March 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: Intel: add nau8824 sound card" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: add nau8824 sound card has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5b00ce63e07fac692400db80d3e1fbd1f1f65701 Mon Sep 17 00:00:00 2001 From: John Hsu <KCHSU0(a)nuvoton.com> Date: Tue, 6 Feb 2018 15:02:43 +0800 Subject: [PATCH] ASoC: Intel: add nau8824 sound card Create sound card with NAU88L24 codec in Intel Cherryview-based platforms, Cherrytrail and Braswell. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cht_bsw_nau8824.c | 282 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cht-match.c | 9 + 4 files changed, 304 insertions(+) create mode 100644 sound/soc/intel/boards/cht_bsw_nau8824.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index eedc83b96fbb..24797482a3d2 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -125,6 +125,17 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_CHT_BSW_NAU8824_MACH + tristate "Cherrytrail & Braswell with NAU88L24 codec" + depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ACPI + select SND_SOC_NAU8824 + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with NAU88L24 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_BYT_CHT_DA7213_MACH tristate "Baytrail & Cherrytrail with DA7212/7213 codec" depends on X86_INTEL_LPSS && I2C && ACPI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 6fae506c40fa..92b5507291af 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -11,6 +11,7 @@ snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-sst-cht-bsw-nau8824-objs := cht_bsw_nau8824.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o @@ -33,6 +34,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_NAU8824_MACH) += snd-soc-sst-cht-bsw-nau8824.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c new file mode 100644 index 000000000000..08d0b8128fab --- /dev/null +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -0,0 +1,282 @@ +/* + * cht-bsw-nau8824.c - ASoc Machine driver for Intel Cherryview-based + * platforms Cherrytrail and Braswell, with nau8824 codec. + * + * Copyright (C) 2018 Intel Corp + * Copyright (C) 2018 Nuvoton Technology Corp + * + * Author: Wang, Joseph C <joequant(a)gmail.com> + * Co-author: John Hsu <KCHSU0(a)nuvoton.com> + * This file is based on cht_bsw_rt5672.c and cht-bsw-max98090.c + * + * 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. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <linux/input.h> +#include "../atom/sst-atom-controls.h" +#include "../../codecs/nau8824.h" + +struct cht_mc_private { + struct snd_soc_jack jack; +}; + +static struct snd_soc_jack_pin cht_bsw_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static const struct snd_soc_dapm_widget cht_dapm_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 cht_audio_map[] = { + {"Ext Spk", NULL, "SPKOUTL"}, + {"Ext Spk", NULL, "SPKOUTR"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"MIC1", NULL, "Int Mic"}, + {"MIC2", NULL, "Int Mic"}, + {"HSMIC1", NULL, "Headset Mic"}, + {"HSMIC2", NULL, "Headset Mic"}, + {"Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_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 int cht_aif1_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; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8824_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + return ret; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, params_rate(params), + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + struct snd_soc_jack *jack = &ctx->jack; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + int ret, jack_type; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* NAU88L24 supports 4 butons headset detection + * KEY_MEDIA + * KEY_VOICECOMMAND + * KEY_VOLUMEUP + * KEY_VOLUMEDOWN + */ + jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, + cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); + if (ret) { + dev_err(runtime->dev, + "Headset Jack creation failed %d\n", ret); + return ret; + } + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + nau8824_enable_jack_detect(codec, jack); + + return ret; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = + hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_none(fmt); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 48000); +} + +static const struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static const struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_DEEP_BUFFER] = { + .name = "Deep-Buffer Audio Port", + .stream_name = "Deep-Buffer Audio", + .cpu_dai_name = "deepbuffer-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = NAU8824_CODEC_DAI, + .codec_name = "i2c-10508824:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "chtnau8824", + .owner = THIS_MODULE, + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + struct cht_mc_private *drv; + int ret_val; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-bsw-nau8824", + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Wang, Joseph C <joequant(a)gmail.com>"); +MODULE_AUTHOR("John Hsu <KCHSU0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-nau8824"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index b50a0d53846b..ad1eb2d644be 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -117,6 +117,15 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .sof_tplg_filename = "intel/reef-cht-max98090.tplg", .asoc_plat_name = "sst-mfld-platform", }, + { + .id = "10508824", + .drv_name = "cht-bsw-nau8824", + .fw_filename = "intel/fw_sst_22a8.bin", + .board = "cht-bsw", + .sof_fw_filename = "intel/reef-cht.ri", + .sof_tplg_filename = "intel/reef-cht-nau8824.tplg", + .asoc_plat_name = "sst-mfld-platform", + }, { .id = "DLGS7212", .drv_name = "bytcht_da7213", -- 2.16.2

