diff options
Diffstat (limited to 'sound')
58 files changed, 335 insertions, 260 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 0d7b25e81643..4e1fda75c1c9 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -106,7 +106,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { .prepare = pxa2xx_ac97_pcm_prepare, }; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_do_suspend(struct snd_card *card) { @@ -243,7 +243,7 @@ static struct platform_driver pxa2xx_ac97_driver = { .driver = { .name = "pxa2xx-ac97", .owner = THIS_MODULE, -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, #endif }, diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index eb4ceb71123e..277ebce23a45 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -452,6 +452,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) dac->regs = ioremap(regs->start, resource_size(regs)); if (!dac->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto out_free_card; } @@ -534,7 +535,7 @@ out_put_pclk: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_abdac_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index bf47025bdf45..9052aff37f64 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -278,14 +278,9 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, if (retval < 0) return retval; /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (cpu_is_at32ap7000()) { - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); - } + if (cpu_is_at32ap7000() && retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -980,6 +975,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) if (!chip->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto err_ioremap; } @@ -1134,7 +1130,7 @@ err_snd_card_new: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_ac97c_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 4e7ec2b49873..d0f00356fc11 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -101,7 +101,7 @@ void *snd_malloc_sgbuf_pages(struct device *device, if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device, chunk, &tmpb) < 0) { if (!sgbuf->pages) - return NULL; + goto _failed; if (!res_size) goto _failed; size = sgbuf->pages * PAGE_SIZE; diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 1128b35b2b05..5a34355e78e8 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1176,7 +1176,7 @@ static int __devexit loopback_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int loopback_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index f7d3bfc6bca8..54bb6644a598 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1064,7 +1064,7 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_dummy_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 6ca59fc6dcb9..ef171295f6d4 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -199,7 +199,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) pcspkr_stop_sound(); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pcsp_suspend(struct device *dev) { struct snd_pcsp *chip = dev_get_drvdata(dev); @@ -212,7 +212,7 @@ static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL); #define PCSP_PM_OPS &pcsp_pm #else #define PCSP_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static void pcsp_shutdown(struct platform_device *dev) { diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 2d67c78c9f4b..f7cdaf51512d 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -233,7 +233,7 @@ static int __devinit snd_card_als100_probe(int dev, irq[dev], dma8[dev], dma16[dev]); } - if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { + if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) { snd_card_free(card); return error; } diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index 733b014ec7d1..b2b3c014221a 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed) if (speed > 0) { int tmp; - int s = speed * devc->channels; + int s; if (speed < 5000) speed = 5000; if (speed > 44100) speed = 44100; + s = speed * devc->channels; + devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; tmp = 256 - devc->tconst; diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f75f5ffdfdfb..a71d1c14a0f6 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX && codec_index != CS46XX_SECONDARY_CODEC_INDEX)) - return -EINVAL; + return 0xffff; chip->active_ctrl(chip, 1); diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 8e40262d4117..2f6e9c762d3f 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1725,8 +1725,10 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, atc_connect_resources(atc); atc->timer = ct_timer_new(atc); - if (!atc->timer) + if (!atc->timer) { + err = -ENOMEM; goto error1; + } err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops); if (err < 0) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 4f502a2bdc3c..0a436626182b 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -326,7 +326,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst for (page = blk->first_page; page <= blk->last_page; page++, idx++) { unsigned long ofs = idx << PAGE_SHIFT; dma_addr_t addr; - addr = snd_pcm_sgbuf_get_addr(substream, ofs); + if (ofs >= runtime->dma_bytes) + addr = emu->silent_page.addr; + else + addr = snd_pcm_sgbuf_get_addr(substream, ofs); if (! is_valid_page(emu, addr)) { printk(KERN_ERR "emu: failure page = %d\n", idx); mutex_unlock(&hdr->block_mutex); diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 647218d69f68..4f7d2dfcef7b 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -332,13 +332,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, if (cfg->dig_outs) snd_printd(" dig-out=0x%x/0x%x\n", cfg->dig_out_pins[0], cfg->dig_out_pins[1]); - snd_printd(" inputs:"); + snd_printd(" inputs:\n"); for (i = 0; i < cfg->num_inputs; i++) { - snd_printd(" %s=0x%x", + snd_printd(" %s=0x%x\n", hda_get_autocfg_input_label(codec, cfg, i), cfg->inputs[i].pin); } - snd_printd("\n"); if (cfg->dig_in_pin) snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0bc2315b181d..0849aac449f2 