December 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: WM8580: Fix R8 initial value
by Jassi Brar 09 Dec '10

09 Dec '10

From: Seungwhan Youn <sw.youn(a)samsung.com> Acc to WM8580 manual, the default value for R8 is 0x10, not 0x1c. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/codecs/wm8580.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6e95332..6be373d 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -179,7 +179,7 @@ static const u16 wm8580_reg[] = { 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/ 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/ - 0x001c, 0x0002, 0x0002, 0x00c2, /*R11*/ + 0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/ 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/ 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/ 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/ -- 1.6.2.5

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[alsa-devel] [PATCH] ASoC: Remove unnecessary structure definitions
by Seungwhan Youn 09 Dec '10

09 Dec '10

This patch removes some legacy structure definitions which are not using in current ASoC drivers. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- include/sound/soc.h | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5c3bce8..6458c71 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -222,10 +222,8 @@ enum snd_soc_bias_level { struct snd_jack; struct snd_soc_card; -struct snd_soc_device; struct snd_soc_pcm_stream; struct snd_soc_ops; -struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; struct snd_soc_dai; struct snd_soc_dai_driver; @@ -235,7 +233,6 @@ struct snd_soc_platform_driver; struct snd_soc_codec; struct snd_soc_codec_driver; struct soc_enum; -struct snd_soc_ac97_ops; struct snd_soc_jack; struct snd_soc_jack_pin; -- 1.6.2.5

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[alsa-devel] [PATCH] ASoC: fix deemphasis control in wm8904/55/60 codecs
by Dmitry Artamonow 09 Dec '10

09 Dec '10

Deemphasis control's .get callback should update control's value instead of returning it - return value of callback function is used for indicating error or success of operation. Signed-off-by: Dmitry Artamonow <mad_soft(a)inbox.ru> --- I discovered this while trying to adapt this deemphasis code to ak4641 codec. sound/soc/codecs/wm8904.c | 3 ++- sound/soc/codecs/wm8955.c | 3 ++- sound/soc/codecs/wm8960.c | 3 ++- 3 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index fca60a0..9001cc4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -818,7 +818,8 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - return wm8904->deemph; + ucontrol->value.enumerated.item[0] = wm8904->deemph; + return 0; } static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index f89ad6c..9cbab8e 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -380,7 +380,8 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - return wm8955->deemph; + ucontrol->value.enumerated.item[0] = wm8955->deemph; + return 0; } static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 8d5efb3..21986c4 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -138,7 +138,8 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - return wm8960->deemph; + ucontrol->value.enumerated.item[0] = wm8960->deemph; + return 0; } static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, -- 1.7.0.4

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[alsa-devel] [PATCH v3 0/3] Add Jack support for
by Olaya, Margarita 09 Dec '10

09 Dec '10

This patch series adds the support for headset jack detection v3: Reorder patches to avoid building issues. Really add the jack detection in the codec Jorge Eduardo Candelaria (3): ASoC: sdp4430: Enable FM stereo pins ASoC: twl6040: Add jack support for headset and handset ASoC: sdp4430: Add Jack support sound/soc/codecs/twl6040.c | 42 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 12 ++++++++++++ sound/soc/omap/sdp4430.c | 39 +++++++++++++++++++++++++++++++++++---- 3 files changed, 89 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH 1/2] ASoC: tlv320dac33: Fix compillation error
by Peter Ujfalusi 09 Dec '10

09 Dec '10

Fix the compilation error introduced by patch: ASoC: tlv320dac33: Avoid multiple soft power up Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index a0ba5d1..d13fda0 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -638,6 +638,7 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { -- 1.7.3.2

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[alsa-devel] [PATCH v2 1/2] ASoC: tlv320dac33: Rename outpup amplifier widget
by Peter Ujfalusi 09 Dec '10

