diff options
Diffstat (limited to 'sound')
38 files changed, 336 insertions, 158 deletions
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 108b643229ba..6205f37d547c 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -75,7 +75,7 @@ int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - if (rtd && rtd->params) + if (rtd && rtd->params && rtd->params->drcmr) *rtd->params->drcmr = 0; snd_pcm_set_runtime_buffer(substream, NULL); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 333e4dd29450..72cfd47af6b8 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -233,6 +233,18 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } + if (xrun_debug(substream, 8)) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, " + "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + name, (unsigned int)pos, + (unsigned int)runtime->period_size, + (unsigned int)runtime->buffer_size, + (unsigned long)old_hw_ptr, + (unsigned long)runtime->hw_ptr_base, + (unsigned long)runtime->hw_ptr_interrupt); + } hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; @@ -244,18 +256,27 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = new_hw_ptr - hw_ptr_interrupt; } if (delta < 0) { - delta += runtime->buffer_size; + if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) + delta += runtime->buffer_size; if (delta < 0) { hw_ptr_error(substream, "Unexpected hw_pointer value " "(stream=%i, pos=%ld, intr_ptr=%ld)\n", substream->stream, (long)pos, (long)hw_ptr_interrupt); +#if 1 + /* simply skipping the hwptr update seems more + * robust in some cases, e.g. on VMware with + * inaccurate timer source + */ + return 0; /* skip this update */ +#else /* rebase to interrupt position */ hw_base = new_hw_ptr = hw_ptr_interrupt; /* align hw_base to buffer_size */ hw_base -= hw_base % runtime->buffer_size; delta = 0; +#endif } else { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) @@ -344,6 +365,19 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } + if (xrun_debug(substream, 16)) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, " + "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + name, (unsigned int)pos, + (unsigned int)runtime->period_size, + (unsigned int)runtime->buffer_size, + (unsigned long)old_hw_ptr, + (unsigned long)runtime->hw_ptr_base, + (unsigned long)runtime->hw_ptr_interrupt); + } + hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index 1bcb360330e5..941f64a853eb 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -3,10 +3,6 @@ # Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz> # -ifeq ($(CONFIG_SND_SEQUENCER_OSS),y) - obj-$(CONFIG_SND_SEQUENCER) += oss/ -endif - snd-seq-device-objs := seq_device.o snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \ seq_fifo.o seq_prioq.o seq_timer.o \ @@ -19,7 +15,8 @@ snd-seq-virmidi-objs := seq_virmidi.o obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o ifeq ($(CONFIG_SND_SEQUENCER_OSS),y) -obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o + obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o + obj-$(CONFIG_SND_SEQUENCER) += oss/ endif obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index edb11eefdfe3..2dcf45bf7293 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -795,13 +795,13 @@ static int snd_gf1_pcm_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ if (!(pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE)) continue; /* load real volume - better precision */ - spin_lock_irqsave(&gus->reg_lock, flags); + spin_lock(&gus->reg_lock); snd_gf1_select_voice(gus, pvoice->number); snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL); vol = pvoice == pcmp->pvoices[0] ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right; snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, vol); pcmp->final_volume = 1; - spin_unlock_irqrestore(&gus->reg_lock, flags); + spin_unlock(&gus->reg_lock); } spin_unlock_irqrestore(&gus->voice_alloc, flags); return change; diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index f24bf1ecb36d..15e4138bce17 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -325,9 +325,9 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = { .rate_max = 192000, .channels_min = 2, .channels_max = 2, - .buffer_bytes_max = ((65536 - 64) * 8), + .buffer_bytes_max = 65536 - 128, .period_bytes_min = 64, - .period_bytes_max = (65536 - 64), + .period_bytes_max = 32768 - 64, .periods_min = 2, .periods_max = 2, .fifo_size = 0, diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index a1db51b3ead8..a7f4a671f7b7 