December 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Add trivial pm_runtime usage to Samsung DAI drivers
by Mark Brown 12 Dec '11

12 Dec '11

Currently this won't actually do anything but using this will help the core SoC code track when the system is idle. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/i2s.c | 6 +++++- sound/soc/samsung/pcm.c | 5 +++++ 2 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ff5d919..87a874d 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -15,6 +15,7 @@ #include <linux/clk.h> #include <linux/io.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -1095,6 +1096,8 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv); + pm_runtime_enable(&pdev->dev); + return 0; err: release_mem_region(regs_base, resource_size(res)); @@ -1105,6 +1108,7 @@ err: static __devexit int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; + struct resource *res; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; @@ -1113,7 +1117,7 @@ static __devexit int samsung_i2s_remove(struct platform_device *pdev) other->pri_dai = NULL; other->sec_dai = NULL; } else { - struct resource *res; + pm_runtime_disable(&pdev->dev); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res) release_mem_region(res->start, resource_size(res)); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 3a29c26..a1dc141c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -14,6 +14,7 @@ #include <linux/clk.h> #include <linux/io.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -581,6 +582,8 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + pm_runtime_enable(&pdev->dev); + ret = snd_soc_register_dai(&pdev->dev, &s3c_pcm_dai[pdev->id]); if (ret != 0) { dev_err(&pdev->dev, "failed to get register DAI: %d\n", ret); @@ -610,6 +613,8 @@ static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) snd_soc_unregister_dai(&pdev->dev); + pm_runtime_disable(&pdev->dev); + iounmap(pcm->regs); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); -- 1.7.7.3

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[alsa-devel] [PATCH] ALSA: HDA: Realtek: Take vmaster dac from multiout dac list
by David Henningsson 12 Dec '11

12 Dec '11

With the auto-parser we can choose the dac nid for vmaster from the DACs we already know, instead of hard-coding it. This is more future-proof and was actually wrong on one machine. Reported-by: David Chen <david.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 32 ++++++++++---------------------- 1 files changed, 10 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a74c1e..31dd675 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3153,6 +3153,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) } } + spec->vmaster_nid = spec->multiout.dac_nids[0]; return 0; } @@ -4175,8 +4176,10 @@ static int patch_alc880(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { + spec->vmaster_nid = 0x0c; setup_preset(codec, &alc880_presets[board_config]); + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4196,8 +4199,6 @@ static int patch_alc880(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4304,8 +4305,10 @@ static int patch_alc260(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc260_presets[board_config]); + spec->vmaster_nid = 0x08; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4325,8 +4328,6 @@ static int patch_alc260(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x08; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4698,8 +4699,10 @@ static int patch_alc882(struct hda_codec *codec) goto error; } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc882_presets[board_config]); + spec->vmaster_nid = 0x0c; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4719,8 +4722,6 @@ static int patch_alc882(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4899,8 +4900,6 @@ static int patch_alc262(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5012,8 +5011,6 @@ static int patch_alc268(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5568,8 +5565,6 @@ static int patch_alc269(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; @@ -5674,8 +5669,6 @@ static int patch_alc861(struct hda_codec *codec) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } - spec->vmaster_nid = 0x03; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -5800,8 +5793,6 @@ static int patch_alc861vd(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - spec->vmaster_nid = 0x02; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -6184,7 +6175,6 @@ static int patch_alc662(struct hda_codec *codec) break; } } - spec->vmaster_nid = 0x02; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6240,8 +6230,6 @@ static int patch_alc680(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Add missed MODULE_LICENSE("GPL") for imx-pcm-fiq
by Axel Lin 12 Dec '11

12 Dec '11

This driver can be built as module and the file header indicates that the driver is published under the GPL. Thus add MODULE_LICENSE("GPL") for it. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/imx-pcm-fiq.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index d7ea0b3..456b7d7 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -332,3 +332,5 @@ static struct platform_driver imx_pcm_driver = { }; module_platform_driver(imx_pcm_driver); + +MODULE_LICENSE("GPL"); -- 1.7.5.4

