July 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: bfin-ac97: Fix prototype error following AC'97 refactoring
by Mark Brown 29 Jul '13

29 Jul '13

From: Mark Brown <broonie(a)linaro.org> As part of the multiplatform refactoring for AC'97 the AC'97 bus ops were staticised meaning that the prototype (which was never needed) conflicts with the declaration causing build failures. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/blackfin/bf5xx-ac97.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 15c635e..0c3e22d 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -9,7 +9,6 @@ #ifndef _BF5XX_AC97_H #define _BF5XX_AC97_H -extern struct snd_ac97_bus_ops bf5xx_ac97_ops; extern struct snd_ac97 *ac97; /* Frame format in memory, only support stereo currently */ struct ac97_frame { -- 1.8.3.2

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[alsa-devel] [PATCH 0/2] tea575x: Move from sound to media
by Ondrej Zary 29 Jul '13

29 Jul '13

Hello, TEA575x is neither a sound device nor an i2c device. Let's finally move it from sound/i2c/other to drivers/media/radio. Tested with snd-es1968, snd-fm801 and radio-sf16fmr2. -- Ondrej Zary

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[alsa-devel] [PATCH 1/2] ASoC: core: Add snd_soc_card_get_kcontrol()
by Dimitris Papastamos 29 Jul '13

29 Jul '13

This is useful for drivers who want to grab a pointer to snd_kcontrol outside of the kcontrol callbacks. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 16 ++++++++++++++++ 2 files changed, 18 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7..b33d1de 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -475,6 +475,8 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, const char *long_name, const char *prefix); +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name); int snd_soc_add_codec_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_platform_controls(struct snd_soc_platform *platform, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d82ee38..d68a486 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2299,6 +2299,22 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev, return 0; } +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name) +{ + struct snd_card *card = soc_card->snd_card; + struct snd_kcontrol *kctl; + + if (unlikely(!name)) + return NULL; + + list_for_each_entry(kctl, &card->controls, list) + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) + return kctl; + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); + /** * snd_soc_add_codec_controls - add an array of controls to a codec. * Convenience function to add a list of controls. Many codecs were -- 1.8.3.4

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[alsa-devel] [PATCH] ALSA: hda - Remove analog mic pin override from STAC9228 dell-bios quirk
by Takashi Iwai 29 Jul '13

29 Jul '13

The current fixup for dell-bios model with STAC9228 codec contains the override of pin 0x0c for analog mic. But this is actually just adding a bogus pin and confuses the parser. Better to remove it for the auto-mic switching. Meanwhile, for a possible regression, keep the old configuration as model=dell-bios-amic, so that people can test it again quickly. Tested on Dell 1420n laptop. Reported-and-tested-by: Eric Shattow <lucent(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_sigmatel.c | 14 ++++++++++++-- 2 files changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 809d72b..a46ddb8 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -244,6 +244,7 @@ STAC9227/9228/9229/927x 5stack-no-fp D965 5stack without front panel dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup + dell-bios-amic Fixes with Dell BIOS setup including analog mic volknob Fixes with volume-knob widget 0x24 auto BIOS setup (default) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e2f8359..8f6c357 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -158,6 +158,7 @@ enum { STAC_D965_VERBS, STAC_DELL_3ST, STAC_DELL_BIOS, + STAC_DELL_BIOS_AMIC, STAC_DELL_BIOS_SPDIF, STAC_927X_DELL_DMIC, STAC_927X_VOLKNOB, @@ -3228,8 +3229,6 @@ static const struct hda_fixup stac927x_fixups[] = { [STAC_DELL_BIOS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - /* configure the analog microphone on some laptops */ - { 0x0c, 0x90a79130 }, /* correct the front output jack as a hp out */ { 0x0f, 0x0227011f }, /* correct the front input jack as a mic */ @@ -3239,6 +3238,16 @@ static const struct hda_fixup stac927x_fixups[] = { .chained = true, .chain_id = STAC_927X_DELL_DMIC, }, + [STAC_DELL_BIOS_AMIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* configure the analog microphone on some laptops */ + { 0x0c, 0x90a79130 }, + {} + }, + .chained = true, + .chain_id = STAC_DELL_BIOS, + }, [STAC_DELL_BIOS_SPDIF] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3267,6 +3276,7 @@ static const struct hda_model_fixup stac927x_models[] = { { .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" }, { .id = STAC_DELL_3ST, .name = "dell-3stack" }, { .id = STAC_DELL_BIOS, .name = "dell-bios" }, + { .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" }, { .id = STAC_927X_VOLKNOB, .name = "volknob" }, {} }; -- 1.8.3.1

