December 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a slightly lengthy patchset, basically targeted for 4.13, thus it's still in RFC. The main purpose of this patchset is to get rid of get_fs() / set_fs() usages in ALSA codes. The patchset starts from a patch that looks fairly irrelevant with the goal (the conversion of HD-audio bind-mute code), but it's a step for the long trail. The biggest change in the patchset is the conversion of copy & silence PCM ops into the new copy_silence ops. It helped for reducing the code size and for handling the in-kernel buffer copies. The current patches are found in topic/kill-set_fs branch of my sound.git tree. thanks, Takashi === Takashi Iwai (26): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage ALSA: pcm: Remove set_fs() in PCM core code ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Kill set_fs() usage in OSS layer and USB gadget .../sound/kernel-api/writing-an-alsa-driver.rst | 110 +++++---- drivers/media/pci/solo6x10/solo6x10-g723.c | 11 +- drivers/usb/gadget/function/u_uac1.c | 8 +- include/sound/pcm.h | 19 +- sound/core/oss/pcm_oss.c | 62 ++--- sound/core/pcm_compat.c | 22 +- sound/core/pcm_lib.c | 174 +++++++++----- sound/core/pcm_native.c | 122 ++++++---- sound/drivers/dummy.c | 13 +- sound/drivers/opl3/opl3_oss.c | 14 -- sound/isa/gus/gus_pcm.c | 43 +--- sound/isa/sb/emu8000_pcm.c | 99 +++----- sound/pci/emu10k1/emufx.c | 127 ++++++----- sound/pci/es1938.c | 11 +- sound/pci/hda/hda_codec.c | 251 +++------------------ sound/pci/hda/hda_generic.c | 46 +++- sound/pci/hda/hda_local.h | 61 ----- sound/pci/hda/patch_realtek.c | 37 ++- sound/pci/korg1212/korg1212.c | 128 +++-------- sound/pci/nm256/nm256.c | 35 ++- sound/pci/rme32.c | 49 ++-- sound/pci/rme96.c | 52 ++--- sound/pci/rme9652/hdsp.c | 44 ++-- sound/pci/rme9652/rme9652.c | 46 ++-- sound/sh/sh_dac_audio.c | 40 +--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 ++-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 696 insertions(+), 1001 deletions(-) -- 2.12.2

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[alsa-devel] [PATCH] ASoC: Intel: Boards: Add CNL RT274 I2S machine driver
by Guneshwor Singh 17 May '18