09 Dec '10

Use better name for the widget, and remove the 'Power' from it's name. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index d13fda0..8e340a5 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -599,9 +599,9 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0, &dac33_dapm_abypassr_control), - SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amp Power", + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amplifier", DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power", + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amplifier", DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0), SND_SOC_DAPM_PRE("Prepare Playback", playback_event), @@ -612,15 +612,15 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Analog Left Bypass", "Switch", "LINEL"}, {"Analog Right Bypass", "Switch", "LINER"}, - {"Output Left Amp Power", NULL, "DACL"}, - {"Output Right Amp Power", NULL, "DACR"}, + {"Output Left Amplifier", NULL, "DACL"}, + {"Output Right Amplifier", NULL, "DACR"}, - {"Output Left Amp Power", NULL, "Analog Left Bypass"}, - {"Output Right Amp Power", NULL, "Analog Right Bypass"}, + {"Output Left Amplifier", NULL, "Analog Left Bypass"}, + {"Output Right Amplifier", NULL, "Analog Right Bypass"}, /* output */ - {"LEFT_LO", NULL, "Output Left Amp Power"}, - {"RIGHT_LO", NULL, "Output Right Amp Power"}, + {"LEFT_LO", NULL, "Output Left Amplifier"}, + {"RIGHT_LO", NULL, "Output Right Amplifier"}, }; static int dac33_add_widgets(struct snd_soc_codec *codec) -- 1.7.3.2

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[alsa-devel] [PATCH] [ALSA] ice1712 - working M-Audio Delta 66E support
by Brian Bloniarz 09 Dec '10

09 Dec '10

Rev. E of the M-Audio Delta 66 is partially supported (commit ef2cd2ccad66b4aba518eca7514eface267ee0f3), but the layout of the GPIO pins was still unclear. This patch adds the GPIO definitions so that communication to the CS8247 & 2x AK4524 works correctly. ALSA bug#3327 has more details; users cap & jhunt report there that the GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 = CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1). There has been a lot of conflicting information in the bug, but given these definitions, my Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz. Signed-off-by: Brian Bloniarz <brian.bloniarz(a)gmail.com> --- sound/pci/ice1712/delta.c | 49 ++++++++++++++++++++++++++++++++++++++++++++- sound/pci/ice1712/delta.h | 11 ++++++++++ 2 files changed, 59 insertions(+), 1 deletions(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 712c171..7b62de0 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -96,6 +96,11 @@ static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice) tmp |= ICE1712_DELTA_AP_CCLK | ICE1712_DELTA_AP_CS_CODEC; tmp &= ~ICE1712_DELTA_AP_CS_DIGITAL; break; + case ICE1712_SUBDEVICE_DELTA66E: + tmp |= ICE1712_DELTA_66E_CCLK | ICE1712_DELTA_66E_CS_CHIP_A | + ICE1712_DELTA_66E_CS_CHIP_B; + tmp &= ~ICE1712_DELTA_66E_CS_CS8427; + break; case ICE1712_SUBDEVICE_VX442: tmp |= ICE1712_VX442_CCLK | ICE1712_VX442_CODEC_CHIP_A | ICE1712_VX442_CODEC_CHIP_B; tmp &= ~ICE1712_VX442_CS_DIGITAL; @@ -119,6 +124,9 @@ static void ap_cs8427_codec_deassert(struct snd_ice1712 *ice, unsigned char tmp) case ICE1712_SUBDEVICE_DELTA410: tmp |= ICE1712_DELTA_AP_CS_DIGITAL; break; + case ICE1712_SUBDEVICE_DELTA66E: + tmp |= ICE1712_DELTA_66E_CS_CS8427; + break; case ICE1712_SUBDEVICE_VX442: tmp |= ICE1712_VX442_CS_DIGITAL; break; @@ -276,6 +284,20 @@ static void delta1010lt_ak4524_lock(struct snd_akm4xxx *ak, int chip) } /* + * AK4524 on Delta66 rev E to choose the chip address + */ +static void delta66e_ak4524_lock(struct snd_akm4xxx *ak, int chip) +{ + struct snd_ak4xxx_private *priv = (void *)ak->private_value[0]; + struct snd_ice1712 *ice = ak->private_data[0]; + + snd_ice1712_save_gpio_status(ice); + priv->cs_mask = + priv->cs_addr = chip == 0 ? ICE1712_DELTA_66E_CS_CHIP_A : + ICE1712_DELTA_66E_CS_CHIP_B; +} + +/* * AK4528 on VX442 to choose the chip mask */ static void vx442_ak4524_lock(struct snd_akm4xxx *ak, int chip) @@ -487,6 +509,29 @@ static struct snd_ak4xxx_private akm_delta1010lt_priv __devinitdata = { .mask_flags = 0, }; +static struct snd_akm4xxx akm_delta66e __devinitdata = { + .type = SND_AK4524, + .num_adcs = 4, + .num_dacs = 4, + .ops = { + .lock = delta66e_ak4524_lock, + .set_rate_val = delta_ak4524_set_rate_val + } +}; + +static struct snd_ak4xxx_private akm_delta66e_priv __devinitdata = { + .caddr = 2, + .cif = 0, /* the default level of the CIF pin from AK4524 */ + .data_mask = ICE1712_DELTA_66E_DOUT, + .clk_mask = ICE1712_DELTA_66E_CCLK, + .cs_mask = 0, + .cs_addr = 0, /* set later */ + .cs_none = 0, + .add_flags = 0, + .mask_flags = 0, +}; + + static struct snd_akm4xxx akm_delta44 __devinitdata = { .type = SND_AK4524, .num_adcs = 4, @@ -644,9 +689,11 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_init(ak, &akm_delta44, &akm_delta44_priv, ice); break; case ICE1712_SUBDEVICE_VX442: - case ICE1712_SUBDEVICE_DELTA66E: err = snd_ice1712_akm4xxx_init(ak, &akm_vx442, &akm_vx442_priv, ice); break; + case ICE1712_SUBDEVICE_DELTA66E: + err = snd_ice1712_akm4xxx_init(ak, &akm_delta66e, &akm_delta66e_priv, ice); + break; default: snd_BUG(); return -EINVAL; diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index 1a0ac6c..11a9c3a 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -144,6 +144,17 @@ extern struct snd_ice1712_card_info snd_ice1712_delta_cards[]; #define ICE1712_DELTA_1010LT_CS_NONE 0x50 /* nothing */ #define ICE1712_DELTA_1010LT_WORDCLOCK 0x80 /* sample clock source: 0 = Word Clock Input, 1 = S/PDIF Input ??? */ +/* M-Audio Delta 66 rev. E definitions. + * Newer revisions of Delta 66 have CS8427 over SPI for + * S/PDIF transceiver instead of CS8404/CS8414. */ +/* 0x01 = DFS */ +#define ICE1712_DELTA_66E_CCLK 0x02 /* SPI clock */ +#define ICE1712_DELTA_66E_DIN 0x04 /* data input */ +#define ICE1712_DELTA_66E_DOUT 0x08 /* data output */ +#define ICE1712_DELTA_66E_CS_CS8427 0x10 /* chip select, low = CS8427 */ +#define ICE1712_DELTA_66E_CS_CHIP_A 0x20 /* AK4524 #0 */ +#define ICE1712_DELTA_66E_CS_CHIP_B 0x40 /* AK4524 #1 */ + /* Digigram VX442 definitions */ #define ICE1712_VX442_CCLK 0x02 /* SPI clock */ #define ICE1712_VX442_DIN 0x04 /* data input */ -- 1.7.1

