January 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: wm8994: Remove ASoC level register cache sync
by Mark Brown 31 Jan '12

31 Jan '12

Now we've switched over to regmap the ASoC level cache sync will be ineffectual and potentially harmful as there is no longer an ASoC level cache. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ae05e29..ca2db79 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2736,11 +2736,6 @@ static int wm8994_resume(struct snd_soc_codec *codec) codec->cache_only = 0; } - /* Restore the registers */ - ret = snd_soc_cache_sync(codec); - if (ret != 0) - dev_err(codec->dev, "Failed to sync cache: %d\n", ret); - wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { -- 1.7.9.rc1

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[alsa-devel] [PATCH] ALSA: HDA: Remove quirk for Toshiba Qosmio G50
by David Henningsson 31 Jan '12

31 Jan '12

The user reports that model=auto works better than current handling on a 3.2 based kernel (with jack detection patches backported). Since model=auto is what we prefer these days anyway, the quirk should be removed. Alsa-info for the relevant machine: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/923316/+attachment/270… BugLink: https://bugs.launchpad.net/bugs/923316 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 ------------- 1 files changed, 0 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 42b6a01..a8e82be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4747,7 +4747,6 @@ enum { ALC262_FIXUP_FSC_H270, ALC262_FIXUP_HP_Z200, ALC262_FIXUP_TYAN, - ALC262_FIXUP_TOSHIBA_RX1, ALC262_FIXUP_LENOVO_3000, ALC262_FIXUP_BENQ, ALC262_FIXUP_BENQ_T31, @@ -4777,16 +4776,6 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, - [ALC262_FIXUP_TOSHIBA_RX1] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x90170110 }, /* speaker */ - { 0x15, 0x0421101f }, /* HP */ - { 0x1a, 0x40f000f0 }, /* N/A */ - { 0x1b, 0x40f000f0 }, /* N/A */ - { 0x1e, 0x40f000f0 }, /* N/A */ - } - }, [ALC262_FIXUP_LENOVO_3000] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -4819,8 +4808,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN), - SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", - ALC262_FIXUP_TOSHIBA_RX1), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ), -- 1.7.8.3

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[alsa-devel] [PATCH] ASoC: wm_hubs: Fix routing of input PGAs to line output mixer
by Mark Brown 31 Jan '12

31 Jan '12

IN1L/R is routed to both line output mixers, we don't route IN1 to LINEOUT1 and IN2 to LINEOUT2. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ea26724..c1a3f8c 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; @@ -848,8 +848,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = { }; static const struct snd_soc_dapm_route lineout2_diff_routes[] = { - { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, - { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, + { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" }, + { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" }, { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, -- 1.7.9.rc1

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[alsa-devel] [PATCH] ASoC: max9768: Fix up review comments in max9768 driver
by Mark Brown 31 Jan '12

31 Jan '12

The Mute Switch should be called Playback Switch to match the volume and the regmap core has been updated so we don't need to mark all the registers as non-volatile. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/max9768.c | 8 +------- 1 files changed, 1 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 6348995..79e9901 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -115,7 +115,7 @@ static const struct snd_kcontrol_new max9768_volume[] = { }; static const struct snd_kcontrol_new max9768_mute[] = { - SOC_SINGLE_BOOL_EXT("Mute Switch", 0, max9768_get_gpio, max9768_set_gpio), + SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio), }; static int max9768_probe(struct snd_soc_codec *codec) @@ -150,18 +150,12 @@ static struct snd_soc_codec_driver max9768_codec_driver = { .num_controls = ARRAY_SIZE(max9768_volume), }; -static bool max9768_always_false(struct device *dev, unsigned int reg) -{ - return false; -} - static const struct regmap_config max9768_i2c_regmap_config = { .reg_bits = 2, .val_bits = 6, .max_register = 3, .reg_defaults = max9768_default_regs, .num_reg_defaults = ARRAY_SIZE(max9768_default_regs), - .volatile_reg = max9768_always_false, .cache_type = REGCACHE_RBTREE, }; -- 1.7.9.rc1