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[alsa-devel] [PATCH v2] Export symbols 'mxs_saif_put_mclk()' and 'mxs_saif_get_mclk()'
by Lothar Waßmann 12 Dec '11

12 Dec '11

Fixes the following errors with MXS sound support built as module: |ERROR: "mxs_saif_put_mclk" [sound/soc/mxs/snd-soc-mxs-sgtl5000.ko] undefined! |ERROR: "mxs_saif_get_mclk" [sound/soc/mxs/snd-soc-mxs-sgtl5000.ko] undefined! Signed-off-by: Lothar Waßmann <LW(a)KARO-electronics.de> --- sound/soc/mxs/mxs-saif.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 487d2ff..ec66a29 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -221,6 +221,7 @@ int mxs_saif_put_mclk(unsigned int saif_id) saif->mclk_in_use = 0; return 0; } +EXPORT_SYMBOL_GPL(mxs_saif_put_mclk); /* * Get MCLK and set clock rate, then enable it @@ -274,7 +275,7 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, return 0; } +EXPORT_SYMBOL_GPL(mxs_saif_get_mclk); /* * SAIF DAI format configuration. -- 1.5.6.5

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[alsa-devel] [PATCH 0/5] ASoC: mxs: Several cleanups and bugfixes for the MXS sound driver
by Lothar Waßmann 12 Dec '11

12 Dec '11

Lothar Waßmann (5): General cleanups Add .owner to struct snd_soc_card Export symbols 'mxs_saif_put_mclk()' and 'mxs_saif_get_mclk()' Add missing MODULE_LICENSE("GPL") Add appropriate MODULE_ALIAS() sound/soc/codecs/sgtl5000.c | 13 ++++++------- sound/soc/mxs/mxs-pcm.c | 3 +++ sound/soc/mxs/mxs-saif.c | 4 +++- sound/soc/mxs/mxs-sgtl5000.c | 2 ++ 4 files changed, 14 insertions(+), 8 deletions(-)

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[alsa-devel] [PATCH] ALSA: HDA: Set position fix to LPIB for an Atom/Poulsbo based device
by David Henningsson 12 Dec '11

12 Dec '11

For the Asus 1101HA, reporting position by reading the DMA position buffer map seems unstable and often wrong. The reporter says that position_fix=LPIB works much better (although not 100%, but this is probably due to other issues). The controller chip is an Intel Poulsbo 8086:811b (rev 07) controller, and complete alsa-info is available here: https://launchpadlibrarian.net/86691768/alsa-info.txt.1TNwyE5Ea7 Cc: stable(a)kernel.org (3.0+) BugLink: http://bugs.launchpad.net/bugs/825709 Tested-by: Stefano Lodi Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d1582dd..0746ab4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2506,6 +2506,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS 1101HA", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), -- 1.7.5.4

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[alsa-devel] [linux-next] ASoC: build errors if CONFIG_SND_SOC=m && CONFIG_SND_SOC_ALL_CODECS=m
by Axel Lin 12 Dec '11