17 May '18

Add CNL I2S machine driver using Realtek RT274 codec in I2S mode configured to ssp0. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/boards/Kconfig | 13 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cnl_rt274.c | 250 +++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/boards/cnl_rt274.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6f754708a48c..29066f0b744f 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -262,4 +262,17 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_CNL_RT274_MACH + tristate "ASoC Audio driver for Cannonlake with RT274 I2S mode" + depends on X86 && ACPI && I2C + select SND_SOC_INTEL_SST + depends on SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT274 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Cannonlake platform + with RT274 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 69d2dfaeb00c..a7a832c072ca 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -19,6 +19,7 @@ snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-cnl-rt274-objs := cnl_rt274.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 @@ -40,3 +41,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_CNL_RT274_MACH) += snd-soc-cnl-rt274.o diff --git a/sound/soc/intel/boards/cnl_rt274.c b/sound/soc/intel/boards/cnl_rt274.c new file mode 100644 index 000000000000..19458fb50f55 --- /dev/null +++ b/sound/soc/intel/boards/cnl_rt274.c @@ -0,0 +1,250 @@ +/* + * cnl_rt274.c - ASOC Machine driver for CNL + * + * Copyright (C) 2016-17 Intel Corp + * Author: Guneshwor Singh <guneshwor.o.singh(a)intel.com> + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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 <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/rt274.h" + +#define CNL_FREQ_OUT 24000000 +#define CNL_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai = + snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + int ret, ratio = 100; + + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, + CNL_FREQ_OUT, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + if (ret) { + dev_err(codec_dai->dev, + "set bclk ratio failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + CNL_BE_FIXUP_RATE * ratio, + CNL_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, + "enable PLL2 failed: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_jack cnl_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin cnl_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new cnl_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget cnl_rt274_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + cnl_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static int cnl_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static const struct snd_soc_dapm_route cnl_map[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + {"DMic", NULL, "SoC DMIC"}, + {"DMIC01 Rx", NULL, "Capture"}, + {"dmic01_hifi", NULL, "DMIC01 Rx"}, + + {"AIF1 Playback", NULL, "ssp0 Tx"}, + {"ssp0 Tx", NULL, "codec1_out"}, + {"ssp0 Tx", NULL, "codec0_out"}, + + {"ssp0 Rx", NULL, "AIF1 Capture"}, + {"codec0_in", NULL, "ssp0 Rx"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"Mic Jack", NULL, "Platform Clock"}, +}; + +static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + int ret; + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET, &cnl_headset, + cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins)); + if (ret) + return ret; + + ret = snd_soc_codec_set_jack(codec, &cnl_headset, NULL); + if (ret) + return ret; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int cnl_be_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); + + rate->min = rate->max = CNL_BE_FIXUP_RATE; + channels->min = channels->max = 2; + snd_mask_none(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT)); + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static struct snd_soc_dai_link cnl_rt274_dailink[] = { + { + .name = "SSP0-Codec", + .cpu_dai_name = "SSP0 Pin", + .codec_name = "i2c-INT34C2:00", + .codec_dai_name = "rt274-aif1", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_be_fixup, + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = cnl_rt274_init, + }, + { + .name = "dmic01", + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_dmic_fixup, + .no_pcm = 1, + .ignore_suspend = 1, + .dpcm_capture = 1, + }, +}; + +static int +cnl_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + link->platform_name = "0000:00:1f.3"; + link->nonatomic = 1; + + return 0; +} + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cnl = { + .name = "cnl-audio", + .dai_link = cnl_rt274_dailink, + .num_links = ARRAY_SIZE(cnl_rt274_dailink), + .dapm_widgets = cnl_rt274_widgets, + .num_dapm_widgets = ARRAY_SIZE(cnl_rt274_widgets), + .dapm_routes = cnl_map, + .num_dapm_routes = ARRAY_SIZE(cnl_map), + .controls = cnl_controls, + .num_controls = ARRAY_SIZE(cnl_controls), + .add_dai_link = cnl_add_dai_link, + .fully_routed = true, +}; + +static int snd_cnl_rt274_probe(struct platform_device *pdev) +{ + snd_soc_card_cnl.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cnl); +} + +static struct platform_driver snd_cnl_rt274_driver = { + .driver = { + .name = "cnl_rt274", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cnl_rt274_probe, +}; + +module_platform_driver(snd_cnl_rt274_driver); + +MODULE_AUTHOR("Guneshwor Singh <guneshwor.o.singh(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cnl_rt274"); -- 2.15.0

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[alsa-devel] [PATCH 43/45] sound/soc/zte: remove duplicate includes
by Pravin Shedge 17 May '18

17 May '18

These duplicate includes have been found with scripts/checkincludes.pl but they have been removed manually to avoid removing false positives. Signed-off-by: Pravin Shedge <pravin.shedge4linux(a)gmail.com> --- sound/soc/zte/zx-i2s.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 9a05659..93428e1 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -20,9 +20,6 @@ #include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/initval.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> #define ZX_I2S_PROCESS_CTRL 0x04 #define ZX_I2S_TIMING_CTRL 0x08 -- 2.7.4