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[alsa-devel] [PATCH 1/3] regmap: Properly round reg_bytes and val_bytes
by Wolfram Sang 30 Jan '12

30 Jan '12

For the upcoming 2/6-format, we don't see debugfs output otherwise, since the current division results in 0. I'd think 10/14 is broken currently, too. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> --- drivers/base/regmap/regmap.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 6555803..8bd0232 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -159,8 +159,8 @@ struct regmap *regmap_init(struct device *dev, mutex_init(&map->lock); map->format.buf_size = (config->reg_bits + config->val_bits) / 8; - map->format.reg_bytes = config->reg_bits / 8; - map->format.val_bytes = config->val_bits / 8; + map->format.reg_bytes = DIV_ROUND_UP(config->reg_bits, 8); + map->format.val_bytes = DIV_ROUND_UP(config->val_bits, 8); map->dev = dev; map->bus = bus; map->max_register = config->max_register; -- 1.7.8.3

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[alsa-devel] [PATCH -next] sound: fix include/sound/core.h warning
by Randy Dunlap 30 Jan '12

30 Jan '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix build warning by providing a struct stub since no fields of the struct are used: include/sound/core.h:454:56: warning: 'struct pci_dev' declared inside parameter list Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- include/sound/core.h | 2 ++ 1 file changed, 2 insertions(+) --- linux-next-20120130.orig/include/sound/core.h +++ linux-next-20120130/include/sound/core.h @@ -450,6 +450,8 @@ struct snd_pci_quirk { {_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)} #endif +struct pci_dev; + const struct snd_pci_quirk * snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list);

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[alsa-devel] ASoC updates for 3.3
by Mark Brown 30 Jan '12

30 Jan '12

The following changes since commit c83f1d7e71625801c72f4013291194e09b6f0a6e: ASoC: wm2000: Fix use-after-free - don't release_firmware() twice on error (2012-01-24 11:34:19 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-3.3 for you to fetch changes up to 1ae5cbc52e7c6619a3f44b87809fd25370df31bb: ASoC: neo1973_wm8753: remove references to the neo1973-gta01 machine (2012-01-30 13:22:13 +0000) ---------------------------------------------------------------- A bunch of small driver specific fixes - nothing terribly exciting here, all of this will only affect people using particular devices and then usually only in some use cases. ---------------------------------------------------------------- Denis 'GNUtoo' Carikli (1): ASoC: neo1973_wm8753: remove references to the neo1973-gta01 machine Mark Brown (6): ASoC: wm5100: Make sure we switch to button reporting mode ASoC: wm5100: Fix microphone configuration ASoC: wm8996: Mark register cache as dirty when regulators are disabled ASoC: wm8962: Mark register cache as dirty when regulators are disabled ASoC: wm5100: Mark register cache as dirty when regulators are disabled ASoC: wm_hubs: Enable line out VMID buffer for single ended line outputs UK KIM (1): ASoC: wm_hubs: fix wrong bits for LINEOUT2 N/P mixer sound/soc/codecs/wm5100.c | 12 ++++++- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm_hubs.c | 10 ++++- sound/soc/samsung/neo1973_wm8753.c | 65 +----------------------------------- 5 files changed, 22 insertions(+), 69 deletions(-)

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[alsa-devel] [PATCH 0/3] Support U8500/AB8500 ASoC
by Ola Lilja 30 Jan '12

