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Commit f43c6064 authored by Linus Torvalds's avatar Linus Torvalds
Browse files

Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound 

Pull sound fixes from Takashi Iwai:
 "This batch contains a few USB audio fixes, a couple of HD-audio
  quirks, various small ASoC driver fixes in addition to an ASoC core
  fix that may lead to memory corruption.

  Unfortunately slightly more volume than the previous pull request, but
  all are reasonable regression fixes"

* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Add a fixup for Gateway LT27
  ASoC: tegra: fix Tegra30 I2S capture parameter setup
  ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525
  ALSA: hda - Fix missing mute controls for CX5051
  ALSA: usb-audio: fix automatic Roland/Yamaha MIDI detection
  ALSA: 6fire: make buffers DMA-able (midi)
  ALSA: 6fire: make buffers DMA-able (pcm)
  ALSA: hda - Add pinfix for LG LW25 laptop
  ASoC: cs42l52: Add new TLV for Beep Volume
  ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume
  ASoC: dapm: Fix empty list check in dapm_new_mux()
  ASoC: sgtl5000: fix buggy 'Capture Attenuate Switch' control
  ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture
parents 89cb9ae2 1801928e

sound/pci/hda/hda_generic.c

+3 −3
Original line number Diff line number Diff line
@@ -522,7 +522,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, 
}



#define nid_has_mute(codec, nid, dir) \

	check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE)

	check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))

#define nid_has_volume(codec, nid, dir) \

	check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS)
@@ -624,7 +624,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, 
		if (enable)

			val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;

	}

	if (caps & AC_AMPCAP_MUTE) {

	if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {

		if (!enable)

			val |= HDA_AMP_MUTE;

	}
@@ -648,7 +648,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec, 
{

	unsigned int mask = 0xff;



	if (caps & AC_AMPCAP_MUTE) {

	if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {

		if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL))

			mask &= ~0x80;

	}

sound/pci/hda/patch_realtek.c

+11 −0
Original line number Diff line number Diff line
@@ -1031,6 +1031,7 @@ enum { 
	ALC880_FIXUP_GPIO2,

	ALC880_FIXUP_MEDION_RIM,

	ALC880_FIXUP_LG,

	ALC880_FIXUP_LG_LW25,

	ALC880_FIXUP_W810,

	ALC880_FIXUP_EAPD_COEF,

	ALC880_FIXUP_TCL_S700,
@@ -1089,6 +1090,14 @@ static const struct hda_fixup alc880_fixups[] = { 
			{ }

		}

	},

	[ALC880_FIXUP_LG_LW25] = {

		.type = HDA_FIXUP_PINS,

		.v.pins = (const struct hda_pintbl[]) {

			{ 0x1a, 0x0181344f }, /* line-in */

			{ 0x1b, 0x0321403f }, /* headphone */

			{ }

		}

	},

	[ALC880_FIXUP_W810] = {

		.type = HDA_FIXUP_PINS,

		.v.pins = (const struct hda_pintbl[]) {
@@ -1341,6 +1350,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { 
	SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG),

	SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG),

	SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG),

	SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_FIXUP_LG_LW25),

	SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700),



	/* Below is the copied entries from alc880_quirks.c.
@@ -4329,6 +4339,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { 
	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),

	SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),

	SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),

	SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC),

	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),

	SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),

	SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),

sound/soc/codecs/cs42l52.c

+4 −1
Original line number Diff line number Diff line
@@ -195,6 +195,8 @@ static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); 


static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);



static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0);



static const unsigned int limiter_tlv[] = {

	TLV_DB_RANGE_HEAD(2),

	0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
@@ -451,7 +453,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { 
	SOC_ENUM("Beep Pitch", beep_pitch_enum),

	SOC_ENUM("Beep on Time", beep_ontime_enum),

	SOC_ENUM("Beep off Time", beep_offtime_enum),

	SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),

	SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL,

			0, 0x07, 0x1f, beep_tlv),

	SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),

	SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),

	SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),

sound/soc/codecs/sgtl5000.c

+14 −4
Original line number Diff line number Diff line
@@ -153,6 +153,8 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, 
static int power_vag_event(struct snd_soc_dapm_widget *w,

	struct snd_kcontrol *kcontrol, int event)

{

	const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;



	switch (event) {

	case SND_SOC_DAPM_POST_PMU:

		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
@@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, 
		break;



	case SND_SOC_DAPM_PRE_PMD:

		/*

		 * Don't clear VAG_POWERUP, when both DAC and ADC are

		 * operational to prevent inadvertently starving the

		 * other one of them.

		 */

		if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) &

				mask) != mask) {

			snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,

				SGTL5000_VAG_POWERUP, 0);

			msleep(400);

		}

		break;

	default:

		break;
@@ -388,7 +398,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 
	SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0),

	SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)",

			SGTL5000_CHIP_ANA_ADC_CTRL,

			8, 2, 0, capture_6db_attenuate),

			8, 1, 0, capture_6db_attenuate),

	SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0),



	SOC_DOUBLE_TLV("Headphone Playback Volume",

sound/soc/soc-dapm.c

+4 −3
Original line number Diff line number Diff line
@@ -679,13 +679,14 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) 
		return -EINVAL;

	}



	path = list_first_entry(&w->sources, struct snd_soc_dapm_path,

				list_sink);

	if (!path) {

	if (list_empty(&w->sources)) {

		dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name);

		return -EINVAL;

	}



	path = list_first_entry(&w->sources, struct snd_soc_dapm_path,

				list_sink);



	ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path);

	if (ret < 0)

		return ret;