2 2

[alsa-devel] Prepare to covert Codec/Platform into Component
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark, Lars-Peter I will post all prepare patches to convert Codec/Platform into Component. It has 3 steps. 1) add new necessary functions 2) remove/don't use codec hw_write/read 3) remove/don't use codec reg_cache If these are accepted, all prepares were completed. Next, I can post "convert Codec/Platform into Component" patch bomb. These are too many convert patches, to avoid review issue, I will post few of them as sample. 4) sample of don't use rtd->codec 5) sample of convert Platform into Component 6) sample of convert Codec into Component I will post some of "remove" patches too for each steps. I will re-post all "remove" patches if all are accepted. For your review, please consider below. If these are not good, all covert patch doesn't work. 1. use devm_snd_soc_register_component() and remove snd_soc_unregister_component() My patches is using devm_ version function, and removed unregister function. This is not a big deal 2. New flags. Because "Codec" and "CPU/Platform" are using different features, we need to use some new flags. .idle_bias_off/on .ignore_pmdown_time/pmdown_time These are because "CPU/Platform" and "Codec" are using different default value. "Codec" side convert patches are using these. .endianness .non_legacy_dai_naming These are because "Codec" only needs fixup format, and not use "legacy" DAI name. "Codec" side convert patches are using these I added its flags info on each patch. 3. rtd->platform rtd->platform will be removed. Now we can find all rtd connected component by using rtdcom_lookup. rtd->platform will be replaced to snd_soc_rtdcom_lookup(rtd, DRV_NAME); Here, DRV_NAME is platform's component driver name. 4. rtd->codec rtd->codec will be removed. In soc_bind_dai_link(), It is rtd->codec = rtd->codec_dai->codec; And now, DAI has codec and component pointer. So, rtd->codec will be replaced into rtd->codec_dai->component. Best regards --- Kuninori Morimoto

2 104

[alsa-devel] Disable DAPM for click avoidance ?
by Ricard Wanderlof 27 Mar '18

27 Mar '18

I'm having an issue with a codec where the normal DAPM procedure of disabling most of the codec when there is no stream ongoing results in large clicks when a playback stream starts and stops, because the output jumps from 0V when the output is disabled to over a volt when the output is enabled. There appears to be a 'slow stop/start' feature in the codec, however, it takes about one second to ramp up and down the output voltage, and that is too long a delay when starting a stream. In our usecase the power consumtion of the codec is negligable compared to the rest of the electronics, so leaving the codec running at all times is perfectly feasible. Is there any way from a machine driver to disable the DAPM switching, i.e. leave parts of the codec running at all times? In this specific case, there are two internal PM controls which enable the output. One of them is controlled by an output mixer element (SND_SOC_DAPM_MIXER_E), and the other one from set_bias_level(). I've had a look in the core code, but it seems that the whole business of changing the bias level when a stream is started is always performed and that there is no way to say "Set the bias level to ON and leave it there". An alternative would be to augment the codec driver with an 'always on' feature, which effectively always enables PM for certain parts of the chip. But it seems like the wrong place to do this type of thing. Another use for this feature is that we have had a lot of problems over the years with buggy I2S interfaces misbehaving when streams are started and stopped, and there would be an advantage to letting the AIF run the whole time after startup. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH v1 0/6] ALSA: usb: UAC3. Add support for Basic Audio Device (BADD)
by Jorge Sanjuan 19 Mar '18

19 Mar '18

This adds functionality for the Basic Audio Device (BADD) subset that is defined in the USB Audio Class 3 (UAC3). In a BADD compliant driver, all Class Specific descriptors are inferred and so they should all be generated in the *host* as the device wont expose those. This also applies to the new HIGH CAPABILITY descriptor as the GET request is not to be supported in a BADD compliant driver. The new class requires the device to have 3 usb configurations as follows: 1: Legacy Mode: UAC1 or UAC2. 2: BADD device with a prefined topology. (Minimum). 3: UAC3 device for more detailed description or more complex devices that can't be covered by the BADD profile. This patch series also includes some minor fixes to sound/usb. It has been tested with an actual UAC3 compliant device in different host machines and all configs (UAC1/2, BADD and UAC3) work. Also, this has been implemented on top of the the patch which adds UAC3 support to the usb sound card driver: commit ddd452d7b04b86fb5f9285a19ac54deca9264ac1 Author: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> Date: Tue Nov 7 04:01:20 2017 +0200 Jorge Sanjuan (6): usb: audio: Add bi-directional terminal types. usb: audio: Fix variable length field to be variable. ALSA: usb: UAC3. Add initial mixer support. ALSA: usb: Only get AudioControl header for UAC1 class. ALSA: usb: Use Class Specific EP for UAC3 devices. ALSA: usb: ADC3: Add initial BADD spec support include/linux/usb/audio-v3.h | 2 +- include/uapi/linux/usb/audio.h | 21 +- sound/usb/Makefile | 3 +- sound/usb/badd.c | 518 +++++++++++++++++++++++++++++++++++++++++ sound/usb/badd.h | 14 ++ sound/usb/card.c | 25 +- sound/usb/mixer.c | 24 ++ sound/usb/stream.c | 165 ++++++++----- sound/usb/usbaudio.h | 1 + 9 files changed, 702 insertions(+), 71 deletions(-) create mode 100644 sound/usb/badd.c create mode 100644 sound/usb/badd.h -- 2.11.0