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[alsa-devel] [PATCH] ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work
by David Henningsson 09 Dec '10

09 Dec '10

BugLink: http://launchpad.net/497546 Confirmed that the ideapad model works better than the current quirk for Dell Vostro 320. With this patch, there is noone left using the DELL_LAPTOP model, should we remove it completely? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[PATCH] ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work
by David Henningsson 09 Dec '10

09 Dec '10

BugLink: http://launchpad.net/497546 Confirmed that the ideapad model works better than the current quirk for Dell Vostro 320. Cc: stable(a)kernel.org (2.6.35+) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 46bbf14..ee9c3ed 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3097,8 +3097,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { static struct snd_pci_quirk cxt5066_cfg_tbl[] = { 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", - CXT5066_DELL_LAPTOP), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), -- 1.7.1 --------------070307010403030000030409--

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[alsa-devel] [PATCH v3 3/3] ASoC: sdp4430: Add Jack support
by Olaya, Margarita 08 Dec '10

08 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Use jack framework to enable detection for the headset microphone and stereo output. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/sdp4430.c | 28 ++++++++++++++++++++++++++++ 1 files changed, 28 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 0c37c51..ca7fb90 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -24,6 +24,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <plat/hardware.h> @@ -65,6 +66,21 @@ static struct snd_soc_ops sdp4430_ops = { .hw_params = sdp4430_hw_params, }; +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -160,7 +176,19 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); ret = snd_soc_dapm_sync(dapm); + if (ret) + return ret; + + /* Headset jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); return ret; } -- 1.7.0.4

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