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -231,16 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); +static bool ctl_has_mute(struct snd_kcontrol *kcontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + return query_amp_caps(codec, get_amp_nid(kcontrol), + get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE; +} + /* get/put callbacks for beep mute mixer switches */ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) { + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = - beep->enabled; + ucontrol->value.integer.value[1] = beep->enabled; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -252,9 +258,20 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) - snd_hda_enable_beep_device(codec, - *ucontrol->value.integer.value); + if (beep) { + u8 chs = get_amp_channels(kcontrol); + int enable = 0; + long *valp = ucontrol->value.integer.value; + if (chs & 1) { + enable |= *valp; + valp++; + } + if (chs & 2) + enable |= *valp; + snd_hda_enable_beep_device(codec, enable); + } + if (!ctl_has_mute(kcontrol)) + return 0; return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20eb7a2..f560051a949e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1386,6 +1386,44 @@ int snd_hda_codec_configure(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_codec_configure); +/* update the stream-id if changed */ +static void update_pcm_stream_id(struct hda_codec *codec, + struct hda_cvt_setup *p, hda_nid_t nid, + u32 stream_tag, int channel_id) +{ + unsigned int oldval, newval; + + if (p->stream_tag != stream_tag || p->channel_id != channel_id) { + oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + newval = (stream_tag << 4) | channel_id; + if (oldval != newval) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + newval); + p->stream_tag = stream_tag; + p->channel_id = channel_id; + } +} + +/* update the format-id if changed */ +static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p, + hda_nid_t nid, int format) +{ + unsigned int oldval; + + if (p->format_id != format) { + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_STREAM_FORMAT, 0); + if (oldval != format) { + msleep(1); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, + format); + } + p->format_id = format; + } +} + /** * snd_hda_codec_setup_stream - set up the codec for streaming * @codec: the CODEC to set up @@ -1400,7 +1438,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, { struct hda_codec *c; struct hda_cvt_setup *p; - unsigned int oldval, newval; int type; int i; @@ -1413,29 +1450,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p = get_hda_cvt_setup(codec, nid); if (!p) return; - /* update the stream-id if changed */ - if (p->stream_tag != stream_tag || p->channel_id != channel_id) { - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - p->stream_tag = stream_tag; - p->channel_id = channel_id; - } - /* update the format-id if changed */ - if (p->format_id != format) { - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, 0); - if (oldval != format) { - msleep(1); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - p->format_id = format; - } + + if (codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + update_pcm_stream_id(codec, p, nid, stream_tag, channel_id); + if (!codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + p->active = 1; p->dirty = 0; @@ -3497,7 +3518,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg { int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); - if (sup < 0) + if (sup == -1) return false; if (sup & power_state) return true; @@ -4433,6 +4454,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) * then there is no need to go through power up here. */ if (codec->power_on) { + if (codec->power_transition < 0) + codec->power_transition = 0; spin_unlock(&codec->power_lock); return; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d330ca54..7fbc1bcaf1a9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -861,6 +861,7 @@ struct hda_codec { unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ + unsigned int pcm_format_first:1; /* PCM format must be set first */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ int power_transition; /* power-state in transition */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced182fd1..60882c62f180 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, LPT_LP}," "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," @@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c20), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c21), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258fc700..6894ec66258c 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer, if (digi1 & AC_DIG1_EMPHASIS) snd_iprintf(buffer, " Preemphasis"); if (digi1 & AC_DIG1_COPYRIGHT) - snd_iprintf(buffer, " Copyright"); + snd_iprintf(buffer, " Non-Copyright"); if (digi1 & AC_DIG1_NONAUDIO) snd_iprintf(buffer, " Non-Audio"); if (digi1 & AC_DIG1_PROFESSIONAL) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index d0d3540e39e7..49750a96d649 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); } - if (dac) + if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); } @@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } - if (adc) + if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP)) snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } @@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) { + snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) { + snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -464,50 +472,17 @@ exit: } /* - * PCM stuffs + * PCM callbacks */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) +static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd("ca0132_setup_stream: " - "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + struct ca0132_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } -/* - * PCM callbacks - */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -515,10 +490,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, + stream_tag, format, substream); } static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -526,92 +499,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dacs[0]); - - return 0; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } /* * Digital out */ -static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_out); - - return 0; -} - -/* - * Analog capture - */ -static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); } -static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->adcs[substream->number]); - - return 0; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); } -/* - * Digital capture - */ -static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format); - - return 0; -} - -static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_in); - - return 0; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } /* @@ -621,6 +547,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_playback_pcm_open, .prepare = ca0132_playback_pcm_prepare, .cleanup = ca0132_playback_pcm_cleanup }, @@ -630,10 +557,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_capture_pcm_prepare, - .cleanup = ca0132_capture_pcm_cleanup - }, }; static struct hda_pcm_stream ca0132_pcm_digital_playback = { @@ -641,6 +564,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_dig_playback_pcm_open, + .close = ca0132_dig_playback_pcm_close, .prepare = ca0132_dig_playback_pcm_prepare, .cleanup = ca0132_dig_playback_pcm_cleanup }, @@ -650,10 +575,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_dig_capture_pcm_prepare, - .cleanup = ca0132_dig_capture_pcm_cleanup - }, }; static int ca0132_build_pcms(struct hda_codec *codec) @@ -928,18 +849,16 @@ static int ca0132_build_controls(struct hda_codec *codec) spec->dig_out); if (err < 0) return err; - err = add_out_volume(codec, spec->dig_out, "IEC958"); + err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); if (err < 0) return err; + /* spec->multiout.share_spdif = 1; */ } if (spec->dig_in) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); if (err < 0) return err; - err = add_in_volume(codec, spec->dig_in, "IEC958"); - if (err < 0) - return err; } return 0; } @@ -961,6 +880,9 @@ static void ca0132_config(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; + /* line-outs */ cfg->line_outs = 1; cfg->line_out_pins[0] = 0x0b; /* front */ @@ -988,14 +910,24 @@ static void ca0132_config(struct hda_codec *codec) /* Mic-in */ spec->input_pins[0] = 0x12; - spec->input_labels[0] = "Mic-In"; + spec->input_labels[0] = "Mic"; spec->adcs[0] = 0x07; /* Line-In */ spec->input_pins[1] = 0x11; - spec->input_labels[1] = "Line-In"; + spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } static void ca0132_init_chip(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 14361184ae1e..5e22a8f43d2e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2967,12 +2967,10 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { }; static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO), SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), @@ -2988,14 +2986,10 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO), - SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), - SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 69b928449789..8f23374fa642 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -877,8 +877,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; - hinfo->nid = 0; /* clear the leftover value */ - /* Validate hinfo */ pin_idx = hinfo_to_pin_index(spec, hinfo); if (snd_BUG_ON(pin_idx < 0)) @@ -1163,6 +1161,14 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } +static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_cleanup_stream(codec, hinfo->nid); + return 0; +} + static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1202,6 +1208,7 @@ static const struct hda_pcm_ops generic_ops = { .open = hdmi_pcm_open, .close = hdmi_pcm_close, .prepare = generic_hdmi_playback_pcm_prepare, + .cleanup = generic_hdmi_playback_pcm_cleanup, }; static int generic_hdmi_build_pcms(struct hda_codec *codec) @@ -1220,7 +1227,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; pstr->substreams = 1; pstr->ops = generic_ops; - pstr->nid = 1; /* FIXME: just for avoiding a debug WARNING */ /* other pstr fields are set in open */ } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 344b221d2102..4f81dd44c837 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6099,6 +6099,8 @@ static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_PCM_44K] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_pcm_44k, + .chained = true, + .chain_id = ALC269_FIXUP_QUANTA_MUTE }, [ALC269_FIXUP_STEREO_DMIC] = { .type = ALC_FIXUP_FUNC, @@ -6206,9 +6208,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), #if 0 diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 94040ccf8e8f..ea5775a1a7db 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -5748,7 +5749,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5773,7 +5773,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5788,7 +5787,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5796,6 +5794,9 @@ again: break; } + if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 80d90cb42853..430771776915 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec) { struct via_spec *spec = codec->spec; vt1708_stop_hp_work(spec); + + if (spec->codec_type == VT1802) { + /* Fix pop noise on headphones */ + int i; + for (i = 0; i < spec->autocfg.hp_outs; i++) + snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0); + } + return 0; } #endif diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index d1ab43706735..5579b08bb35b 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -851,6 +851,8 @@ static int __devinit lx_pcm_create(struct lx6464es *chip) /* hardcoded device name & channel count */ err = snd_pcm_new(chip->card, (char *)card_name, 0, 1, 1, &pcm); + if (err < 0) + return