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[alsa-devel] [PATCH v1 00/15] ASoC: fsl_ssi: Clean up - program flow level
by Nicolin Chen 22 Feb '18

22 Feb '18

==Background== The fsl_ssi driver was designed for PPC originally and then it has been updated to support different modes for i.MX Series, including SDMA, I2S Master mode, AC97 and older i.MXs with FIQ, by different contributors for different use cases in different coding styles. Additionally, in order to fix/work-around hardware bugs and design flaws, the driver made a lot of compromise so now its program flow looks very complicated and it's getting hard to maintain or update. So I am going to clean up the driver on both coding style level and program flow level. ==Introduction== This series of patches is the second set to clean up fsl_ssi driver in the program flow level. Any patch here may impact a fundamental test case like playback or record. ==Verification== This series of patches require fully tested. I have done such tests on i.MX6SoloX with WM8962 using imx_v6_v7_defconfig as: - Playback via I2S Master and Slave mode - Record via I2S Master and Slave mode - Simultaneous playback and record via I2S Master and Slave mode - Background playback with foreground record (starting at different time) via I2S Master and Slave mode - Background record with foreground playback (starting at different time) via I2S Master and Slave mode * All tests above by hacking offline_config to true in imx51. Example of uncovered tests: TDM, AC97, PowerPC and FIQ. Nicolin Chen (15): ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to dai_probe() ASoC: fsl_ssi: Setup AC97 in dai_probe() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Remove cpu_dai_drv from fsl_ssi structure ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register sound/soc/fsl/fsl_ssi.c | 710 ++++++++++++++++++++++++------------------------ 1 file changed, 348 insertions(+), 362 deletions(-) -- 2.7.4

5 34

[alsa-devel] [PATCH] drm/i915: Remove unused IRQ chip data of HDMI LPE audio
by Augustine.Chen 20 Feb '18

20 Feb '18

The chip data of HDMI LPE audio is set to drm_i915_private which is not consistent with the expectation by x86 APIC driver. In the case of not enabling CONFIG_CPUMASK_OFFSTACK, this would cause kernel panic while doing CPU hotplug. Since the dependency of IRQ chip data was removed from HDMI LPE audio by Commit 9bd9590997b92fbd79fd028f704f6c584b4439d7 ("drm/i915: Stop pretending to mask/unmask LPE audio interrupts"), remove the code of setting IRQ chip data to resolve this issue. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=103731 Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> Cc: Jerome Anand <jerome.anand(a)intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Augustine.Chen <augustine.chen(a)intel.com> --- drivers/gpu/drm/i915/intel_lpe_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/i915/intel_lpe_audio.c b/drivers/gpu/drm/i915/intel_lpe_audio.c index 3bf6528..56176f9 100644 --- a/drivers/gpu/drm/i915/intel_lpe_audio.c +++ b/drivers/gpu/drm/i915/intel_lpe_audio.c @@ -176,7 +176,7 @@ static int lpe_audio_irq_init(struct drm_i915_private *dev_priv) handle_simple_irq, "hdmi_lpe_audio_irq_handler"); - return irq_set_chip_data(irq, dev_priv); + return 0; } static bool lpe_audio_detect(struct drm_i915_private *dev_priv) -- 1.9.1

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[alsa-devel] [PATCH 0/8] add UniPhier audio system support
by Katsuhiro Suzuki 12 Feb '18