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -242,13 +242,12 @@ static int get_amixer_rsc(struct amixer_mgr *mgr, /* Allocate mem for amixer resource */ amixer = kzalloc(sizeof(*amixer), GFP_KERNEL); - if (NULL == amixer) { - err = -ENOMEM; - return err; - } + if (!amixer) + return -ENOMEM; /* Check whether there are sufficient * amixer resources to meet request. */ + err = 0; spin_lock_irqsave(&mgr->mgr_lock, flags); for (i = 0; i < desc->msr; i++) { err = mgr_get_resource(&mgr->mgr, 1, &idx); @@ -397,12 +396,11 @@ static int get_sum_rsc(struct sum_mgr *mgr, /* Allocate mem for sum resource */ sum = kzalloc(sizeof(*sum), GFP_KERNEL); - if (NULL == sum) { - err = -ENOMEM; - return err; - } + if (!sum) + return -ENOMEM; /* Check whether there are sufficient sum resources to meet request. */ + err = 0; spin_lock_irqsave(&mgr->mgr_lock, flags); for (i = 0; i < desc->msr; i++) { err = mgr_get_resource(&mgr->mgr, 1, &idx); diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 082e35c08c02..deb6cfa73600 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -57,9 +57,9 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { [LINEO1] = {.left = 0x40, .right = 0x41}, - [LINEO2] = {.left = 0x70, .right = 0x71}, + [LINEO2] = {.left = 0x60, .right = 0x61}, [LINEO3] = {.left = 0x50, .right = 0x51}, - [LINEO4] = {.left = 0x60, .right = 0x61}, + [LINEO4] = {.left = 0x70, .right = 0x71}, [LINEIM] = {.left = 0x45, .right = 0xc5}, [SPDIFOO] = {.left = 0x00, .right = 0x01}, [SPDIFIO] = {.left = 0x05, .right = 0x85}, diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index e1c145d8b702..df43a5cd3938 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -724,12 +724,11 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr, /* Allocate mem for SRCIMP resource */ srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL); - if (NULL == srcimp) { - err = -ENOMEM; - return err; - } + if (!srcimp) + return -ENOMEM; /* Check whether there are sufficient SRCIMP resources. */ + err = 0; spin_lock_irqsave(&mgr->mgr_lock, flags); for (i = 0; i < desc->msr; i++) { err = mgr_get_resource(&mgr->mgr, 1, &idx); diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index e617acaf10e3..61b8ab39800f 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -644,7 +644,7 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * int err; int capture=1; - /* snd_printk("KERN_DEBUG snd_p16v_pcm called. device=%d\n", device); */ + /* snd_printk(KERN_DEBUG "snd_p16v_pcm called. device=%d\n", device); */ emu->p16v_device_offset = device; if (rpcm) *rpcm = NULL; diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 29272f2e95a0..b0275a050870 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -50,19 +50,22 @@ static void snd_hda_generate_beep(struct work_struct *work) * The tone frequency of beep generator on IDT/STAC codecs is * defined from the 8bit tone parameter, in Hz, * freq = 48000 * (257 - tone) / 1024 - * that is from 12kHz to 93.75kHz in step of 46.875 hz + * that is from 12kHz to 93.75Hz in steps of 46.875 Hz */ static int beep_linear_tone(struct hda_beep *beep, int hz) { + if (hz <= 0) + return 0; hz *= 1000; /* fixed point */ - hz = hz - DIGBEEP_HZ_MIN; + hz = hz - DIGBEEP_HZ_MIN + + DIGBEEP_HZ_STEP / 2; /* round to nearest step */ if (hz < 0) hz = 0; /* turn off PC beep*/ else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) - hz = 0xff; + hz = 1; /* max frequency */ else { hz /= DIGBEEP_HZ_STEP; - hz++; + hz = 255 - hz; } return hz; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 462e2cedaa6a..88480c0c58a0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -332,6 +332,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, AC_VERB_GET_CONNECT_LIST, i); range_val = !!(parm & (1 << (shift-1))); /* ranges */ val = parm & mask; + if (val == 0) { + snd_printk(KERN_WARNING "hda_codec: " + "invalid CONNECT_LIST verb %x[%i]:%x\n", + nid, i, parm); + return 0; + } parm >>= shift; if (range_val) { /* ranges between the previous and this one */ @@ -3470,10 +3476,16 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, } mutex_lock(&codec->spdif_mutex); if (mout->share_spdif) { - runtime->hw.rates &= mout->spdif_rates; - runtime->hw.formats &= mout->spdif_formats; - if (mout->spdif_maxbps < hinfo->maxbps) - hinfo->maxbps = mout->spdif_maxbps; + if ((runtime->hw.rates & mout->spdif_rates) && + (runtime->hw.formats & mout->spdif_formats)) { + runtime->hw.rates &= mout->spdif_rates; + runtime->hw.formats &= mout->spdif_formats; + if (mout->spdif_maxbps < hinfo->maxbps) + hinfo->maxbps = mout->spdif_maxbps; + } else { + mout->share_spdif = 0; + /* FIXME: need notify? */ + } } mutex_unlock(&codec->spdif_mutex); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4e9ea7080270..77c1b840ca8b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1454,6 +1454,18 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&chip->open_mutex); return err; } + snd_pcm_limit_hw_rates(runtime); + /* sanity check */ + if (snd_BUG_ON(!runtime->hw.channels_min) || + snd_BUG_ON(!runtime->hw.channels_max) || + snd_BUG_ON(!runtime->hw.formats) || + snd_BUG_ON(!runtime->hw.rates)) { + azx_release_device(azx_dev); + hinfo->ops.close(hinfo, apcm->codec, substream); + snd_hda_power_down(apcm->codec); + mutex_unlock(&chip->open_mutex); + return -EINVAL; + } spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = substream; azx_dev->running = 0; @@ -1462,7 +1474,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->private_data = azx_dev; snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); - return 0; } @@ -2322,9 +2333,19 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap = azx_readw(chip, GCAP); snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); - /* ATI chips seems buggy about 64bit DMA addresses */ - if (chip->driver_type == AZX_DRIVER_ATI) - gcap &= ~ICH6_GCAP_64OK; + /* disable SB600 64bit support for safety */ + if ((chip->driver_type == AZX_DRIVER_ATI) || + (chip->driver_type == AZX_DRIVER_ATIHDMI)) { + struct pci_dev *p_smbus; + p_smbus = pci_get_device(PCI_VENDOR_ID_ATI, + PCI_DEVICE_ID_ATI_SBX00_SMBUS, + NULL); + if (p_smbus) { + if (p_smbus->revision < 0x30) + gcap &= ~ICH6_GCAP_64OK; + pci_dev_put(p_smbus); + } + } /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1988582d1ab8..3da85caf8af1 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3746,9 +3746,30 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { } /* end */ }; +static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + int mute = (!ucontrol->value.integer.value[0] && + !ucontrol->value.integer.value[1]); + /* toggle GPIO1 according to the mute state */ + snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + mute ? 0x02 : 0x0); + return ret; +} + 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_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, 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 Capture Volume", 0x14, 0x0, HDA_INPUT), @@ -3869,6 +3890,10 @@ static struct hda_verb ad1884a_mobile_verbs[] = { /* 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}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ { } /* end */ }; @@ -3978,6 +4003,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 392d108c3558..019ca7cb56d7 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -510,7 +510,7 @@ static int ca0110_parse_auto_config(struct hda_codec *codec) } -int patch_ca0110(struct hda_codec *codec) +static int patch_ca0110(struct hda_codec *codec) { struct ca0110_spec *spec; int err; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a8e58c483df..8c8b273116fb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4505,6 +4505,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec) &dig_nid, 1); if (err < 0) continue; + if (dig_nid > 0x7f) { + printk(KERN_ERR "alc880_auto: invalid dig_nid " + "connection 0x%x for NID 0x%x\n", dig_nid, + spec->autocfg.dig_out_pins[i]); + continue; + } if (!i) spec->multiout.dig_out_nid = dig_nid; else { @@ -6919,9 +6925,6 @@ static struct hda_verb alc882_targa_verbs[] = { {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, { } /* end */ }; @@ -7241,7 +7244,8 @@ static struct alc_config_preset alc882_presets[] = { }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_targa_verbs}, + .init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs, + alc882_targa_verbs}, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, @@ -9238,7 +9242,8 @@ static struct alc_config_preset alc883_presets[] = { }, [ALC883_TARGA_DIG] = { .mixers = { alc883_targa_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, @@ -9251,7 +9256,8 @@ static struct alc_config_preset alc883_presets[] = { }, [ALC883_TARGA_2ch_DIG] = { .mixers = { alc883_targa_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, @@ -10625,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec, alc262_lenovo_3000_automute(codec, 1); } +static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid, + int dir, int idx, long *valp) +{ + int i, change = 0; + + for (i = 0; i < 2; i++, valp++) + change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE); + return change; +} + /* bind hp and internal speaker mute (with plug check) */ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -10633,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp ? 