1 0

[alsa-devel] [RFC 0/7] ASoC: rt5651: Drop pdata, move over to device-properties
by Hans de Goede 01 Mar '18

01 Mar '18

Hi Mark, et al., Here is a second attempt to stop having duplicated DMI quirk entries between the rt5651 codec and machine drivers. Note this is only compile tested, I'm looking for some quick feedback if this seems to be going in the right direction, as I'm planning on rebuilding the rest of my jack-detect cleanup / fix series on top of this. The diffstat for this entire set is: Documentation/devicetree/bindings/sound/rt5651.txt | 6 include/sound/rt5651.h | 12 sound/soc/codecs/rt5651.c | 375 +++++++++------------ sound/soc/codecs/rt5651.h | 7 sound/soc/intel/boards/bytcr_rt5651.c | 46 ++ 5 files changed, 213 insertions(+), 233 deletions(-) So besides making the code cleaner (IMHO) it also makes it somewhat smaller and as an added bonus it fixes support the DMIC_EN quirk which was broken. Regards, Hans

2 9

[alsa-devel] [PATCH 0/7] Let Atmel use simple-audio-card
by Ladislav Michl 01 Mar '18

01 Mar '18

Alexandre, following patchset is sent as a bare minimum I need to use MAX9867 codec in master mode on AT91SAM9G20 based board. Codec's MCLK is driven by PCK0 and SSC is wired as follows: RK0 <-- BCLK RF0 <-- LRCLK TD0 --> SDIN RD0 <-- SDOUT Following DT framgment was used: i2c-gpio-0 { codec: max9867@18 { compatible = "maxim,max9867"; reg = <0x18>; #sound-dai-cells = <0>; }; }; sound { compatible = "simple-audio-card"; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_pck0_as_mck>; simple-audio-card,name = "simear4-max9867"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&codec_dai>; simple-audio-card,frame-master = <&codec_dai>; simple-audio-card,routing = "LINE_IN", "Line In Jack", "Speaker", "HPOUT"; simple-audio-card,widgets = "Line", "Line In Jack", "Speaker", "Speaker"; cpu_dai: simple-audio-card,cpu { sound-dai = <&ssc0>; }; codec_dai: simple-audio-card,codec { sound-dai = <&codec>; clocks = <&pck0>; clock-frequency = <12384000>; }; }; &ssc0 { status = "okay"; pinctrl-0 = <&pinctrl_board_ssc0_tx &pinctrl_ssc0_rx>; atmel,clk-from-rk-pin; #sound-dai-cells = <0>; }; Codec code itself seems to be unused since it was merged and I'm wondering whenever it worked at all. A comment from codec author would help a lot, just to be sure I did not break anything. By no means this code is ready to be merged (only except those patches with commit log and after carefull consideration :)), I'm posting it as a base for discussion. Thank you, ladis Ladislav Michl (7): ASoC: atmel: Remove redundant dev_err() call in probe function ASoC: atmel_ssc_dai: Fix TCMR settings in I2S slave mode ASoC: simple_card_utils: Set clock frequency ASoC: max9867: Show Kconfig entry ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix BSEL value in master mode. ASoC: max9867: Take chip out of shutdown sound/soc/atmel/atmel-classd.c | 7 +-- sound/soc/atmel/atmel-pdmic.c | 7 +-- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/max9867.c | 110 +++------------------------------- sound/soc/generic/simple-card-utils.c | 4 ++ 6 files changed, 19 insertions(+), 114 deletions(-) -- 2.15.1

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[alsa-devel] [PATCH - THREAD_SAFE_API 1/1] pcm: Do not access lock_enabled if thread safe API
by twischer＠de.adit-jv.com 01 Mar '18

01 Mar '18

From: Timo Wischer <twischer(a)de.adit-jv.com> is not enabled. Without this commit compiling fails when THREAD_SAFE_API is not enabled Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1be3833..d53ed98 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -823,8 +823,11 @@ int snd_pcm_async(snd_pcm_t *pcm, int sig, pid_t pid) sig = SIGIO; if (pid == 0) pid = getpid(); + +#ifdef THREAD_SAFE_API /* async handler may lead to a deadlock; suppose no multi thread */ pcm->lock_enabled = 0; +#endif return pcm->ops->async(pcm->op_arg, sig, pid); } #endif -- 2.7.4

2 1

[alsa-devel] [PATCH] sound/pci/ice1712: replace strcpy() with scnprintf()
by Joey Pabalinas 01 Mar '18