err; pcm->private_data = chip; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b8ac8710f47f..b12308b5ba2a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6585,7 +6585,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, snd_printk(KERN_ERR "HDSPM: " "unable to kmalloc Mixer memory of %d Bytes\n", (int)sizeof(struct hdspm_mixer)); - return err; + return -ENOMEM; } hdspm->port_names_in = NULL; diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 512434efcc31..805ab6e9a78f 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1377,8 +1377,9 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out_cleanup; - if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, - sis)) { + rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, + sis); + if (rc) { dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq); goto error_out_cleanup; } diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index f5ceb6f282de..210cafe04890 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -143,7 +143,7 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_pmac_driver_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 1aa52eff526a..9b18b5243a56 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -1040,6 +1040,7 @@ static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev) GFP_KERNEL); if (!the_card.null_buffer_start_vaddr) { pr_info("%s: nullbuffer alloc failed\n", __func__); + ret = -ENOMEM; goto clean_preallocate; } pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__, diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index 318c5ba5360f..dfb744381c42 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create); void sport_delete(struct sport_device *sport) { + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); sport_free_resource(sport); + kfree(sport); } EXPORT_SYMBOL(sport_delete); diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 3c795921c5f6..23b40186f9b8 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2406,6 +2406,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + + /* Inform SoC Core that we have our own I/O arrangements. */ + codec->control_data = (void *)true; + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 8c39dddd7d00..11b1b714b8b5 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -186,6 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 6276e352125f..8f726c063f42 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -581,6 +581,8 @@ static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + codec->control_data = priv->mc13xxx; + mc13xxx_lock(priv->mc13xxx); /* these are the reset values */ diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8af6a5245b18..df2f99d1d428 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -239,6 +239,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ + {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ @@ -1357,8 +1358,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; - snd_soc_dapm_new_widgets(&codec->dapm); - return 0; err: diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 982e437799a8..33c0f3d39c87 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -340,6 +340,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6537f16d383e..e33d327396ad 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), -SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, - ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), @@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); @@ -349,10 +343,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, - NULL, 0), -SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, - NULL, 0), SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, NULL, 0), @@ -466,8 +456,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), -ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), -ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), @@ -553,8 +541,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "EQ4", "EQ4" }, \ { name, "DRC1L", "DRC1L" }, \ { name, "DRC1R", "DRC1R" }, \ - { name, "DRC2L", "DRC2L" }, \ - { name, "DRC2R", "DRC2R" }, \ { name, "LHPF1", "LHPF1" }, \ { name, "LHPF2", "LHPF2" }, \ { name, "LHPF3", "LHPF3" }, \ @@ -639,6 +625,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -675,8 +670,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), - ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), - ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8033f7065189..01ebbcc5c6a4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -681,6 +681,18 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + + { "IN4L PGA", NULL, "IN4L" }, + { "IN4R PGA", NULL, "IN4R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index eaf65863ec21..ce6720073798 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2501,6 +2501,9 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*250k */ snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x100); + + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + msleep(100); break; case SND_SOC_BIAS_OFF: @@ -3730,21 +3733,6 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); - regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); - - /* Bias enable at 2*50k for ramp */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, - WM8962_BIAS_ENA | 0x180); - - msleep(5); - - /* VMID back to 2x250k for standby */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK, 0x100); - return 0; } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bb62f4b3d563..6c9eeca85b95 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2649,7 +2649,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_rate = params_rate(params) * 2; + bclk_rate = params_rate(params) * 4; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: bclk_rate *= 16; @@ -3253,10 +3253,13 @@ static void wm8994_mic_work(struct work_struct *work) int ret; int report; + pm_runtime_get_sync(dev); + ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, ®); if (ret < 0) { dev_err(dev, "Failed