diff options
Diffstat (limited to 'sound')
-rw-r--r-- | sound/arm/aaci.c | 6 | ||||
-rw-r--r-- | sound/core/oss/pcm_oss.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 9 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.h | 1 | ||||
-rw-r--r-- | sound/pci/hda/hda_proc.c | 3 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 14 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 6 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 4 | ||||
-rw-r--r-- | sound/pci/intel8x0.c | 2 | ||||
-rw-r--r-- | sound/soc/atmel/atmel_ssc_dai.c | 2 | ||||
-rw-r--r-- | sound/soc/atmel/atmel_ssc_dai.h | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8350.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8753.c | 9 | ||||
-rw-r--r-- | sound/soc/codecs/wm8990.c | 3 | ||||
-rw-r--r-- | sound/soc/omap/omap-mcbsp.c | 4 | ||||
-rw-r--r-- | sound/soc/omap/omap-pcm.c | 5 | ||||
-rw-r--r-- | sound/usb/usbaudio.c | 1 |
17 files changed, 55 insertions, 20 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 89096e811a4b..772901e41ecb 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--); + } while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout); if (!timeout) dev_err(&aaci->dev->dev, @@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & SLFR_1TXB) && timeout--); + } while ((v & SLFR_1TXB) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n"); @@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) do { cond_resched(); v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV); - } while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--); + } while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on RX valid\n"); diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e17836680f49..0a1798eafb0b 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1767,7 +1767,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) AFMT_S8 | AFMT_U16_LE | AFMT_U16_BE | AFMT_S32_LE | AFMT_S32_BE | - AFMT_S24_LE | AFMT_S24_LE | + AFMT_S24_LE | AFMT_S24_BE | AFMT_S24_PACKED; params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b7bba7dc7cf1..0b708134d12f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -487,7 +487,6 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, { struct hda_bus *bus; int err; - char qname[8]; static struct snd_device_ops dev_ops = { .dev_register = snd_hda_bus_dev_register, .dev_free = snd_hda_bus_dev_free, @@ -517,10 +516,12 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, mutex_init(&bus->cmd_mutex); INIT_LIST_HEAD(&bus->codec_list); - snprintf(qname, sizeof(qname), "hda%d", card->number); - bus->workq = create_workqueue(qname); + snprintf(bus->workq_name, sizeof(bus->workq_name), + "hd-audio%d", card->number); + bus->workq = create_singlethread_workqueue(bus->workq_name); if (!bus->workq) { - snd_printk(KERN_ERR "cannot create workqueue %s\n", qname); + snd_printk(KERN_ERR "cannot create workqueue %s\n", + bus->workq_name); kfree(bus); return -ENOMEM; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5810ef588402..09a332ada0c6 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -614,6 +614,7 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; + char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ /* assigned PCMs */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7ca66d654148..144b85276d5a 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -399,7 +399,8 @@ static void print_conn_list(struct snd_info_buffer *buffer, { int c, curr = -1; - if (conn_len > 1 && wid_type != AC_WID_AUD_MIX) + if (conn_len > 1 && wid_type != AC_WID_AUD_MIX && + wid_type != AC_WID_VOL_KNB) curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); snd_iprintf(buffer, " Connection: %d\n", conn_len); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 75de40aaab0a..0177ef8f4c9e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -347,6 +347,7 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, &spec->cur_mux[adc_idx]); } +#ifdef CONFIG_SND_JACK static int conexant_add_jack(struct hda_codec *codec, hda_nid_t nid, int type) { @@ -394,7 +395,6 @@ static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) static int conexant_init_jacks(struct hda_codec *codec) { -#ifdef CONFIG_SND_JACK struct conexant_spec *spec = codec->spec; int i; @@ -422,10 +422,19 @@ static int conexant_init_jacks(struct hda_codec *codec) ++hv; } } -#endif return 0; } +#else +static inline void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ +} + +static inline int conexant_init_jacks(struct hda_codec *codec) +{ + return 0; +} +#endif static int conexant_init(struct hda_codec *codec) { @@ -1566,6 +1575,7 @@ static struct snd_pci_quirk cxt5047_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6700", CXT5047_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5d249a547fbf..ae5c8a0d1479 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1037,6 +1037,7 @@ do_sku: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0272: case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: @@ -1065,6 +1066,7 @@ do_sku: case 0x10ec0882: case 0x10ec0883: case 0x10ec0885: + case 0x10ec0887: case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); @@ -7012,12 +7014,14 @@ static int patch_alc882(struct hda_codec *codec) break; case 0x106b1000: /* iMac 24 */ case 0x106b2800: /* AppleTV */ + case 0x106b3e00: /* iMac 24 Aluminium */ board_config = ALC885_IMAC24; break; case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ + case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; default: @@ -8513,6 +8517,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550", + ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530", ALC888_FUJITSU_XA3530), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3dd4eee70b7c..38428e22428f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1804,6 +1804,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv4", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, "HP dv7", STAC_HP_M4), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600, + "HP dv5", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603, "HP dv5", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, @@ -2539,6 +2541,8 @@ static int stac92xx_build_pcms(struct hda_codec *codec) info->name = "STAC92xx Analog"; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 19d3391e229f..e900cdc84849 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -617,7 +617,7 @@ static int snd_intel8x0_ali_codec_semaphore(struct intel8x0 *chip) int time = 100; if (chip->buggy_semaphore) return 0; /* just ignore ... */ - while (time-- && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY)) + while (--time && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY)) udelay(1); if (! time && ! chip->in_ac97_init) snd_printk(KERN_WARNING "ali_codec_semaphore timeout\n"); diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index c5d67900d666..ff0054b76502 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino@endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index a828746e8a2f..391135f9c6c1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino@endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e3989d406f54..35d99750c383 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -3,7 +3,7 @@ * * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com> + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6c21b50c9375..77620ab98756 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1451,7 +1451,14 @@ static const struct snd_soc_dai wm8753_all_dai[] = { }, }; -struct snd_soc_dai wm8753_dai[2]; +struct snd_soc_dai wm8753_dai[] = { + { + .name = "WM8753 DAI 0", + }, + { + .name = "WM8753 DAI 1", + }, +}; EXPORT_SYMBOL_GPL(wm8753_dai); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5b5afc144478..1cbb7b9b51ce 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -2,8 +2,7 @@ * wm8990.c -- WM8990 ALSA Soc Audio driver * * Copyright 2008 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index ec5e18a78758..05dd5abcddf4 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -302,6 +302,10 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->spcr1 |= RINTM(3); regs->rcr2 |= RFIG; regs->xcr2 |= XFIG; + if (cpu_is_omap2430() || cpu_is_omap34xx()) { + regs->xccr = DXENDLY(1) | XDMAEN; + regs->rccr = RFULL_CYCLE | RDMAEN; + } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b0362dfd5b71..dd3bb2933762 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -175,9 +175,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_pcm_runtime *runtime = substream->runtime; struct omap_runtime_data *prtd = runtime->private_data; + unsigned long flags; int ret = 0; - spin_lock_irq(&prtd->lock); + spin_lock_irqsave(&prtd->lock, flags); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -195,7 +196,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) default: ret = -EINVAL; } - spin_unlock_irq(&prtd->lock); + spin_unlock_irqrestore(&prtd->lock, flags); return ret; } diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c709b9563226..2ab83129d9b0 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2966,6 +2966,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); usb_set_interface(chip->dev, fp->iface, 0); init_usb_pitch(chip->dev, fp->iface, alts, fp); init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max); |