Alsa-devel June 2014

alsa-devel@alsa-project.org

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250 discussions

[alsa-devel] [PATCH] sound: pci: ice1712: wm8776.c: Cleaning up uninitialized variables
by Rickard Strandqvist 01 Jun '14

01 Jun '14

There is a risk that the variable will be used without being initialized. This was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/ice1712/wm8776.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c index e66c0da..52da0ff 100644 --- a/sound/pci/ice1712/wm8776.c +++ b/sound/pci/ice1712/wm8776.c @@ -510,7 +510,7 @@ static int snd_wm8776_ctl_get(struct snd_kcontrol *kcontrol, { struct snd_wm8776 *wm = snd_kcontrol_chip(kcontrol); int n = kcontrol->private_value; - u16 val1, val2; + u16 val1 = 0, val2 = 0; if (wm->ctl[n].get) wm->ctl[n].get(wm, &val1, &val2); -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: ASoC: wm9713: correct mono out PGA sources
by Matt Reimer 01 Jun '14

01 Jun '14

The mono output PGA input only has four possible sources, so omit the rest. Signed-off-by: Matt Reimer <mreimer(a)sdgsystems.com> --- sound/soc/codecs/wm9713.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5f69b52..caf0269 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -74,8 +74,7 @@ static const char *wm9713_rec_src[] = "Mono Out", "Zh"}; static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; -static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv", - "Mono Vmid", "Inv Vmid"}; +static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv"}; static const char *wm9713_spk_pga[] = {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid", "Speaker Vmid", "Inv Vmid"}; -- 1.7.5.4

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[alsa-devel] [PATCH] sound: synth: emux: soundfont.c: Cleaning up memory leak
by Rickard Strandqvist 01 Jun '14

01 Jun '14

There is a risk for memory leak in when something unexpected happens and the function returns. This was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/synth/emux/soundfont.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 1137b85..09c5eeb 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -1021,6 +1021,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, data, count); if (rc < 0) { sf_sample_delete(sflist, sf, smp); + free(zone); return rc; } /* memory offset is updated after */ -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
by Alexander Shiyan 01 Jun '14

01 Jun '14

Eukrea-i.MX51 board was converted to use DT, ie we no longer have a MACH_EUKREA_MBIMXSD51_BASEBOARD symbol. Transformation of other boards planned for the near future, so this patch removes all these dependencies and restricts build of this driver to ARCH_MXC. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d262ec0..2989319 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -207,12 +207,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD \ - || MACH_EUKREA_MBIMXSD25_BASEBOARD \ - || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD \ - || (OF && ARM) - depends on I2C + depends on ARCH_MXC && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI -- 1.8.3.2

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[alsa-devel] [PATCH v5 0/8] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 01 Jun '14

01 Jun '14

Hi, These are the remaining patches of this fsl-ssi cleanup series. v5 adds the possibility to set the bitclock frequency using .set_dai_sysclk as suggested. If it is not explicitly set, the bitclock is calculated using the hw_params. I also added the missing signed-off-by tags to the patches. Best Regards, Markus Markus Pargmann (4): ASoC: fsl-ssi: make fsl,mode property optional ASoC: fsl-ssi: Fix baudclock handling ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Use regmap Sascha Hauer (4): ASoC: fsl-ssi: introduce SoC specific data ASoC: fsl-ssi: set bitclock in master mode from hw_params ASoC: fsl-ssi: remove unnecessary spinlock ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_ssi.c | 574 ++++++++++++++++++++++++++++-------------------- sound/soc/fsl/fsl_ssi.h | 50 ++--- 3 files changed, 360 insertions(+), 265 deletions(-) -- 2.0.0.rc2

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[alsa-devel] [PATCH 1/4] ASoC: Intel: Clear stored Baytrail DSP DMA pointer before stream start
by Jarkko Nikula 01 Jun '14

01 Jun '14

Stored DSP DMA pointer must be cleared before starting the stream since PCM pointer callback sst_byt_pcm_pointer() can be called before pointer is updated. In that case last position of previous stream was wronly returned. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 3af38576e91e..8eab97368ea7 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -180,6 +180,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: + pcm_data->hw_ptr = 0; sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: -- 2.0.0.rc2

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[alsa-devel] [PATCH v3] ASoC: cs42l56: Fix new value argument in snd_soc_update_bits calls
by Axel Lin 01 Jun '14