12 Dec '11

hi, I got below build errors if CONFIG_SND_SOC=m && CONFIG_SND_SOC_ALL_CODECS=m. ( build test on linux-next 20111212 ). Kernel: arch/arm/boot/zImage is ready Building modules, stage 2. MODPOST 87 modules ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8974.ko] undefined! ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8940.ko] undefined! ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8753.ko] undefined! ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-wm8510.ko] undefined! make[1]: *** [__modpost] Error 1 make: *** [modules] Error 2 Here is my gcc version: $ arm-eabi-gcc -v Using built-in specs. Target: arm-eabi Configured with: /home/jingyu/projects/gcc/android-toolchainsrc/build/../gcc/gcc-4.4.3/configure --prefix=/usr/local --target=arm-eabi --host=x86_64-linux-gnu --build=x86_64-linux-gnu --with-gnu-as --with-gnu-ld --enable-languages=c,c++ --with-gmp=/home/jingyu/projects/gcc/toolchain_build/obj/temp-install --with-mpfr=/home/jingyu/projects/gcc/toolchain_build/obj/temp-install --disable-libssp --enable-threads --disable-nls --disable-libmudflap --disable-libgomp --disable-libstdc__-v3 --disable-sjlj-exceptions --disable-shared --disable-tls --with-float=soft --with-fpu=vfp --with-arch=armv5te --enable-target-optspace --with-abi=aapcs --with-gcc-version=4.4.3 --with-binutils-version=2.19 --with-gmp-version=4.2.4 --with-mpfr-version=2.4.1 --with-gdb-version=7.1.x --with-arch=armv5te --with-multilib-list=mandroid --with-sysroot=/usr/local/google/home/android/cupcake_rel_root --program-transform-name='s&^&arm-eabi-&' Thread model: single gcc version 4.4.3 (GCC) Regards, Axel

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[alsa-devel] [PATCH v3] sound/soc/lapis: add machine driver v3 updated
by Tomoya MORINAGA 12 Dec '11