0 : HDA_AMP_MUTE); - + change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp); + change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp); if (change) alc262_fujitsu_automute(codec, 0); return change; @@ -10674,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp ? 0 : HDA_AMP_MUTE); - + change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp); if (change) alc262_lenovo_3000_automute(codec, 0); return change; @@ -11848,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[0] ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[1] ? 0 : HDA_AMP_MUTE); + change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp); if (change) alc268_acer_automute(codec, 0); return change; @@ -12876,20 +12881,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { } }; -/* 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), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -12902,12 +12898,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { }; /* FSC amilo */ -static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol), - { } /* end */ -}; +#define alc269_fujitsu_mixer alc269_eeepc_mixer static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 14f3c3e0f62d..512f3b9b9a45 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,8 +1590,6 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, - "SigmaTel",STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ @@ -2344,6 +2342,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, + "SigmaTel", STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_9205_REF), /* Dell */ @@ -2378,6 +2378,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, "Dell Vostro 1500", STAC_9205_DELL_M42), /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD), SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), {} /* terminator */ }; @@ -4065,7 +4066,7 @@ static int stac92xx_add_jack(struct hda_codec *codec, jack->nid = nid; jack->type = type; - sprintf(name, "%s at %s %s Jack", + snprintf(name, sizeof(name), "%s at %s %s Jack", snd_hda_get_jack_type(def_conf), snd_hda_get_jack_connectivity(def_conf), snd_hda_get_jack_location(def_conf)); @@ -5854,6 +5855,8 @@ static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = { }; static struct snd_pci_quirk stac9872_cfg_tbl[] = { + SND_PCI_QUIRK_MASK(0x104d, 0xfff0, 0x81e0, + "Sony VAIO F/S", STAC_9872_VAIO), {} /* terminator */ }; @@ -5866,6 +5869,8 @@ static int patch_stac9872(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; codec->spec = spec; + spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); + spec->pin_nids = stac9872_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, stac9872_models, @@ -5877,8 +5882,6 @@ static int patch_stac9872(struct hda_codec *codec) stac92xx_set_config_regs(codec, stac9872_brd_tbl[spec->board_config]); - spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); - spec->pin_nids = stac9872_pin_nids; spec->multiout.dac_nids = spec->dac_nids; spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); spec->adc_nids = stac9872_adc_nids; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 8e004fb6961a..9008b4b013aa 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -210,7 +210,9 @@ struct via_spec { /* capture */ unsigned int num_adc_nids; hda_nid_t *adc_nids; + hda_nid_t mux_nids[3]; hda_nid_t dig_in_nid; + hda_nid_t dig_in_pin; /* capture source */ const struct hda_input_mux *input_mux; @@ -319,6 +321,9 @@ static void via_auto_set_output_and_unmute(struct hda_codec *codec, pin_type); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x02); } @@ -387,27 +392,12 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - unsigned int vendor_id = codec->vendor_id; - - /* AIW0 lydia 060801 add for correct sw0 input select */ - if (IS_VT1708_VENDORID(vendor_id) && (adc_idx == 0)) - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - 0x18, &spec->cur_mux[adc_idx]); - else if ((IS_VT1709_10CH_VENDORID(vendor_id) || - IS_VT1709_6CH_VENDORID(vendor_id)) && (adc_idx == 0)) - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - 0x19, &spec->cur_mux[adc_idx]); - else if ((IS_VT1708B_8CH_VENDORID(vendor_id) || - IS_VT1708B_4CH_VENDORID(vendor_id)) && (adc_idx == 0)) - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - 0x17, &spec->cur_mux[adc_idx]); - else if (IS_VT1702_VENDORID(vendor_id) && (adc_idx == 