to read microphone status: %d\n", ret); + pm_runtime_put(dev); return; } @@ -3299,6 +3302,8 @@ static void wm8994_mic_work(struct work_struct *work) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); + + pm_runtime_put(dev); } static irqreturn_t wm8994_mic_irq(int irq, void *data) @@ -3421,12 +3426,15 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) int reg; bool present; + pm_runtime_get_sync(codec->dev); + mutex_lock(&wm8994->accdet_lock); reg = snd_soc_read(codec, WM1811_JACKDET_CTRL); if (reg < 0) { dev_err(codec->dev, "Failed to read jack status: %d\n", reg); mutex_unlock(&wm8994->accdet_lock); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3491,6 +3499,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + pm_runtime_put(codec->dev); return IRQ_HANDLED; } @@ -3602,6 +3611,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) return IRQ_HANDLED; + pm_runtime_get_sync(codec->dev); + /* We may occasionally read a detection without an impedence * range being provided - if that happens loop again. */ @@ -3612,6 +3623,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_err(codec->dev, "Failed to read mic detect status: %d\n", reg); + pm_runtime_put(codec->dev); return IRQ_NONE; } @@ -3639,6 +3651,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_warn(codec->dev, "Accessory detection with no callback\n"); out: + pm_runtime_put(codec->dev); return IRQ_HANDLED; } @@ -4025,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_intercon, + ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 099e6ec32125..c6d2076a796b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -148,7 +148,7 @@ SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), -SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), +SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0), SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), @@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]); /* Mic select */ static const struct snd_kcontrol_new wm9712_mic_src_controls = -SOC_DAPM_ENUM("Route", wm9712_enum[7]); +SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]); /* diff select */ static const struct snd_kcontrol_new wm9712_diff_sel_controls = @@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectl_controls), SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectr_controls), -SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0, +SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, + &wm9712_mic_src_controls), +SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, &wm9712_mic_src_controls), SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, &wm9712_diff_sel_controls), @@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), +SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), @@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { {"Mic PGA", NULL, "MIC1"}, {"Mic PGA", NULL, "MIC2"}, + /* microphones */ + {"Differential Mic", NULL, "MIC1"}, + {"Differential Mic", NULL, "MIC2"}, + {"Left Mic Select Source", "Mic 1", "MIC1"}, + {"Left Mic Select Source", "Mic 2", "MIC2"}, + {"Left Mic Select Source", "Stereo", "MIC1"}, + {"Left Mic Select Source", "Differential", "Differential Mic"}, + {"Right Mic Select Source", "Mic 1", "MIC1"}, + {"Right Mic Select Source", "Mic 2", "MIC2"}, + {"Right Mic Select Source", "Stereo", "MIC2"}, + {"Right Mic Select Source", "Differential", "Differential Mic"}, + /* left capture selector */ {"Left Capture Select", "Mic", "MIC1"}, {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, @@ -619,6 +634,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; + codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 3eb19fb71d17..d0b8a3287a85 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1196,6 +1196,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (wm9713 == NULL) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); + codec->control_data = wm9713; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 95441bfc8190..ce5e5cd254dd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -380,14 +380,20 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* enable FIFO */ + if (dev->txnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } mcasp_start_tx(dev); } else { - if (dev->rxnumevt) /* enable FIFO */ + if (dev->rxnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } mcasp_start_rx(dev); } } diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 28dd76c7cb1c..81d7728cf67f 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - .channels_min = 1, + /* The SSI does not support monaural audio. */ + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 99a997f19bb9..b6fa77678d97 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC if SND_MXS_SOC config SND_SOC_MXS_SGTL5000 - tristate "SoC Audio support for i.MX boards with sgtl5000" + tristate "SoC Audio support for MXS boards with sgtl5000" depends on I2C select SND_SOC_SGTL5000 help diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aba71bfa33b1..b3030718c228 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -394,9 +394,14 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; u32 scr, stat; int ret; + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + /* mclk should already be set */ if (!saif->mclk && saif->mclk_in_use) { dev_err(cpu_dai->dev, "set mclk first\n"); @@ -420,6 +425,25 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, return ret; } + /* prepare clk in hw_param, enable in trigger */ + clk_prepare(saif->clk); + if (saif != master_saif) { + /* + * Set an initial clock rate for the saif internal logic to work + * properly. This is important when working in EXTMASTER mode + * that uses the other saif's BITCLK&LRCLK but it still needs a + * basic clock which should be fast enough for the internal + * logic. + */ + clk_enable(saif->clk); + ret = clk_set_rate(saif->clk, 24000000); + clk_disable(saif->clk); + if (ret) + return ret; + + clk_prepare(master_saif->clk); + } + scr = __raw_readl(saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_WORD_LENGTH; diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 34835e8a9160..d33c48baaf71 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -745,7 +745,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - if (mcbsp->pdata->mux_signal) + if (!mcbsp->pdata->mux_signal) return -EINVAL; switch (mux) { diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1046083e90a0..acdd3ef14e08 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -820,3 +820,4 @@ module_platform_driver(asoc_mcbsp_driver); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-mcbsp"); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5a649da9122a..f0feb06615f8 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-pcm-audio"); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index b7b2a1f91425..89b064650f14 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,7 @@ #include <sound/pcm_params.h> #include <plat/audio.h> -#include <plat/dma.h> +#include <mach/dma.h> #include "dma.h" #include "pcm.h" diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f219b2f7ee68..c501af6d8dbe 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->cpu_dai) { - dev_dbg(card->dev, "CPU DAI %s not registered\n", + dev_err(card->dev, "CPU DAI %s not registered\n", dai_link->cpu_dai_name); return -EPROBE_DEFER; } @@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->codec_dai) { - dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dev_err(card->dev, "CODEC DAI %s not registered\n", dai_link->codec_dai_name); return -EPROBE_DEFER; } } if (!rtd->codec) { - dev_dbg(card->dev, "CODEC %s not registered\n", + dev_err(card->dev, "CODEC %s not registered\n", dai_link->codec_name); return -EPROBE_DEFER; } @@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) rtd->platform = platform; } if (!rtd->platform) { - dev_dbg(card->dev, "platform %s not registered\n", + dev_err(card->dev, "platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } @@ -1096,7 +1096,7 @@ static int soc_probe_codec(struct snd_soc_card *card, } /* If the driver didn't set I/O up try regmap */ - if (!codec->control_data) + if (!codec->write && dev_get_regmap(codec->dev, NULL)) snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (driver->controls) @@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num) return 0; } + dev_err(card->dev, "%s not registered\n", aux_dev->codec_name); + return -EPROBE_DEFER; } diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7f8b3b7428bb..0c172938b82a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -103,7 +103,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) } /* Report before the DAPM sync to help users updating micbias status */ - blocking_notifier_call_chain(&jack->notifier, status, jack); + blocking_notifier_call_chain(&jack->notifier, jack->status, jack); snd_soc_dapm_sync(dapm); diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index d684df294c0c..e463529b38bb 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -177,7 +177,7 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) } alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (alc5632->gpio_hp_det == -ENODEV) + if (alc5632->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; ret = snd_soc_of_parse_card_name(card, "nvidia,model"); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0c5bb33d258e..d4f14e492341 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -284,27 +284,27 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } else if (np) { pdata->gpio_spkr_en = of_get_named_gpio(np, "nvidia,spkr-en-gpios", 0); - if (pdata->gpio_spkr_en == -ENODEV) + if (pdata->gpio_spkr_en == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_mute = of_get_named_gpio(np, "nvidia,hp-mute-gpios", 0); - if (pdata->gpio_hp_mute == -ENODEV) + if (pdata->gpio_hp_mute == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (pdata->gpio_hp_det == -ENODEV) + if (pdata->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_int_mic_en = of_get_named_gpio(np, "nvidia,int-mic-en-gpios", 0); - if (pdata->gpio_int_mic_en == -ENODEV) + if (pdata->gpio_int_mic_en == -EPROBE_DEFER) return -EPROBE_DEFER; pdata->gpio_ext_mic_en = of_get_named_gpio(np, "nvidia,ext-mic-en-gpios", 0); - if (pdata->gpio_ext_mic_en == -ENODEV) + if (pdata->gpio_ext_mic_en == -EPROBE_DEFER) return -EPROBE_DEFER; } diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 62ac0285bfaf..057e28ef770e 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -21,7 +21,7 @@ #include <linux/mfd/dbx500-prcmu.h> #include <mach/hardware.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #include <sound/soc.h> #include <sound/soc-dai.h> diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index ee14d2dac2f5..5c472f335a64 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -19,7 +19,7 @@ #include <linux/slab.h> #include <mach/hardware.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #include <sound/soc.h> diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 7f71b4a0d4bc..2d9136da9865 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -17,7 +17,7 @@ #include <linux/platform_device.h> -#include <mach/board-mop500-msp.h> +#include <mach/msp.h> #define MSP_INPUT_FREQ_APB 48000000 diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0f647d22cb4a..c41181202688 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -821,10 +821,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (++ep->use_count != 1) return 0; - /* just to be sure */ - deactivate_urbs(ep, 0, 1); - wait_clear_urbs(ep); - ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a1298f379428..62ec808ed792 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -544,6 +544,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->last_frame_number = 0; runtime->delay = 0; + /* clear the pending deactivation on the target EPs */ + deactivate_endpoints(subs); + /* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) |
