 = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", - .codec_name = "tlv320aic3x-codec.0-0x1a", - .cpu_dai_name = "s3c24xx-i2s", + .codec_name = "tlv320aic3x-codec.0-001a", + .cpu_dai_name = "s3c24xx-iis", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "samsung-audio", .init = simtec_tlv320aic23_init, diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index cdc8ecbcb8ef..2c09e93dd566 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { .stream_name = "UDA134X", .codec_name = "uda134x-hifi", .codec_dai_name = "uda134x-hifi", - .cpu_dai_name = "s3c24xx-i2s", + .cpu_dai_name = "s3c24xx-iis", .ops = &s3c24xx_uda134x_ops, .platform_name = "samsung-audio", }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bac7291b6ff6..c3f6f1e72790 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd = &card->rtd_aux[num]; name = aux_dev->name; } + rtd->card = card; /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; @@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card, /* register the rtd device */ rtd->codec = codec; - rtd->card = card; rtd->dev.parent = card->dev; rtd->dev.release = rtd_release; rtd->dev.init_name = name; @@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) goto out; found: - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - ret = soc_probe_codec(card, codec); if (ret < 0) return ret; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 499730ab5638..8194f150bab7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1742,7 +1742,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned int val, val_mask; + unsigned int val; int connect, change; struct snd_soc_dapm_update update; @@ -1750,13 +1750,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - val_mask = mask << shift; + mask = mask << shift; val = val << shift; mutex_lock(&widget->codec->mutex); widget->value = val; - change = snd_soc_test_bits(widget->codec, reg, val_mask, val); + change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { if (val) /* new connection */ diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 68b97477577b..66eabafb1c24 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -785,7 +785,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) } dev->pcm->private_data = dev; - strcpy(dev->pcm->name, dev->product_name); + strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name)); memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 2f218c77fff2..a1a47088fd0c 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -136,7 +136,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (ret < 0) return ret; - strcpy(rmidi->name, device->product_name); + strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name)); rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; rmidi->private_data = device; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7df89b3d7ded..85af6051b52d 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -95,7 +95,7 @@ enum { }; -/*E-mu 0202(0404) eXtension Unit(XU) control*/ +/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/ enum { USB_XU_CLOCK_RATE = 0xe301, USB_XU_CLOCK_SOURCE = 0xe302, @@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre + /* E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 35999874d301..921a86fd9884 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -79,6 +79,13 @@ .idProduct = 0x3f0a, .bInterfaceClass = USB_CLASS_AUDIO, }, +{ + /* E-Mu 0204 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f19, + .bInterfaceClass = USB_CLASS_AUDIO, +}, /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index cf8bf088394b..e314cdb85003 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat } /* - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device, * not for interface. */ @@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; } |