01 Mar '18

Replace unsafe uses of strcpy() to copy the name argument into the sid.name buffer with scnprintf() to guard against possible buffer overflows. Even though we don't actually care about the return value in this specific case, scnprintf() is still preferred over snprintf() due to scnprintf() returning the much more logical length of what was *actually* encoded into the destination array instead of returning length the resulting string *would* be, assuming it all fit into the destination array as snprintf() does. Maybe one day someone will use the return value, and since the behavior is exactly the same apart from the return value it would be better to account for that possibility and program defensively. Signed-off-by: Joey Pabalinas <joeypabalinas(a)gmail.com> 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 0dbaccf61f33270608..0abacc64168f895d53 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -26,6 +26,7 @@ #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/kernel.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/tlv.h> @@ -425,10 +426,9 @@ DECLARE_TLV_DB_SCALE(juli_master_db_scale, -6350, 50, 1); static struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name) { - struct snd_ctl_elem_id sid; - memset(&sid, 0, sizeof(sid)); - /* FIXME: strcpy is bad. */ - strcpy(sid.name, name); + struct snd_ctl_elem_id sid = {0}; + + scnprintf(sid.name, sizeof(sid.name), "%s", name); sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; return snd_ctl_find_id(card, &sid); } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index d145b5eb7ff86d978d..332f6226548c6a089a 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -25,6 +25,7 @@ #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/kernel.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/tlv.h> @@ -785,10 +786,9 @@ DECLARE_TLV_DB_SCALE(qtet_master_db_scale, -6350, 50, 1); static struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name) { - struct snd_ctl_elem_id sid; - memset(&sid, 0, sizeof(sid)); - /* FIXME: strcpy is bad. */ - strcpy(sid.name, name); + struct snd_ctl_elem_id sid = {0}; + + scnprintf(sid.name, sizeof(sid.name), "%s", name); sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; return snd_ctl_find_id(card, &sid); } -- 2.16.2

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[alsa-devel] [PATCH v1 0/4] ASoC: Add support for DAC PCM1789
by Mylène Josserand 01 Mar '18

01 Mar '18

Hello everyone, You will find in this series the support of Texas Instrument's DAC PCM1789. This DAC is very minimalist and is similar to PCM1792a except for some differences in registers. Series based on asoc tree, "for-next" branch (last commit 130c3888dfdc). It is important to notice that this DAC needs to always have clocks enabled (even without any data) otherwise it will be in a "desynchronized" state and can not send data correctly. This issue has been solved by performing a reset each time a sound is played (see patch 04). This reset can produce a "pop" noise. Thank you in advance for any review. Best regards, Mylène Mylène Josserand (4): ASoC: codecs: pcm179x: Add PCM1789 id ASoC: codecs: pcm179x: Add support for PCM1789 ASoC: codecs: pcm179x: Add reset gpio ASoC: codecs: pcm179x: Add trigger function to perform a reset .../devicetree/bindings/sound/pcm179x.txt | 2 +- sound/soc/codecs/pcm179x-i2c.c | 17 +- sound/soc/codecs/pcm179x.c | 242 ++++++++++++++++++++- sound/soc/codecs/pcm179x.h | 10 +- 4 files changed, 264 insertions(+), 7 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v2] ASoC: atmel: Remove redundant dev_err() call in probe function
by Ladislav Michl 01 Mar '18

01 Mar '18

There is an error message within devm_ioremap_resource already, so remove the redundant dev_err() call. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Acked-by: Nicolas Ferre <nicolas.ferre(a)microchip.com> --- Changes: - v2: None, just added Ack. sound/soc/atmel/atmel-classd.c | 7 ++----- sound/soc/atmel/atmel-pdmic.c | 7 ++----- 2 files changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 9baaf6bc5fbd..3d70061901a0 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -583,11 +583,8 @@ static int atmel_classd_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); io_base = devm_ioremap_resource(dev, res); - if (IS_ERR(io_base)) { - ret = PTR_ERR(io_base); - dev_err(dev, "failed to remap register memory: %d\n", ret); - return ret; - } + if (IS_ERR(io_base)) + return PTR_ERR(io_base); dd->phy_base = res->start; dd->dev = dev; diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 94f852574a44..915c2b064de1 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -633,11 +633,8 @@ static int atmel_pdmic_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); io_base = devm_ioremap_resource(dev, res); - if (IS_ERR(io_base)) { - ret = PTR_ERR(io_base); - dev_err(dev, "failed to remap register memory: %d\n", ret); - return ret; - } + if (IS_ERR(io_base)) + return PTR_ERR(io_base); dd->phy_base = res->start; -- 2.16.2

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