30 Jan '12

First try to upload the ASoC-driver for ST-Ericsson U8500-platform. The audio system in the Ux500 platform consists of an external mixed signal ASIC called AB8500: http://www.stericsson.com/developers/CD00291561_UM1031_AB8500_user_manual-r… The audio portions of the AB8500 are configured over I2C and the raw data streams are sent using I2S/TDM. For this reason the I2S/TDM drivers for the DB8500 chip are included in this submission. Two of the patches mainly concern adding proper DMA channels and regulators to the Ux500 platform and may be treated individually, but preferably all patches shall be merged through the ASoC tree. Since we have more codecs for related platforms named U5500 (etc) the soc directory is name ux500. Ola Lilja (3): ARM: ux500: Add placeholder for clk_set_parent ARM: ux500: Add DMA-channels for MSP ASoC: Ux500: Add driver for Ux500/AB8500 platform/codec arch/arm/mach-ux500/Makefile | 3 +- arch/arm/mach-ux500/board-mop500-msp.c | 195 ++ arch/arm/mach-ux500/board-mop500-regulators.c | 28 + arch/arm/mach-ux500/board-mop500.c | 7 +- arch/arm/mach-ux500/board-mop500.h | 1 + arch/arm/mach-ux500/clock.c | 15 +- arch/arm/mach-ux500/clock.h | 1 + arch/arm/mach-ux500/devices-common.h | 2 +- arch/arm/mach-ux500/devices-db8500.c | 6 + arch/arm/mach-ux500/include/mach/hardware.h | 2 + arch/arm/mach-ux500/include/mach/msp.h | 1030 +++++++++ include/sound/ux500_ab8500.h | 36 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/Makefile.rej | 10 + sound/soc/codecs/ab8500_audio.c | 2928 +++++++++++++++++++++++++ sound/soc/codecs/ab8500_audio.h | 673 ++++++ sound/soc/ux500/Kconfig | 33 + sound/soc/ux500/Makefile | 25 + sound/soc/ux500/u8500.c | 150 ++ sound/soc/ux500/ux500_ab8500.c | 790 +++++++ sound/soc/ux500/ux500_msp_dai.c | 998 +++++++++ sound/soc/ux500/ux500_msp_dai.h | 83 + sound/soc/ux500/ux500_msp_i2s.c | 1004 +++++++++ sound/soc/ux500/ux500_msp_i2s.h | 40 + sound/soc/ux500/ux500_pcm.c | 428 ++++ sound/soc/ux500/ux500_pcm.h | 44 + 29 files changed, 8535 insertions(+), 9 deletions(-) create mode 100644 arch/arm/mach-ux500/board-mop500-msp.c create mode 100644 arch/arm/mach-ux500/include/mach/msp.h create mode 100644 include/sound/ux500_ab8500.h create mode 100644 sound/soc/codecs/Makefile.rej create mode 100644 sound/soc/codecs/ab8500_audio.c create mode 100644 sound/soc/codecs/ab8500_audio.h create mode 100644 sound/soc/ux500/Kconfig create mode 100644 sound/soc/ux500/Makefile create mode 100644 sound/soc/ux500/u8500.c create mode 100644 sound/soc/ux500/ux500_ab8500.c create mode 100644 sound/soc/ux500/ux500_msp_dai.c create mode 100644 sound/soc/ux500/ux500_msp_dai.h create mode 100644 sound/soc/ux500/ux500_msp_i2s.c create mode 100644 sound/soc/ux500/ux500_msp_i2s.h create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h -- 1.7.8.3

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[alsa-devel] [Patch v2] ASOC: neo1973_wm8753: remove references to the neo1973-gta01 machine
by Denis 'GNUtoo' Carikli 30 Jan '12