0)) - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - 0x13, &spec->cur_mux[adc_idx]); - else - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->adc_nids[adc_idx], - &spec->cur_mux[adc_idx]); + + if (!spec->mux_nids[adc_idx]) + return -EINVAL; + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + spec->mux_nids[adc_idx], + &spec->cur_mux[adc_idx]); } static int via_independent_hp_info(struct snd_kcontrol *kcontrol, @@ -998,25 +988,11 @@ static int via_init(struct hda_codec *codec) /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ - if (IS_VT1708_VENDORID(codec->vendor_id)) { - snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_OUT); - snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, - AC_VERB_SET_EAPD_BTLENABLE, 0x02); - } else if (IS_VT1709_10CH_VENDORID(codec->vendor_id) || - IS_VT1709_6CH_VENDORID(codec->vendor_id)) { - snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, + if (spec->dig_in_pin) { + snd_hda_codec_write(codec, spec->dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, - AC_VERB_SET_EAPD_BTLENABLE, 0x02); - } else if (IS_VT1708B_8CH_VENDORID(codec->vendor_id) || - IS_VT1708B_4CH_VENDORID(codec->vendor_id)) { - snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_OUT); - snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, + snd_hda_codec_write(codec, spec->dig_in_pin, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } } else /* enable SPDIF-input pin */ @@ -1326,6 +1302,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708_DIGOUT_NID; + spec->dig_in_pin = VT1708_DIGIN_PIN; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708_DIGIN_NID; @@ -1352,6 +1329,34 @@ static int via_auto_init(struct hda_codec *codec) return 0; } +static int get_mux_nids(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + hda_nid_t nid, conn[8]; + unsigned int type; + int i, n; + + for (i = 0; i < spec->num_adc_nids; i++) { + nid = spec->adc_nids[i]; + while (nid) { + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; + if (type == AC_WID_PIN) + break; + n = snd_hda_get_connections(codec, nid, conn, + ARRAY_SIZE(conn)); + if (n <= 0) + break; + if (n > 1) { + spec->mux_nids[i] = nid; + break; + } + nid = conn[0]; + } + } + return 0; +} + static int patch_vt1708(struct hda_codec *codec) { struct via_spec *spec; @@ -1799,6 +1804,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1709_DIGOUT_NID; + spec->dig_in_pin = VT1709_DIGIN_PIN; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1709_DIGIN_NID; @@ -1859,6 +1865,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1709_capture_mixer; spec->num_mixers++; } @@ -1952,6 +1959,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1709_capture_mixer; spec->num_mixers++; } @@ -2344,6 +2352,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID; + spec->dig_in_pin = VT1708B_DIGIN_PIN; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708B_DIGIN_NID; @@ -2404,6 +2413,7 @@ static int patch_vt1708B_8ch(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708B_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1708B_capture_mixer; spec->num_mixers++; } @@ -2455,6 +2465,7 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708B_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1708B_capture_mixer; spec->num_mixers++; } @@ -2889,6 +2900,7 @@ static int patch_vt1708S(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1708S_capture_mixer; spec->num_mixers++; } @@ -3206,6 +3218,7 @@ static int patch_vt1702(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1702_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1702_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1702_capture_mixer; spec->num_mixers++; } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index bf971f7cfdc6..6ebcb6bdd712 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -635,6 +635,8 @@ static void xonar_d2_resume(struct oxygen *chip) static void xonar_d1_resume(struct oxygen *chip) { + oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); + msleep(1); cs43xx_init(chip); xonar_enable_output(chip); } diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 235a71e5ac8d..b5ca02e2038c 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -2197,9 +2197,12 @@ static int __init alsa_card_riptide_init(void) if (err < 0) return err; #if defined(SUPPORT_JOYSTICK) - pci_register_driver(&joystick_driver); + err = pci_register_driver(&joystick_driver); + /* On failure unregister formerly registered audio driver */ + if (err < 0) + pci_unregister_driver(&driver); #endif - return 0; + return