01 Jun '14

The new value argument needs proper shift to match the mask bit fields. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v3: Define CS42L56_MUTE_ALL as 0xff, so all the mask bits will be set. v2: Address Brian's comment in v1. sound/soc/codecs/cs42l56.c | 70 ++++++++++++++++++++-------------------------- sound/soc/codecs/cs42l56.h | 10 ++++--- 2 files changed, 37 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 5bb134b..fdc4bd2 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -763,14 +763,14 @@ static int cs42l56_set_sysclk(struct snd_soc_dai *codec_dai, case CS42L56_MCLK_11P2896MHZ: case CS42L56_MCLK_12MHZ: case CS42L56_MCLK_12P288MHZ: - cs42l56->mclk_div2 = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; cs42l56->mclk_prediv = 0; break; case CS42L56_MCLK_22P5792MHZ: case CS42L56_MCLK_24MHZ: case CS42L56_MCLK_24P576MHZ: - cs42l56->mclk_div2 = 1; - cs42l56->mclk_prediv = 1; + cs42l56->mclk_div2 = CS42L56_MCLK_DIV2; + cs42l56->mclk_prediv = CS42L56_MCLK_PREDIV; break; default: return -EINVAL; @@ -844,57 +844,49 @@ static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) /* Hit the DSP Mixer first */ snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_MUTE); + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_MUTE_ALL); /* Mute ADC's */ snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_MUTE); + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_MUTE_ALL); /* HP And LO */ snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_MUTE_ALL); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_MUTE); + CS42L56_HP_MUTE_MASK, CS42L56_MUTE_ALL); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); + CS42L56_LO_MUTE_MASK, CS42L56_MUTE_ALL); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_MUTE); - - + CS42L56_LO_MUTE_MASK, CS42L56_MUTE_ALL); } else { snd_soc_update_bits(codec, CS42L56_DSP_MUTE_CTL, CS42L56_ADCAMIX_MUTE_MASK | - CS42L56_ADCBMIX_MUTE_MASK | - CS42L56_PCMAMIX_MUTE_MASK | - CS42L56_PCMBMIX_MUTE_MASK | - CS42L56_MSTB_MUTE_MASK | - CS42L56_MSTA_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_ADCBMIX_MUTE_MASK | + CS42L56_PCMAMIX_MUTE_MASK | + CS42L56_PCMBMIX_MUTE_MASK | + CS42L56_MSTB_MUTE_MASK | + CS42L56_MSTA_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_MISC_ADC_CTL, - CS42L56_ADCA_MUTE_MASK | - CS42L56_ADCB_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_ADCA_MUTE_MASK | + CS42L56_ADCB_MUTE_MASK, + CS42L56_UNMUTE); + snd_soc_update_bits(codec, CS42L56_HPA_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_HPB_VOLUME, - CS42L56_HP_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_HP_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_LOA_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); snd_soc_update_bits(codec, CS42L56_LOB_VOLUME, - CS42L56_LO_MUTE_MASK, - CS42L56_UNMUTE); + CS42L56_LO_MUTE_MASK, CS42L56_UNMUTE); } return 0; } diff --git a/sound/soc/codecs/cs42l56.h b/sound/soc/codecs/cs42l56.h index ad2b50a..5025ec9 100644 --- a/sound/soc/codecs/cs42l56.h +++ b/sound/soc/codecs/cs42l56.h @@ -80,19 +80,21 @@ #define CS42L56_PDN_HPB_MASK 0xc0 /* serial port and clk masks */ -#define CS42L56_MASTER_MODE 1 +#define CS42L56_MASTER_MODE 0x40 #define CS42L56_SLAVE_MODE 0 #define CS42L56_MS_MODE_MASK 0x40 -#define CS42L56_SCLK_INV 1 +#define CS42L56_SCLK_INV 0x20 #define CS42L56_SCLK_INV_MASK 0x20 #define CS42L56_SCLK_MCLK_MASK 0x18 +#define CS42L56_MCLK_PREDIV 0x04 #define CS42L56_MCLK_PREDIV_MASK 0x04 +#define CS42L56_MCLK_DIV2 0x02 #define CS42L56_MCLK_DIV2_MASK 0x02 #define CS42L56_MCLK_DIS_MASK 0x01 #define CS42L56_CLK_AUTO_MASK 0x20 #define CS42L56_CLK_RATIO_MASK 0x1f #define CS42L56_DIG_FMT_I2S 0 -#define CS42L56_DIG_FMT_LEFT_J 1 +#define CS42L56_DIG_FMT_LEFT_J 0x08 #define CS42L56_DIG_FMT_MASK 0x08 /* Class H and misc ctl masks */ @@ -116,7 +118,7 @@ #define CS42L56_DEEMPH_MASK 0x40 #define CS42L56_PLYBCK_GANG_MASK 0x10 #define CS42L56_PCM_INV_MASK 0x0c -#define CS42L56_MUTE 1 +#define CS42L56_MUTE_ALL 0xff #define CS42L56_UNMUTE 0 #define CS42L56_ADCAMIX_MUTE_MASK 0x40 #define CS42L56_ADCBMIX_MUTE_MASK 0x80 -- 1.8.3.2