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[alsa-devel] [patch] ALSA: usx2y: remove an unneeded check
by Dan Carpenter 29 Jul '13

29 Jul '13

The test here is always true because S[i].urb is an array not a pointer. Also it's bogus because the intent was to test: if (S->urb[i]) { instead of: if (S[i].urb) { Anway, usb_kill_urb() and usb_free_urb() accept NULL pointers so we can just remove this. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 1f9bbd5..5a51b18 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -305,11 +305,9 @@ static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S) { int i; for (i = 0; i < URBS_AsyncSeq; ++i) { - if (S[i].urb) { - usb_kill_urb(S->urb[i]); - usb_free_urb(S->urb[i]); - S->urb[i] = NULL; - } + usb_kill_urb(S->urb[i]); + usb_free_urb(S->urb[i]); + S->urb[i] = NULL; } kfree(S->buffer); }

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[alsa-devel] [PATCH] ALSA: compress: fix the return value for SNDRV_COMPRESS_VERSION
by Vinod Koul 29 Jul '13

29 Jul '13

the return value of SNDRV_COMPRESS_VERSION always return default -ENOTTY as the return value was never updated for this call assign return value from put_user() Reported-by: Haynes <hgeorge(a)codeaurora.org> CC: stable(a)vger.kernel.org Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892..9896954 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): -- 1.7.0.4

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[alsa-devel] [PATCH - Preliminary Xonar Xense support 1/1] Added support for Xonar Xense. Only speaker out and mic in working.
by Gen 29 Jul '13

29 Jul '13

Signed-off-by: Gen <madeallup.gen(a)gmail.com> diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda..df99ec9 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8428) }, { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index c8c7f2c..2567810 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -499,6 +499,17 @@ static void xonar_stx_init(struct oxygen *chip) xonar_st_init_common(chip); } +static void xonar_xense_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_st_init(chip); +} + static void xonar_d2_cleanup(struct oxygen *chip) { xonar_disable_output(chip); @@ -1138,6 +1149,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x8428: + chip->model = model_xonar_st; + chip->model.shortname = "Xonar Xense"; + chip->model.chip = "AV100"; + chip->model.init = xonar_xense_init; + chip->model.dac_channels_pcm = 8; + chip->model.dac_channels_mixer = 8; + break; default: return -EINVAL; } -- 1.8.1.2

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[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Selectively call snd_hda_shutup_pins()
by Takashi Iwai 29 Jul '13

29 Jul '13

Instead of calling snd_hda_shutup_pins() unconditionally, allow it be called in spec->shutup callback. In this way, we can avoid calling this function if it causes a problem like we see in the next patch following this. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..dbd59df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -282,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec) { alc_auto_setup_eapd(codec, false); msleep(200); + snd_hda_shutup_pins(codec); } /* generic EAPD initialization */ @@ -826,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec) if (spec && spec->shutup) spec->shutup(codec); - snd_hda_shutup_pins(codec); + else + snd_hda_shutup_pins(codec); } #define alc_free snd_hda_gen_free @@ -2573,15 +2575,13 @@ static void alc269_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (spec->codec_variant != ALC269_TYPE_ALC269VB) - return; - if (spec->codec_variant == ALC269_TYPE_ALC269VB) alc269vb_toggle_power_output(codec, 0); if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } + snd_hda_shutup_pins(codec); } static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, -- 1.8.3.1

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[alsa-devel] [PATCH] ALSA: compress: fix the return value for SNDRV_COMPRESS_VERSION
by Vinod Koul 29 Jul '13

29 Jul '13

Reported-by: Haynes <hgeorge(a)codeaurora.org> CC: stable(a)vger.kernel.org Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892..9896954 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): -- 1.7.0.4

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[alsa-devel] Broken headphone and speaker amplifier supplies for the rt5640 driver
by Lars-Peter Clausen 29 Jul '13

29 Jul '13

Hi, The rt5640 driver sets for both the headphone and speaker amplifier widget the shift to SND_SOC_NOPM, which is -1. This doesn't make much sense and breaks compilation with some changes I've made. I guess for the heaphone amplifier the right fix is to set the shift to RT5640_PWR_HA_BIT. For the speaker amplifier I'd assume that the right bit is RT5640_PWR_CLS_D_BIT. But the speaker amplifier widget also as an event callback in which it already seems to set/clear the bit. I think the right fix is to remove the register change from the event callback and set the shift to RT5640_PWR_CLS_D_BIT. But I neither have the hardware nor the datasheet, so this is all just an educated guess. - Lars

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