12 Feb '18

This series adds support for Socionext audio system for UniPhier LD11/LD20 SoCs. This driver supports I2S output for Line-In, Line-Out and S/PDIF output. UniPhier AIO DAI driver provides sound devices such as I2S, S/PDIF. Since the AIO has mixed register map for those I/Os, it is hard to split register areas for each sound devices. UniPhier EVEA is a codec core for Line-In, Line-Out. Since this core is only in internal of UniPhier series SoC, this driver works independently but only UniPhier SoC can use it. Katsuhiro Suzuki (8): ASoC: spdif: Add S32_LE support for S/PDIF dummy codec drivers ASoC: uniphier: add DT bindings documentation for UniPhier EVEA ASoC: uniphier: add DT bindings documentation for UniPhier AIO ASoC: uniphier: add support for UniPhier EVEA codec ASoC: uniphier: add support for UniPhier AIO driver ASoC: uniphier: add support for UniPhier LD11/LD20 AIO driver MAINTAINERS: add entries for UniPhier ASoC sound drivers arm64: dts: uniphier: add sound node for UniPhier .../devicetree/bindings/sound/uniphier,aio.txt | 36 ++ .../devicetree/bindings/sound/uniphier,evea.txt | 26 + MAINTAINERS | 6 + .../boot/dts/socionext/uniphier-ld11-global.dts | 72 +++ arch/arm64/boot/dts/socionext/uniphier-ld11.dtsi | 31 + .../boot/dts/socionext/uniphier-ld20-global.dts | 72 +++ arch/arm64/boot/dts/socionext/uniphier-ld20.dtsi | 35 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/spdif_receiver.c | 5 +- sound/soc/codecs/spdif_transmitter.c | 5 +- sound/soc/uniphier/Kconfig | 34 + sound/soc/uniphier/Makefile | 8 + sound/soc/uniphier/aio-core.c | 368 +++++++++++ sound/soc/uniphier/aio-dma.c | 266 ++++++++ sound/soc/uniphier/aio-ld11.c | 343 ++++++++++ sound/soc/uniphier/aio-regctrl.c | 699 +++++++++++++++++++++ sound/soc/uniphier/aio-regctrl.h | 495 +++++++++++++++ sound/soc/uniphier/aio.h | 261 ++++++++ sound/soc/uniphier/evea.c | 567 +++++++++++++++++ 20 files changed, 3327 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/uniphier,aio.txt create mode 100644 Documentation/devicetree/bindings/sound/uniphier,evea.txt create mode 100644 sound/soc/uniphier/Kconfig create mode 100644 sound/soc/uniphier/Makefile create mode 100644 sound/soc/uniphier/aio-core.c create mode 100644 sound/soc/uniphier/aio-dma.c create mode 100644 sound/soc/uniphier/aio-ld11.c create mode 100644 sound/soc/uniphier/aio-regctrl.c create mode 100644 sound/soc/uniphier/aio-regctrl.h create mode 100644 sound/soc/uniphier/aio.h create mode 100644 sound/soc/uniphier/evea.c -- 2.15.0

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[alsa-devel] [PATCH 0/x] ASoC: replace platform to component
by Kuninori Morimoto 12 Feb '18

12 Feb '18

Hi Mark These are for replace "platform" to "component". It needs 3 steps I think. step1. "component" needs _add_ function, like "platform" step2. "component" level pcm_new/free support step3. replace platform to component We can get platform pointer from rtd->platform on current style. But these are very categorized pointer, in other words, non generic pointer. This cleanup is 1st step of merging all feature into component. I think new style wil be CPU + Codec + Platform = Component I think merging CPU into Component is already done. This patch set merges Platform feature into Component, and remove Platform. Merging Codec into Component is next step. This patch set will add new rtd connected component list. All CPU/Codec/Platform are connected to this list. And we can get each component pointer by using rtd and its driver name. Here, We can replace rtd->platform like below - struct device *dev = rtd->platform->dev; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRIVER_NAME); + struct device *dev = component->dev; - if(rtd->platform->pcm_new) - rtd->platform->pcm_new(yyy); + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; /* rtd connected CPU/Codec/Platform */ + if (component->pcm_new) + component->pcm_new(yyy); + } Now, only platform has pcm_new/free, but in new style, all component can have it. I will post step1/2/3 patch-set. step1 is normal patches. step2/3 are now still [RFC]. Best regards --- Kuninori Morimoto

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