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[alsa-devel] [PATCHv2] ASoC: Intel: Add Baytrail byt-max98090 machine driver
by Jarkko Nikula 01 Jun '14

01 Jun '14

Add machine driver and ACPI probing for Baytrail SST with MAX98090 codec. Jack detect code from Kevin Strasser <kevin.strasser(a)intel.com>, GPIO resolving from Mika Westerberg <mika.westerberg(a)linux.intel.com> and fixes and cleanups from Liam Girdwood <liam.r.girdwood(a)linux.intel.com>. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- This is for for-next branch since depends on commit f025d3b9c64e("ASoC: jack: Add support for GPIO descriptor defined jack pins"). Or merge topic/gpio or for-next to topic/intel before applying. v2: - Needless DAPM calls removed from init - Using snd_soc_pm_ops for suspend/resume - Using new GPIO descriptors support from ASoC jack - devm_snd_soc_register_card - Bogus Kconfig symbol removed from Makefile - Various cleanups --- sound/soc/intel/Kconfig | 9 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/byt-max98090.c | 203 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-acpi.c | 1 + 4 files changed, 215 insertions(+) create mode 100644 sound/soc/intel/byt-max98090.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 3c81b3891209..cd3498736e91 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -49,3 +49,12 @@ config SND_SOC_INTEL_BYT_RT5640_MACH help This adds audio driver for Intel Baytrail platform based boards with the RT5640 audio codec. + +config SND_SOC_INTEL_BYT_MAX98090_MACH + tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_BAYTRAIL + select SND_SOC_MAX98090 + help + This adds audio driver for Intel Baytrail platform based boards + with the MAX98090 audio codec. diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 0db4e2f336dc..4bfca79a42ba 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -23,6 +23,8 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o # Machine support snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o +snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c new file mode 100644 index 000000000000..5fc98c64a3f4 --- /dev/null +++ b/sound/soc/intel/byt-max98090.c @@ -0,0 +1,203 @@ +/* + * Intel Baytrail SST MAX98090 machine driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/max98090.h" + +struct byt_max98090_private { + struct snd_soc_jack jack; +}; + +static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { + {"IN34", NULL, "Headset Mic"}, + {"IN34", NULL, "MICBIAS"}, + {"MICBIAS", NULL, "Headset Mic"}, + {"DMICL", NULL, "Int Mic"}, + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Ext Spk", NULL, "SPKL"}, + {"Ext Spk", NULL, "SPKR"}, +}; + +static const struct snd_kcontrol_new byt_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Ext Spk", + .mask = SND_JACK_LINEOUT, + }, + { + .pin = "Int Mic", + .mask = SND_JACK_LINEIN, + }, +}; + +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .name = "hp-gpio", + .idx = 0, + .report = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, + .debounce_time = 200, + }, + { + .name = "mic-gpio", + .idx = 1, + .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN, + .debounce_time = 200, + }, +}; + +static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); + struct snd_soc_jack *jack = &drv->jack; + + card->dapm.idle_bias_off = true; + + ret = snd_soc_dai_set_sysclk(runtime->codec_dai, + M98090_REG_SYSTEM_CLOCK, + 25000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Can't set codec clock %d\n", ret); + return ret; + } + + /* Enable jack detection */ + ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpiods(card->dev->parent, jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + if (ret) + return ret; + + return max98090_mic_detect(codec, jack); +} + +static struct snd_soc_dai_link byt_max98090_dais[] = { + { + .name = "Baytrail Audio", + .stream_name = "Audio", + .cpu_dai_name = "baytrail-pcm-audio", + .codec_dai_name = "HiFi", + .codec_name = "i2c-193C9890:00", + .platform_name = "baytrail-pcm-audio", + .init = byt_max98090_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card byt_max98090_card = { + .name = "byt-max98090", + .dai_link = byt_max98090_dais, + .num_links = ARRAY_SIZE(byt_max98090_dais), + .dapm_widgets = byt_max98090_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_max98090_widgets), + .dapm_routes = byt_max98090_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), + .controls = byt_max98090_controls, + .num_controls = ARRAY_SIZE(byt_max98090_controls), +}; + +static int byt_max98090_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct byt_max98090_private *priv; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); + if (!priv) { + dev_err(&pdev->dev, "allocation failed\n"); + return -ENOMEM; + } + + byt_max98090_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&byt_max98090_card, priv); + ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_max98090_card); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + + return ret_val; +} + +static int byt_max98090_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct byt_max98090_private *priv = snd_soc_card_get_drvdata(card); + + snd_soc_jack_free_gpios(&priv->jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return 0; +} + +static struct platform_driver byt_max98090_driver = { + .probe = byt_max98090_probe, + .remove = byt_max98090_remove, + .driver = { + .name = "byt-max98090", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(byt_max98090_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); +MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:byt-max98090"); diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 18aee77f8d4a..42edc6f4fc4a 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -247,6 +247,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { static struct sst_acpi_mach baytrail_machines[] = { { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" }, {} }; -- 2.0.0.rc2

