diff options
Diffstat (limited to 'sound')
29 files changed, 227 insertions, 151 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index b2a2db47aff5..4247406160e7 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -28,11 +28,6 @@ config SOUND and read <file:Documentation/sound/oss/README.modules>; the module will be called soundcore. - I'm told that even without a sound card, you can make your computer - say more than an occasional beep, by programming the PC speaker. - Kernel patches and supporting utilities to do that are in the pcsp - package, available at <ftp://ftp.infradead.org/pub/pcsp/>. - source "sound/oss/dmasound/Kconfig" if !M68K diff --git a/sound/core/sound.c b/sound/core/sound.c index 812f91b3de5b..6c8ab48c689a 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -259,8 +259,9 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, return minor; } snd_minors[minor] = preg; - preg->dev = device_create(sound_class, device, MKDEV(major, minor), - "%s", name); + preg->dev = device_create_drvdata(sound_class, device, + MKDEV(major, minor), + private_data, "%s", name); if (IS_ERR(preg->dev)) { snd_minors[minor] = NULL; mutex_unlock(&sound_mutex); @@ -269,9 +270,6 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, return minor; } - if (preg->dev) - dev_set_drvdata(preg->dev, private_data); - mutex_unlock(&sound_mutex); return 0; } diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index a78a8d045175..602b58e3b55d 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -5,9 +5,10 @@ menu "Generic devices" config SND_PCSP - tristate "Internal PC speaker support" - depends on X86_PC && HIGH_RES_TIMERS + tristate "PC-Speaker support (READ HELP!)" + depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS depends on INPUT + depends on EXPERIMENTAL depends on SND select SND_PCM help @@ -18,11 +19,21 @@ config SND_PCSP You can compile this as a module which will be called snd-pcsp. + WARNING: if you already have a soundcard, enabling this + driver may lead to a problem. Namely, it may get loaded + before the other sound driver of yours, making the + pc-speaker a default sound device. Which is likely not + what you want. To make this driver play nicely with other + sound driver, you can add this into your /etc/modprobe.conf: + options snd-pcsp index=2 + You don't need this driver if you only want your pc-speaker to beep. You don't need this driver if you have a tablet piezo beeper in your PC instead of the real speaker. - It should not hurt to say Y or M here in all other cases. + Say N if you have a sound card. + Say M if you don't. + Say Y only if you really know what you do. config SND_MPU401_UART tristate diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 59203511e77d..1899cf0685bc 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -96,7 +96,7 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) return -EINVAL; hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); - pcsp_chip.timer.cb_mode = HRTIMER_CB_IRQSAFE; + pcsp_chip.timer.cb_mode = HRTIMER_CB_SOFTIRQ; pcsp_chip.timer.function = pcsp_do_timer; card = snd_card_new(index, id, THIS_MODULE, 0); @@ -194,6 +194,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) spin_unlock_irq(&chip->substream_lock); } +#ifdef CONFIG_PM static int pcsp_suspend(struct platform_device *dev, pm_message_t state) { struct snd_pcsp *chip = platform_get_drvdata(dev); @@ -201,6 +202,9 @@ static int pcsp_suspend(struct platform_device *dev, pm_message_t state) snd_pcm_suspend_all(chip->pcm); return 0; } +#else +#define pcsp_suspend NULL +#endif /* CONFIG_PM */ static void pcsp_shutdown(struct platform_device *dev) { diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h index f07cc1ee1fe7..1d661f795e8c 100644 --- a/sound/drivers/pcsp/pcsp.h +++ b/sound/drivers/pcsp/pcsp.h @@ -24,7 +24,8 @@ static DEFINE_SPINLOCK(i8253_lock); /* default timer freq for PC-Speaker: 18643 Hz */ #define DIV_18KHZ 64 #define MAX_DIV DIV_18KHZ -#define CUR_DIV() (MAX_DIV >> chip->treble) +#define CALC_DIV(d) (MAX_DIV >> (d)) +#define CUR_DIV() CALC_DIV(chip->treble) #define PCSP_MAX_TREBLE 1 /* unfortunately, with hrtimers 37KHz does not work very well :( */ @@ -36,7 +37,8 @@ static DEFINE_SPINLOCK(i8253_lock); #define PCSP_DEFAULT_SDIV (DIV_18KHZ >> 1) #define PCSP_DEFAULT_SRATE (PIT_TICK_RATE / PCSP_DEFAULT_SDIV) #define PCSP_INDEX_INC() (1 << (PCSP_MAX_TREBLE - chip->treble)) -#define PCSP_RATE() (PIT_TICK_RATE / CUR_DIV()) +#define PCSP_CALC_RATE(i) (PIT_TICK_RATE / CALC_DIV(i)) +#define PCSP_RATE() PCSP_CALC_RATE(chip->treble) #define