err; } static void __exit alsa_card_riptide_exit(void) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ab099f482487..cb0d1bf34b57 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; u16 pll_d = 1; + u8 reg; /* select data word length */ data = @@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, pll_q &= 0xf; aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); - } else + /* disable PLL if it is bypassed */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); + + } else { aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + /* enable PLL when it is used */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); + } /* Route Left DAC to left channel input and * right DAC to right channel input */ diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d28eeaceb857..49c4b2898aff 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -79,7 +79,7 @@ static const u16 wm8753_reg[] = { 0x0097, 0x0097, 0x0000, 0x0004, 0x0000, 0x0083, 0x0024, 0x01ba, 0x0000, 0x0083, 0x0024, 0x01ba, - 0x0000, 0x0000 + 0x0000, 0x0000, 0x0000 }; /* codec private data */ @@ -1660,11 +1660,11 @@ static int wm8753_register(struct wm8753_priv *wm8753) codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; - codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache); + codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache) + 1; codec->reg_cache = &wm8753->reg_cache; codec->private_data = wm8753; - memcpy(codec->reg_cache, wm8753_reg, sizeof(codec->reg_cache)); + memcpy(codec->reg_cache, wm8753_reg, sizeof(wm8753->reg_cache)); INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); ret = wm8753_reset(codec); diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index c05f71803aa8..8c0fdf84aac3 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -1037,14 +1037,14 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi) codec->control_data = spi; codec->dev = &spi->dev; - spi->dev.driver_data = wm8988; + dev_set_drvdata(&spi->dev, wm8988); return wm8988_register(wm8988); } static int __devexit wm8988_spi_remove(struct spi_device *spi) { - struct wm8988_priv *wm8988 = spi->dev.driver_data; + struct wm8988_priv *wm8988 = dev_get_drvdata(&spi->dev); wm8988_unregister(wm8988); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 5dbebf82249c..8cb65ccad35f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -33,7 +33,7 @@ config SND_SOC_MPC5200_I2S config SND_SOC_MPC5200_AC97 tristate "Freescale MPC5200 PSC in AC97 mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM - select AC97_BUS + select SND_SOC_AC97_BUS select SND_MPC52xx_DMA select PPC_BESTCOMM_GEN_BD help @@ -41,7 +41,7 @@ config SND_SOC_MPC5200_AC97 config SND_MPC52xx_SOC_PCM030 tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" - depends on PPC_MPC5200_SIMPLE && BROKEN + depends on PPC_MPC5200_SIMPLE select SND_SOC_MPC5200_AC97 select SND_SOC_WM9712 help @@ -50,7 +50,7 @@ config SND_MPC52xx_SOC_PCM030 config SND_MPC52xx_SOC_EFIKA tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" - depends on PPC_EFIKA && BROKEN + depends on PPC_EFIKA select SND_SOC_MPC5200_AC97 select SND_SOC_STAC9766 help diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index efec33a1c5bd..f0a2d4071998 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -456,6 +456,7 @@ int mpc5200_audio_dma_create(struct of_device *op) return -ENODEV; spin_lock_init(&psc_dma->lock); + mutex_init(&psc_dma->mutex); psc_dma->id = be32_to_cpu(*prop); psc_dma->irq = irq; psc_dma->psc_regs = regs; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 2000803f06a7..8d396bb9d9fe 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -55,6 +55,7 @@ struct psc_dma { unsigned int irq; struct device *dev; spinlock_t lock; + struct mutex mutex; u32 sicr; uint sysclk; int imr; diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 794a247b3eb5..7eb549985d49 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -34,13 +34,20 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) int status; unsigned int val; + mutex_lock(&psc_dma->mutex); + /* Wait for command send status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } + + /* Force clear the data valid bit */ + in_be32(&psc_dma->psc_regs->ac97_data); + /* Send the read */ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); @@ -50,16 +57,19 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } /* Get the data */ val = in_be32(&psc_dma->psc_regs->ac97_data); if (((val >> 24) & 0x7f) != reg) { pr_err("reg echo error on ac97 read\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } val = (val >> 8) & 0xffff; + mutex_unlock(&psc_dma->mutex); return (unsigned short) val; } @@ -68,16 +78,21 @@ static void psc_ac97_write(struct snd_ac97 *ac97, { int status; + mutex_lock(&psc_dma->mutex); + /* Wait for command status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); - return; + goto out; } /* Write data */ out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + out: + mutex_unlock(&psc_dma->mutex); } static void psc_ac97_warm_reset(struct snd_ac97 *ac97) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 6454e15f7d28..84a1950880eb 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -216,12 +216,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ptr = omap_get_dma_src_pos(prtd->dma_ch); - else + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else if (!(cpu_is_omap1510())) { + ptr = omap_get_dma_src_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else + offset = prtd->period_index * runtime->period_size; - offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); if (offset >= runtime->buffer_size) offset = 0; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 4743e262895d..6b8f655d1ad8 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -167,6 +167,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); + dai->private_data = dai; pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -255,7 +256,10 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); - clk_disable(clk_i2s); + if (dai->private_data != NULL) { + clk_disable(clk_i2s); + dai->private_data = NULL; + } } } @@ -336,6 +340,7 @@ static int pxa2xx_i2s_probe(struct platform_device *dev) return PTR_ERR(clk_i2s); pxa_i2s_dai.dev = &dev->dev; + pxa_i2s_dai.private_data = NULL; ret = snd_soc_register_dai(&pxa_i2s_dai); if (ret != 0) clk_put(clk_i2s); diff --git a/sound/sound_core.c b/sound/sound_core.c index 12522e6913d9..a41f8b127f49 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -10,6 +10,8 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/err.h> +#include <linux/kdev_t.h> +#include <linux/major.h> #include <sound/core.h> #ifdef CONFIG_SOUND_OSS_CORE @@ -29,6 +31,8 @@ MODULE_LICENSE("GPL"); static char *sound_nodename(struct device *dev) { + if (MAJOR(dev->devt) == SOUND_MAJOR) + return NULL; return kasprintf(GFP_KERNEL, "snd/%s", dev_name(dev)); } @@ -104,7 +108,6 @@ module_exit(cleanup_soundcore); #include <linux/types.h> #include <linux/kernel.h> #include <linux/sound.h> -#include <linux/major.h> #include <linux/kmod.h> #define SOUND_STEP 16 diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 523aec188ccf..73525c048e7f 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -48,6 +48,7 @@ config SND_USB_CAIAQ * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 2 DJ * Native Instruments Audio 4 DJ * Native Instruments Audio 8 DJ * Native Instruments Guitar Rig Session I/O diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 8f9b60c5d74c..121af0644fd9 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE): dev->samplerates |= SNDRV_PCM_RATE_192000; /* fall thru */ + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): dev->samplerates |= SNDRV_PCM_RATE_88200; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 0e5db719de24..83e6c1312d47 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,13 +35,14 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 2 DJ}," "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," "{Native Instruments, Session I/O}," @@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_AUDIO4DJ }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_AUDIO2DJ + }, { /* terminator */ } }; @@ -349,7 +355,9 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) log("Unable to set up control system (ret=%d)\n", ret); } -static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) +static int create_card(struct usb_device *usb_dev, + struct usb_interface *intf, + struct snd_card **cardp) { int devnum; int err; @@ -374,7 +382,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) 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); + snd_card_set_dev(card, &intf->dev); *cardp = card; return 0; @@ -461,7 +469,7 @@ static int __devinit snd_probe(struct usb_interface *intf, struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - ret = create_card(device, &card); + ret = create_card(device, intf, &card); if (ret < 0) return ret; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index ece73514854e..44e3edf88bef 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -10,6 +10,7 @@ #define