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[alsa-devel] [PATCH] ASoC: Intel: byt/hsw: Add missing kthread_stop to error/cleanup path
by Jarkko Nikula 01 Jun '14

01 Jun '14

From: Imre Deak <imre.deak(a)intel.com> Baytrail and Haswell SST IPC don't stop the kernel thread in error and cleanup path thus leaving orphan kernel thread behind in such a case. Also while at it, fix one error path in sst-haswell-ipc.c that doesn't free hsw->msg. [Jarkko: I edited the commit log a little] Signed-off-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-ipc.c | 5 ++++- sound/soc/intel/sst-haswell-ipc.c | 12 ++++++++---- 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 0d31dbbf4806..d4b8c827c8b8 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -824,7 +824,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) byt->dsp = sst_dsp_new(dev, &byt_dev, pdata); if (byt->dsp == NULL) { err = -ENODEV; - goto err_free_msg; + goto dsp_err; } /* keep the DSP in reset state for base FW loading */ @@ -856,6 +856,8 @@ boot_err: sst_fw_free(byt_sst_fw); fw_err: sst_dsp_free(byt->dsp); +dsp_err: + kthread_stop(byt->tx_thread); err_free_msg: kfree(byt->msg); @@ -870,6 +872,7 @@ void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata) sst_dsp_reset(byt->dsp); sst_fw_free_all(byt->dsp); sst_dsp_free(byt->dsp); + kthread_stop(byt->tx_thread); kfree(byt->msg); } EXPORT_SYMBOL_GPL(sst_byt_dsp_free); diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index e7996b39a484..bee192d42032 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1730,7 +1730,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) ret = msg_empty_list_init(hsw); if (ret < 0) - goto list_err; + return -ENOMEM; /* start the IPC message thread */ init_kthread_worker(&hsw->kworker); @@ -1740,7 +1740,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) if (IS_ERR(hsw->tx_thread)) { ret = PTR_ERR(hsw->tx_thread); dev_err(hsw->dev, "error: failed to create message TX task\n"); - goto list_err; + goto err_free_msg; } init_kthread_work(&hsw->kwork, ipc_tx_msgs); @@ -1750,7 +1750,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) hsw->dsp = sst_dsp_new(dev, &hsw_dev, pdata); if (hsw->dsp == NULL) { ret = -ENODEV; - goto list_err; + goto dsp_err; } /* keep the DSP in reset state for base FW loading */ @@ -1794,8 +1794,11 @@ boot_err: sst_fw_free(hsw_sst_fw); fw_err: sst_dsp_free(hsw->dsp); +dsp_err: + kthread_stop(hsw->tx_thread); +err_free_msg: kfree(hsw->msg); -list_err: + return ret; } EXPORT_SYMBOL_GPL(sst_hsw_dsp_init); @@ -1808,6 +1811,7 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) sst_fw_free_all(hsw->dsp); sst_dsp_free(hsw->dsp); kfree(hsw->scratch); + kthread_stop(hsw->tx_thread); kfree(hsw->msg); } EXPORT_SYMBOL_GPL(sst_hsw_dsp_free); -- 2.0.0.rc2

2 1

[alsa-devel] [PATCH] ASoC: tlv320aci3x: Fix custom snd_soc_dapm_put_volsw_aic3x() function
by Peter Ujfalusi 01 Jun '14

01 Jun '14

For some unknown reason the parameters for snd_soc_test_bits() were in wrong order: It was: snd_soc_test_bits(codec, val, mask, reg); /* WRONG!!! */ while it should be: snd_soc_test_bits(codec, reg, mask, val); Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, so the failures I was seeing was indeed caused by something one never expects: wrong parameter order for snd_soc_test_bits() in the tlv320aic3x driver, so regmap failed time to time when we asked it to read a register out of the valid range, since the value was sent as register number :o Regards, Peter sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d7349bc89ad3..e12fafbb1e09 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -169,7 +169,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, mask <<= shift; val <<= shift; - change = snd_soc_test_bits(codec, val, mask, reg); + change = snd_soc_test_bits(codec, reg, mask, val); if (change) { update.kcontrol = kcontrol; update.reg = reg; -- 1.9.3

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Alsa-devel June 2014