30 Jan '12

The Openmoko GTA01 machine has been removed from the machine ID database, so we need to remove references to it as well. Without that fix we have: sound/soc/samsung/neo1973_wm8753.c: In function ‘neo1973_wm8753_init’: sound/soc/samsung/neo1973_wm8753.c:325:2: error: implicit declaration of function ‘machine_is_neo1973_gta01’ Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo(a)no-log.org> --- sound/soc/samsung/neo1973_wm8753.c | 65 +----------------------------------- 1 files changed, 1 insertions(+), 64 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 1bf0c32..d23b19a 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -230,8 +230,6 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { /* GTA02 specific routes and controls */ -#ifdef CONFIG_MACH_NEO1973_GTA02 - static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -311,10 +309,6 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) return 0; } -#else -static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; } -#endif - static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; @@ -322,10 +316,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) int ret; /* set up NC codec pins */ - if (machine_is_neo1973_gta01()) { - snd_soc_dapm_nc_pin(dapm, "LOUT2"); - snd_soc_dapm_nc_pin(dapm, "ROUT2"); - } snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "OUT4"); snd_soc_dapm_nc_pin(dapm, "LINE1"); @@ -370,50 +360,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return 0; } -/* GTA01 specific controls */ - -#ifdef CONFIG_MACH_NEO1973_GTA01 - -static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = { - {"Amp IN", NULL, "ROUT1"}, - {"Amp IN", NULL, "LOUT1"}, - - {"Handset Spk", NULL, "Amp EP"}, - {"Stereo Out", NULL, "Amp LS"}, - {"Headphone", NULL, "Amp HP"}, -}; - -static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Handset Spk", NULL), - SND_SOC_DAPM_SPK("Stereo Out", NULL), - SND_SOC_DAPM_HP("Headphone", NULL), -}; - -static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) -{ - int ret; - - ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets, - ARRAY_SIZE(neo1973_lm4857_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes, - ARRAY_SIZE(neo1973_lm4857_routes)); - if (ret) - return ret; - - snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); - snd_soc_dapm_ignore_suspend(dapm, "Headphone"); - - return 0; -} - -#else -static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; }; -#endif - static struct snd_soc_dai_link neo1973_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", @@ -440,11 +386,6 @@ static struct snd_soc_aux_dev neo1973_aux_devs[] = { .name = "dfbmcs320", .codec_name = "dfbmcs320.0", }, - { - .name = "lm4857", - .codec_name = "lm4857.0-007c", - .init = neo1973_lm4857_init, - }, }; static struct snd_soc_codec_conf neo1973_codec_conf[] = { @@ -454,14 +395,10 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = { }, }; -#ifdef CONFIG_MACH_NEO1973_GTA02 static const struct gpio neo1973_gta02_gpios[] = { { GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, { GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, }; -#else -static const struct gpio neo1973_gta02_gpios[] = {}; -#endif static struct snd_soc_card neo1973 = { .name = "neo1973", @@ -480,7 +417,7 @@ static int __init neo1973_init(void) { int ret; - if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02()) + if (!machine_is_neo1973_gta02()) return -ENODEV; if (machine_is_neo1973_gta02()) { -- 1.7.4.1

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[alsa-devel] [PATCH v2] ALSA: Add #ifdef CONFIG_PCI guard for snd_pci_quirk_* functions
by Axel Lin 30 Jan '12

30 Jan '12

This fixes below build warning when CONFIG_PCI is not set. CC sound/sound_core.o In file included from sound/sound_core.c:15: include/sound/core.h:454: warning: 'struct pci_dev' declared inside parameter list include/sound/core.h:454: warning: its scope is only this definition or declaration, which is probably not what you want Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: begin the ifdef including the definition of struct snd_pci_quirk. include/sound/core.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 5ab255f..cea1b54 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -417,6 +417,7 @@ static inline int __snd_bug_on(int cond) #define gameport_get_port_data(gp) (gp)->port_data #endif +#ifdef CONFIG_PCI /* PCI quirk list helper */ struct snd_pci_quirk { unsigned short subvendor; /* PCI subvendor ID */ @@ -456,5 +457,6 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list); const struct snd_pci_quirk * snd_pci_quirk_lookup_id(u16 vendor, u16 device, const struct snd_pci_quirk *list); +#endif #endif /* __SOUND_CORE_H */ -- 1.7.5.4

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