PCSP_MIN_RATE__1 MAX_DIV/PIT_TICK_RATE #define PCSP_MAX_RATE__1 MIN_DIV/PIT_TICK_RATE #define PCSP_MAX_PERIOD_NS (1000000000ULL * PCSP_MIN_RATE__1) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index ac6238e93513..e341f3f83b6a 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -9,7 +9,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <sound/pcm.h> -#include <linux/interrupt.h> #include <asm/io.h> #include "pcsp.h" @@ -18,36 +17,12 @@ module_param(nforce_wa, bool, 0444); MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround " "(expect bad sound)"); -static void pcsp_start_timer(unsigned long dummy) -{ - hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); -} - -/* - * We need the hrtimer_start as a tasklet to avoid - * the nasty locking problem. :( - * The problem: - * - The timer handler is called with the cpu_base->lock - * already held by hrtimer code. - * - snd_pcm_period_elapsed() takes the - * substream->self_group.lock. - * So far so good. - * But the snd_pcsp_trigger() is called with the - * substream->self_group.lock held, and it calls - * hrtimer_start(), which takes the cpu_base->lock. - * You see the problem. We have the code pathes - * which take two locks in a reverse order. This - * can deadlock and the lock validator complains. - * The only solution I could find was to move the - * hrtimer_start() into a tasklet. -stsp - */ -static DECLARE_TASKLET(pcsp_start_timer_tasklet, pcsp_start_timer, 0); +#define DMIX_WANTS_S16 1 enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) { - unsigned long flags; unsigned char timer_cnt, val; - int periods_elapsed; + int fmt_size, periods_elapsed; u64 ns; size_t period_bytes, buffer_bytes; struct snd_pcm_substream *substream; @@ -64,9 +39,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) return HRTIMER_RESTART; } - /* hrtimer calls us from both hardirq and softirq contexts, - * so irqsave :( */ - spin_lock_irqsave(&chip->substream_lock, flags); + spin_lock_irq(&chip->substream_lock); /* Takashi Iwai says regarding this extra lock: If the irq handler handles some data on the DMA buffer, it should @@ -92,8 +65,11 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) goto exit_nr_unlock2; runtime = substream->runtime; - /* assume it is u8 mono */ - val = runtime->dma_area[chip->playback_ptr]; + fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3; + /* assume it is mono! */ + val = runtime->dma_area[chip->playback_ptr + fmt_size - 1]; + if (snd_pcm_format_signed(runtime->format)) + val ^= 0x80; timer_cnt = val * CUR_DIV() / 256; if (timer_cnt && chip->enable) { @@ -111,12 +87,14 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) period_bytes = snd_pcm_lib_period_bytes(substream); buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - chip->playback_ptr += PCSP_INDEX_INC(); + chip->playback_ptr += PCSP_INDEX_INC() * fmt_size; periods_elapsed = chip->playback_ptr - chip->period_ptr; if (periods_elapsed < 0) { - printk(KERN_WARNING "PCSP: playback_ptr inconsistent " +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: buffer_bytes mod period_bytes != 0 ? " "(%zi %zi %zi)\n", chip->playback_ptr, period_bytes, buffer_bytes); +#endif periods_elapsed += buffer_bytes; } periods_elapsed /= period_bytes; @@ -132,7 +110,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) chip->period_ptr %= buffer_bytes; } - spin_unlock_irqrestore(&chip->substream_lock, flags); + spin_unlock_irq(&chip->substream_lock); if (!atomic_read(&chip->timer_active)) return HRTIMER_NORESTART; @@ -146,7 +124,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) exit_nr_unlock2: snd_pcm_stream_unlock(substream); exit_nr_unlock1: - spin_unlock_irqrestore(&chip->substream_lock, flags); + spin_unlock_irq(&chip->substream_lock); return HRTIMER_NORESTART; } @@ -167,7 +145,7 @@ static void pcsp_start_playing(struct snd_pcsp *chip) atomic_set(&chip->timer_active, 1); chip->thalf = 0; - tasklet_schedule(&pcsp_start_timer_tasklet); + hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); } static void pcsp_stop_playing(struct snd_pcsp *chip) @@ -270,7 +248,11 @@ static struct snd_pcm_hardware snd_pcsp_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_HALF_DUPLEX | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_U8, + .formats = (SNDRV_PCM_FMTBIT_U8 +#if DMIX_WANTS_S16 + | SNDRV_PCM_FMTBIT_S16_LE +#endif + ), .rates = SNDRV_PCM_RATE_KNOT, .rate_min = PCSP_DEFAULT_SRATE, .rate_max = PCSP_DEFAULT_SRATE, diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 64a695fef74e..caeb0f57fcca 