USB_PID_KORECONTROLLER 0x4711 #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO2DJ 0x041c #define USB_PID_AUDIO4DJ 0x0839 #define USB_PID_AUDIO8DJ 0x1978 #define USB_PID_SESSIONIO 0x1915 diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c7b902358b7b..44b9cdc8a83b 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2661,7 +2661,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) struct usb_interface_descriptor *altsd; int i, altno, err, stream; int format; - struct audioformat *fp; + struct audioformat *fp = NULL; unsigned char *fmt, *csep; int num; @@ -2734,6 +2734,18 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; } + /* + * Blue Microphones workaround: The last altsetting is identical + * with the previous one, except for a larger packet size, but + * is actually a mislabeled two-channel setting; ignore it. + */ + if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && + fp && fp->altsetting == 1 && fp->channels == 1 && + fp->format == SNDRV_PCM_FORMAT_S16_LE && + le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == + fp->maxpacksize * 2) + continue; + csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT); /* Creamware Noah has this descriptor after the 2nd endpoint */ if (!csep && altsd->bNumEndpoints >= 2) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 4bd3a7a0edc1..ec9cdf986928 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -990,20 +990,35 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, break; } - /* quirk for UDA1321/N101 */ - /* note that detection between firmware 2.1.1.7 (N101) and later 2.1.1.21 */ - /* is not very clear from datasheets */ - /* I hope that the min value is -15360 for newer firmware --jk */ + /* volume control quirks */ switch (state->chip->usb_id) { case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): case USB_ID(0x0672, 0x1041): + /* quirk for UDA1321/N101. + * note that detection between firmware 2.1.1.7 (N101) + * and later 2.1.1.21 is not very clear from datasheets. + * I hope that the min value is -15360 for newer firmware --jk + */ if (!strcmp(kctl->id.name, "PCM Playback Volume") && cval->min == -15616) { - snd_printk(KERN_INFO "using volume control quirk for the UDA1321/N101 chip\n"); + snd_printk(KERN_INFO + "set volume quirk for UDA1321/N101 chip\n"); cval->max = -256; } + break; + + case USB_ID(0x046d, 0x09a4): + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO + "set volume quirk for QuickCam E3500\n"); + cval->min = 6080; + cval->max = 8768; + cval->res = 192; + } + break; + } snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index a5aae9d67f31..fd44946ce4b3 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -514,7 +514,6 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) US122L(card)->chip.dev->bus->busnum, US122L(card)->chip.dev->devnum ); - snd_card_set_dev(card, &device->dev); *cardp = card; return 0; } @@ -531,6 +530,7 @@ static int us122l_usb_probe(struct usb_interface *intf, if (err < 0) return err; + snd_card_set_dev(card, &intf->dev); if (!us122l_create_card(card)) { snd_card_free(card); return -EINVAL; diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 5ce0da23ee96..cb4bb8373ca2 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -364,7 +364,6 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) 0,//us428(card)->usbmidi.ifnum, usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum ); - snd_card_set_dev(card, &device->dev); *cardp = card; return 0; } @@ -388,6 +387,7 @@ static int usX2Y_usb_probe(struct usb_device *device, err = usX2Y_create_card(device, &card); if (err < 0) return err; + snd_card_set_dev(card, &intf->dev); if ((err = usX2Y_hwdep_new(card, device)) < 0 || (err = snd_card_register(card)) < 0) { snd_card_free(card); diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index dd1ab6177840..9efd27f6b52f 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -296,9 +296,10 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, struct snd_usX2Y_substream *subs, struct urb *urb) { - snd_printk(KERN_ERR "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" - KERN_ERR "Most propably some urb of usb-frame %i is still missing.\n" - KERN_ERR "Cause could be too long delays in usb-hcd interrupt handling.\n", + snd_printk(KERN_ERR +"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" +"Most propably some urb of usb-frame %i is still missing.\n" +"Cause could be too long delays in usb-hcd interrupt handling.\n", usb_get_current_frame_number(usX2Y->chip.dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); |