12 Dec '11

Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- V3 Update Mark's comments - Delete i2c_board_info data structure. - Delete i2c_new_device/i2c_put_adapter. - Delete .startup method. - Add error processing for failure of snd_soc_unregister_card. - Add MODULE_ALIAS - Change platform_driver name - Set to .dai_fmt of snd_soc_dai_link not use snd_soc_dai_set_fmt --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/lapis/Kconfig | 5 + sound/soc/lapis/Makefile | 5 + sound/soc/lapis/ml7213ioh-machine.c | 158 +++++++++++++++++++++++++++++++++++ 5 files changed, 170 insertions(+), 0 deletions(-) create mode 100644 sound/soc/lapis/Kconfig create mode 100644 sound/soc/lapis/Makefile create mode 100644 sound/soc/lapis/ml7213ioh-machine.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 35e662d..63f8582 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -34,6 +34,7 @@ source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/imx/Kconfig" source "sound/soc/jz4740/Kconfig" +source "sound/soc/lapis/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 9ea8ac8..002951f 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -11,6 +11,7 @@ obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += jz4740/ +obj-$(CONFIG_SND_SOC) += lapis/ obj-$(CONFIG_SND_SOC) += mid-x86/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ diff --git a/sound/soc/lapis/Kconfig b/sound/soc/lapis/Kconfig new file mode 100644 index 0000000..f6c2fe5 --- /dev/null +++ b/sound/soc/lapis/Kconfig @@ -0,0 +1,5 @@ +config SND_SOC_ML7213_MACHINE + tristate "ML7213 IOH ASoC machine driver" + select SND_SOC_ML7213_PLATFORM + help + This is ASoC machine driver for ML7213 IOH diff --git a/sound/soc/lapis/Makefile b/sound/soc/lapis/Makefile new file mode 100644 index 0000000..3560bd7 --- /dev/null +++ b/sound/soc/lapis/Makefile @@ -0,0 +1,5 @@ +# Platform +snd-soc-ml7213-machine-objs := ml7213ioh-machine.o + +obj-$(CONFIG_SND_SOC_ML7213_PLATFORM) += snd-soc-ml7213-machine.o + diff --git a/sound/soc/lapis/ml7213ioh-machine.c b/sound/soc/lapis/ml7213ioh-machine.c new file mode 100644 index 0000000..28e7919 --- /dev/null +++ b/sound/soc/lapis/ml7213ioh-machine.c @@ -0,0 +1,158 @@ +/* + * ml7213ioh-machine.c -- SoC Audio for LAPIS Semiconductor ML7213 IOH CRB + * + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <linux/i2c.h> +#include "../codecs/ml26124.h" + +static const struct snd_soc_dapm_widget ml7213_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_LINE("LINEOUT", NULL), + SND_SOC_DAPM_LINE("LINEIN", NULL), + SND_SOC_DAPM_MIC("AnalogMIC", NULL), + SND_SOC_DAPM_MIC("DigitalMIC", NULL), +}; + +static const struct snd_soc_dapm_route ml7213_routes[] = { + {"Speaker", NULL, "SPOUT"}, + {"LINEOUT", NULL, "LOUT"}, + {"MIN", NULL, "AnalogMIC"}, + {"MDIN", NULL, "DigitalMIC"}, + {"MIN", NULL, "LINEIN"}, +}; + +static int ml7213_ioh_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, ml7213_dapm_widgets, + ARRAY_SIZE(ml7213_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, ml7213_routes, ARRAY_SIZE(ml7213_routes)); + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int ml7213_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 16000: + case 32000: + case 48000: + clk = 12288800; + break; + default: + return -EINVAL; + } + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, ML26124_USE_PLL, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops ml7213_ops = { + .hw_params = ml7213_hw_params, +}; + +static struct snd_soc_dai_link ioh_i2s_dai = { + .name = "ML7213", + .stream_name = "I2S HiFi", + .cpu_dai_name = "ml7213-i2s", + .codec_dai_name = "ml26124-hifi", + .platform_name = "ml7213-i2s-pcm-audio", + .codec_name = "ml26124-00GD", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .init = ml7213_ioh_init, + .ops = &ml7213_ops, +}; + +static struct snd_soc_card ioh_i2s_card = { + .name = "ml7213", + .dai_link = &ioh_i2s_dai, + .num_links = 1, + .dapm_widgets = ml7213_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml7213_dapm_widgets), + .dapm_routes = ml7213_routes, + .num_dapm_routes = ARRAY_SIZE(ml7213_routes), +}; + +static int __init ioh_i2s_probe(struct platform_device *pdev) +{ + ioh_i2s_card.dev = &pdev->dev; + snd_soc_register_card(&ioh_i2s_card); + + return 0; +} + +static int __exit ioh_i2s_remove(struct platform_device *pdev) +{ + int ret = snd_soc_unregister_card(&ioh_i2s_card); + + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + ioh_i2s_card.dev = NULL; + return ret; + } + + return 0; +} + +static struct platform_driver ioh_i2s_driver = { + .remove = ioh_i2s_remove, + .driver = { + .name = "ml7213ioh-i2s", + .owner = THIS_MODULE, + }, + .probe = ioh_i2s_probe, + .remove = __devexit_p(ioh_i2s_remove), +}; + +static int __init ioh_i2s_init(void) +{ + return platform_driver_register(&ioh_i2s_driver); +} + +static void __exit ioh_i2s_exit(void) +{ + return platform_driver_unregister(&ioh_i2s_driver); +} + +module_init(ioh_i2s_init); +module_exit(ioh_i2s_exit); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML7213 IOH ALSA SoC machine driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ml7213ioh-i2s"); -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: Fix hx4700 error handling to free gpios if snd_soc_register_card fails
by Axel Lin 12 Dec '11

12 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/hx4700.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index e32afaf..03ef9f3 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -209,9 +209,10 @@ static int __devinit hx4700_audio_probe(struct platform_device *pdev) snd_soc_card_hx4700.dev = &pdev->dev; ret = snd_soc_register_card(&snd_soc_card_hx4700); if (ret) - return ret; + gpio_free_array(hx4700_audio_gpios, + ARRAY_SIZE(hx4700_audio_gpios)); - return 0; + return ret; } static int __devexit hx4700_audio_remove(struct platform_device *pdev) -- 1.7.5.4

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Re: [alsa-devel] [PATCH] ASoC: WM8958: correctly show firmware magic on mismatch
by Mark Brown 12 Dec '11

12 Dec '11

On Sun, Dec 11, 2011 at 07:15:49PM +0100, Jonathan Neuschäfer wrote: > Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> Applied, thanks.

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