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -50,7 +50,8 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol, uinfo->value.enumerated.items = chip->max_treble + 1; if (uinfo->value.enumerated.item > chip->max_treble) uinfo->value.enumerated.item = chip->max_treble; - sprintf(uinfo->value.enumerated.name, "%d", PCSP_RATE()); + sprintf(uinfo->value.enumerated.name, "%d", + PCSP_CALC_RATE(uinfo->value.enumerated.item)); return 0; } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 91d14224f6b3..73d4572d136b 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -925,7 +925,7 @@ static unsigned char als4000_saved_regs[] = { static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs) { unsigned char *val = chip->saved_regs; - snd_assert(num_regs > ARRAY_SIZE(chip->saved_regs), return); + snd_assert(num_regs <= ARRAY_SIZE(chip->saved_regs), return); for (; num_regs; num_regs--) *val++ = snd_sbmixer_read(chip, *regs++); } @@ -933,7 +933,7 @@ static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs) static void restore_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs) { unsigned char *val = chip->saved_regs; - snd_assert(num_regs > ARRAY_SIZE(chip->saved_regs), return); + snd_assert(num_regs <= ARRAY_SIZE(chip->saved_regs), return); for (; num_regs; num_regs--) snd_sbmixer_write(chip, *regs++, *val++); } diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 857008bb7167..3be2dc1025b5 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -79,7 +79,7 @@ config SOUND_TRIDENT config SOUND_MSNDCLAS tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" - depends on SOUND_PRIME && (m || !STANDALONE) + depends on SOUND_PRIME && (m || !STANDALONE) && ISA help Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or Monterey (not for the Pinnacle or Fiji). @@ -143,7 +143,7 @@ config MSNDCLAS_IO config SOUND_MSNDPIN tristate "Support for Turtle Beach MultiSound Pinnacle, Fiji" - depends on SOUND_PRIME && (m || !STANDALONE) + depends on SOUND_PRIME && (m || !STANDALONE) && ISA help Say M here if you have a Turtle Beach MultiSound Pinnacle or Fiji. See <file:Documentation/sound/oss/MultiSound> for important information diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index dfe670f12e67..eb9bc365530d 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -67,7 +67,7 @@ static int __devinit probe_one(struct pci_dev *pdev, const struct pci_device_id return 1; mem = ioremap(base, 128); - if(mem == 0UL) + if (!mem) return 1; map = readw(mem + 0x18); /* Read the SMI enables */ iounmap(mem); diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 581debf37dcb..7e4742109572 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -515,19 +515,16 @@ config SND_FM801 config SND_FM801_TEA575X_BOOL bool "ForteMedia FM801 + TEA5757 tuner" depends on SND_FM801 + depends on VIDEO_V4L1=y || VIDEO_V4L1=SND_FM801 help Say Y here to include support for soundcards based on the ForteMedia FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media Forte SF256-PCS-02) into the snd-fm801 driver. - This will enable support for the old V4L1 API. - config SND_FM801_TEA575X tristate depends on SND_FM801_TEA575X_BOOL default SND_FM801 - select VIDEO_V4L1 - select VIDEO_DEV config SND_HDA_INTEL tristate "Intel HD Audio" diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 39198e505b12..1292dcee072d 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1971,6 +1971,9 @@ static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd val = ac97->regs[AC97_AD_MISC]; ucontrol->value.integer.value[0] = !(val & AC97_AD198X_LOSEL); + if (ac97->spec.ad18xx.lo_as_master) + ucontrol->value.integer.value[0] = + !ucontrol->value.integer.value[0]; return 0; } @@ -1979,8 +1982,10 @@ static int snd_ac97_ad1888_lohpsel_put(struct snd_kcontrol *kcontrol, struct snd struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = !ucontrol->value.integer.value[0] - ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0; + val = !ucontrol->value.integer.value[0]; + if (ac97->spec.ad18xx.lo_as_master) + val = !val; + val = val ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0; return snd_ac97_update_bits(ac97, AC97_AD_MISC, AC97_AD198X_LOSEL | AC97_AD198X_HPSEL, val); } @@ -2031,7 +2036,7 @@ static void ad1888_update_jacks(struct snd_ac97 *ac97) { unsigned short val = 0; /* clear LODIS if shared jack is to be used for Surround out */ - if (is_shared_linein(ac97)) + if (!ac97->spec.ad18xx.lo_as_master && is_shared_linein(ac97)) val |= (1 << 12); /* clear CLDIS if shared jack is to be used for C/LFE out */ if (is_shared_micin(ac97)) @@ -2067,9 +2072,13 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = { static int patch_ad1888_specific(struct snd_ac97 *ac97) { - /* rename 0x04 as "Master" and 0x02 as "Master Surround" */ - snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Master Surround Playback"); - snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Master Playback"); + if (!ac97->spec.ad18xx.lo_as_master) { + /* rename 0x04 as "Master" and 0x02 as "Master Surround" */ + snd_ac97_rename_vol_ctl(ac97, "Master Playback", + "Master Surround Playback"); + snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", + "Master Playback"); + } return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls)); } @@ -2088,16 +2097,27 @@ static int patch_ad1888(struct snd_ac97 * ac97) patch_ad1881(ac97); ac97->build_ops = &patch_ad1888_build_ops; - /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */ - /* it seems that most vendors connect line-out connector to headphone out of AC'97 */ + + /* + * LO can be used as a real line-out on some devices, + * and we need to revert the front/surround mixer switches + */ + if (ac97->subsystem_vendor == 0x1043 && + ac97->subsystem_device == 0x1193) /* ASUS A9T laptop */ + ac97->spec.ad18xx.lo_as_master = 1; + + misc = snd_ac97_read(ac97, AC97_AD_MISC); /* AD-compatible mode */ /* Stereo mutes enabled */ - misc = snd_ac97_read(ac97, AC97_AD_MISC); - snd_ac97_write_cache(ac97, AC97_AD_MISC, misc | - AC97_AD198X_LOSEL | - AC97_AD198X_HPSEL | - AC97_AD198X_MSPLT | - AC97_AD198X_AC97NC); + misc |= AC97_AD198X_MSPLT | AC97_AD198X_AC97NC; + if (!ac97->spec.ad18xx.lo_as_master) + /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */ + /* it seems that most vendors connect line-out connector to + * headphone out of AC'97 + */ + misc |= AC97_AD198X_LOSEL | AC97_AD198X_HPSEL; + + snd_ac97_write_cache(ac97, AC97_AD_MISC, misc); ac97->flags |= AC97_STEREO_MUTES; return 0; } @@ -3446,6 +3466,7 @@ static const struct snd_kcontrol_new snd_ac97_controls_vt1617a[] = { int patch_vt1617a(struct snd_ac97 * ac97) { int err = 0; + int val; /* we choose to not fail out at this point, but we tell the caller when we return */ @@ -3456,7 +3477,13 @@ int patch_vt1617a(struct snd_ac97 * ac97) /* bring analog power consumption to normal by turning off the * headphone amplifier, like WinXP driver for EPIA SP */ - snd_ac97_write_cache(ac97, 0x5c, 0x20); + /* We need to check the bit before writing it. + * On some (many?) hardwares, setting bit actually clears it! + */ + val = snd_ac97_read(ac97, 0x5c); + if (!(val & 0x20)) + snd_ac97_write_cache(ac97, 0x5c, 0x20); + ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; ac97->build_ops = &patch_vt1616_ops; diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 56f87cd33c19..3f00ddf450f8 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -316,6 +316,8 @@ static int __devinit snd_aw2_create(struct snd_card *card, return -ENOMEM; } + /* (2) initialization of the chip hardware */ + snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt); if (request_irq(pci->irq, snd_aw2_saa7146_interrupt, IRQF_SHARED, "Audiowerk2", chip)) { @@ -329,8 +331,6 @@ static int __devinit snd_aw2_create(struct snd_card *card, } chip->irq = pci->irq; - /* (2) initialization of the chip hardware */ - snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt); err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { free_irq(chip->irq, (void *)chip); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index abde5b901884..548c9cc81af5 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1818,13 +1818,6 @@ int __devinit snd_emu10k1_create(struct snd_card *card, } emu->port = pci_resource_start(pci, 0); - if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED, - "EMU10K1", emu)) { - err = -EBUSY; - goto error; - } - emu->irq = pci->irq; - emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT; if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 32 * 1024, &emu->ptb_pages) < 0) { @@ -1887,6 +1880,14 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->fx8010.etram_pages.area = NULL; emu->fx8010.etram_pages.bytes = 0; + /* irq handler must be registered after I/O ports are activated */ + if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED, + "EMU10K1", emu)) { + err = -EBUSY; + goto error; + } + emu->irq = pci->irq; + /* * Init to 0x02109204 : * Clock accuracy = 0 (1000ppm) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e0a605adde42..a99e86d74278 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2858,6 +2858,7 @@ static const char *ad1988_models[AD1988_MODEL_LAST] = { static struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), {} }; @@ -3643,33 +3644,17 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { } /* end */ }; -static struct hda_input_mux ad1884a_mobile_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x1 }, /* port-C */ - { "Mix", 0x3 }, - }, -}; - static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, { } /* end */ }; @@ -3686,14 +3671,31 @@ static void ad1884a_hp_automute(struct hda_codec *codec) present ? 0x00 : 0x02); } +/* switch to external mic if plugged */ +static void ad1884a_hp_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, + present ? 0 : 1); +} + #define AD1884A_HP_EVENT 0x37 +#define AD1884A_MIC_EVENT 0x36 /* unsolicited event for HP jack sensing */ static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) != AD1884A_HP_EVENT) - return; - ad1884a_hp_automute(codec); + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1884a_hp_automic(codec); + break; + } } /* initialize jack-sensing, too */ @@ -3701,6 +3703,7 @@ static int ad1884a_hp_init(struct hda_codec *codec) { ad198x_init(codec); ad1884a_hp_automute(codec); + ad1884a_hp_automic(codec); return 0; } @@ -3714,10 +3717,15 @@ static struct hda_verb ad1884a_laptop_verbs[] = { /* Port-F pin */ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-C pin - internal mic-in */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ /* analog mix */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* unsolicited event for pin-sense */ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, { } /* end */ }; @@ -3877,7 +3885,6 @@ static int patch_ad1884a(struct hda_codec *codec) spec->mixers[0] = ad1884a_mobile_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs; spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1884a_mobile_capture_source; codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; codec->patch_ops.init = ad1884a_hp_init; break; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index c73ce074a6ea..6ef57fbfb6eb 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -611,6 +611,7 @@ static const char *cmi9880_models[CMI_MODELS] = { static struct snd_pci_quirk cmi9880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG), + SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL), SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG), {} /* terminator */ }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d9783a4263e0..b0a2a262ece2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -853,6 +853,7 @@ do_sku: case 0x10ec0269: case 0x10ec0862: case 0x10ec0662: + case 0x10ec0889: snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_EAPD_BTLENABLE, 2); snd_hda_codec_write(codec, 0x15, 0, @@ -877,6 +878,7 @@ do_sku: case 0x10ec0883: case 0x10ec0885: case 0x10ec0888: + case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); tmp = snd_hda_codec_read(codec, 0x20, 0, @@ -940,7 +942,6 @@ do_sku: AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); spec->unsol_event = alc_sku_unsol_event; - spec->init_hook = alc_sku_automute; } /* @@ -2981,7 +2982,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */ SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x814e, "ASUS", ALC880_ASUS), + SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), @@ -7743,6 +7744,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), @@ -8640,6 +8642,7 @@ static struct hda_verb alc262_sony_unsol_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} }; /* mute/unmute internal speaker according to the hp jack and mute state */ @@ -8757,35 +8760,39 @@ static struct hda_input_mux alc262_HP_D7000_capture_source = { }, }; -/* mute/unmute internal speaker according to the hp jack and mute state */ +/* mute/unmute internal speaker according to the hp jacks and mute state */ static void alc262_fujitsu_automute(struct hda_codec *codec, int force) { struct alc_spec *spec = codec->spec; unsigned int mute; if (force || !spec->sense_updated) { - unsigned int present_int_hp, present_dock_hp; + unsigned int present; /* need to execute and sync at first */ snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); - present_int_hp = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0); - snd_hda_codec_read(codec, 0x1B, 0, AC_VERB_SET_PIN_SENSE, 0); - present_dock_hp = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present_int_hp & 0x80000000) != 0; - spec->jack_present |= (present_dock_hp & 0x80000000) != 0; + /* check laptop HP jack */ + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0); + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + /* check docking HP jack */ + present |= snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + if (present & AC_PINSENSE_PRESENCE) + spec->jack_present = 1; + else + spec->jack_present = 0; spec->sense_updated = 1; } - if (spec->jack_present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ + /* unmute internal speaker only if both HPs are unplugged and + * master switch is on + */ + if (spec->jack_present) + mute = HDA_AMP_MUTE; + else mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); } /* unsolicited event for HP jack sensing */ @@ -8797,6 +8804,11 @@ static void alc262_fujitsu_unsol_event(struct hda_codec *codec, alc262_fujitsu_automute(codec, 1); } +static void alc262_fujitsu_init_hook(struct hda_codec *codec) +{ + alc262_fujitsu_automute(codec, 1); +} + /* bind volumes of both NID 0x0c and 0x0d */ static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = { .ops = &snd_hda_bind_vol, @@ -9570,6 +9582,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_fujitsu_capture_source, .unsol_event = alc262_fujitsu_unsol_event, + .init_hook = alc262_fujitsu_init_hook, }, [ALC262_HP_BPC] = { .mixers = { alc262_HP_BPC_mixer }, @@ -10500,6 +10513,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), @@ -11902,7 +11916,10 @@ static void alc861_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { - alc_set_pin_output(codec, nid, pin_type); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_type); + snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); } static void alc861_auto_init_multi_out(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b3a15d616873..a4f44a00bae8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -840,7 +840,7 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), { } /* end */ }; @@ -4289,6 +4289,8 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847635, .name = "STAC9250D", .patch = patch_stac925x }, { .id = 0x83847636, .name = "STAC9251", .patch = patch_stac925x }, { .id = 0x83847637, .name = "STAC9250D", .patch = patch_stac925x }, + { .id = 0x83847645, .name = "92HD206X", .patch = patch_stac927x }, + { .id = 0x83847646, .name = "92HD206D", .patch = patch_stac927x }, /* The following does not take into account .id=0x83847661 when subsys = * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are * currently not fully supported. diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 52b1d81a26f7..e7e43524f8c7 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -447,6 +447,23 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = { }, }; +static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .nid = 0x10, /* NID to query formats and rates */ + /* We got noisy outputs on the right channel on VT1708 when + * 24bit samples are used. Until any workaround is found, + * disable the 24bit format, so far. + */ + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_pcm_prepare, + .cleanup = via_playback_pcm_cleanup + }, +}; + static struct hda_pcm_stream vt1708_pcm_analog_capture = { .substreams = 2, .channels_min = 2, @@ -899,6 +916,9 @@ static int patch_vt1708(struct hda_codec *codec) spec->stream_name_analog = "VT1708 Analog"; spec->stream_analog_playback = &vt1708_pcm_analog_playback; + /* disable 32bit format on VT1708 */ + if (codec->vendor_id == 0x11061708) + spec->stream_analog_playback = &vt1708_pcm_analog_s16_playback; spec->stream_analog_capture = &vt1708_pcm_analog_capture; spec->stream_name_digital = "VT1708 Digital"; diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index cc0cddadd589..6facac5aed90 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -936,11 +936,13 @@ static int add_controls(struct oxygen *chip, for (i = 0; i < count; ++i) { template = controls[i]; - err = chip->model->control_filter(&template); - if (err < 0) - return err; - if (err == 1) - continue; + if (chip->model->control_filter) { + err = chip->model->control_filter(&template); + if (err < 0) + return err; + if (err == 1) + continue; + } if (!strcmp(template.name, "Master Playback Volume") && chip->model->dac_tlv) { template.tlv.p = chip->model->dac_tlv; diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index 67c88e322fb1..ccac6bd2889c 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -103,7 +103,8 @@ static void at91_pcm_dma_irq(u32 ssc_sr, if (prtd->period_ptr >= prtd->dma_buffer_end) { prtd->period_ptr = prtd->dma_buffer; } - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr); + at91_ssc_write(params->ssc_base + params->pdc->xnpr, + prtd->period_ptr); at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->period_size / params->pdc_xfer_size); } @@ -191,10 +192,12 @@ static int at91_pcm_trigger(struct snd_pcm_substream *substream, at91_ssc_write(params->ssc_base + AT91_SSC_IER, params->mask->ssc_endx | params->mask->ssc_endbuf); - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); + at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, + params->mask->pdc_enable); - DBG("sr=%lx imr=%lx\n", at91_ssc_read(params->ssc_base + AT91_SSC_SR), - at91_ssc_read(params->ssc_base + AT91_SSC_IER)); + DBG("sr=%lx imr=%lx\n", + at91_ssc_read(params->ssc_base + AT91_SSC_SR), + at91_ssc_read(params->ssc_base + AT91_SSC_IMR)); break; case SNDRV_PCM_TRIGGER_STOP: diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index f642d2dd4ec3..bc35d00a38f8 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -590,7 +590,7 @@ static int at91_ssc_hw_params(struct snd_pcm_substream *substream, printk(KERN_WARNING "at91-ssc: request_irq failure\n"); DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid); + at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); return ret; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 630684f4a0bc..09b1661b8a3a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -539,8 +539,8 @@ static const char *intercon[][3] = { {"HPRCOM", NULL, "Right HP Com"}, /* Mono Output */ - {"MONOLOUT", NULL, "Mono Out"}, - {"MONOLOUT", NULL, "Mono Out"}, + {"MONO_LOUT", NULL, "Mono Out"}, + {"MONO_LOUT", NULL, "Mono Out"}, /* Left Input */ {"Left Line1L Mux", "single-ended", "LINE1L"}, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b2a11b0d2e4c..f588545698f3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -416,7 +416,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) * to put data into its FIFO. Without it, ALSA starts * to complain about overruns. */ - msleep(1); + mdelay(1); } break; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 83b1eb4e40f3..6533563a6011 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -188,8 +188,8 @@ static const char *audio_map[][3] = { static const char *spk_function[] = {"Off", "On"}; static const char *jack_function[] = {"Off", "Headphone"}; static const struct soc_enum n810_enum[] = { - SOC_ENUM_SINGLE_EXT(2, spk_function), - SOC_ENUM_SINGLE_EXT(3, jack_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), }; static const struct snd_kcontrol_new aic33_n810_controls[] = { diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 4ebcd6a8bf28..1ed6afd45459 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -224,6 +224,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, iismod |= S3C2410_IISMOD_SLAVE; break; case SND_SOC_DAIFMT_CBS_CFS: + iismod &= ~S3C2410_IISMOD_SLAVE; break; default: return -EINVAL; @@ -234,6 +235,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, iismod |= S3C2410_IISMOD_MSB; break; case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2410_IISMOD_MSB; break; default: return -EINVAL; diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 6c70a81c730c..7806ae614617 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -171,7 +171,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, ret = s3c2410_dma_request(prtd->params->channel, prtd->params->client, NULL); - if (ret) { + if (ret < 0) { DBG(KERN_ERR "failed to get dma channel\n"); return ret; } diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 478369bb38c3..b343818dbb96 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -341,8 +341,12 @@ snd_emux_control(void *p, int type, struct snd_midi_channel *chan) case MIDI_CTL_SOFT_PEDAL: #ifdef SNDRV_EMUX_USE_RAW_EFFECT /* FIXME: this is an emulation */ - snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, -160, + if (chan->control[type] >= 64) + snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, -160, EMUX_FX_FLAG_ADD); + else + snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, 0, + EMUX_FX_FLAG_OFF); #endif break; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index e97d8b2ac16a..a972f77bd785 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -351,8 +351,8 @@ static struct snd_card* create_card(struct usb_device* usb_dev) dev = caiaqdev(card); dev->chip.dev = usb_dev; dev->chip.card = card; - dev->chip.usb_id = USB_ID(usb_dev->descriptor.idVendor, - usb_dev->descriptor.idProduct); + dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor), + le16_to_cpu(usb_dev->descriptor.idProduct)); spin_lock_init(&dev->spinlock); snd_card_set_dev(card, &usb_dev->dev); |