February 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: max98088: Add TLV data for volume controls.
by Dylan Reid 20 Feb '13

20 Feb '13

Specify volumes as defined in the MAX98088/9 data sheet. Allows ALSA lib snd_mixer_selem_get_playback_dB_range and related functions to work. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98088.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index a4c16fd..3a7b7fd 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -739,14 +739,32 @@ static const unsigned int max98088_micboost_tlv[] = { 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), }; +static const unsigned int max98088_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98088_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + static const struct snd_kcontrol_new max98088_snd_controls[] = { - SOC_DOUBLE_R("Headphone Volume", M98088_REG_39_LVL_HP_L, - M98088_REG_3A_LVL_HP_R, 0, 31, 0), - SOC_DOUBLE_R("Speaker Volume", M98088_REG_3D_LVL_SPK_L, - M98088_REG_3E_LVL_SPK_R, 0, 31, 0), - SOC_DOUBLE_R("Receiver Volume", M98088_REG_3B_LVL_REC_L, - M98088_REG_3C_LVL_REC_R, 0, 31, 0), + SOC_DOUBLE_R_TLV("Headphone Volume", M98088_REG_39_LVL_HP_L, + M98088_REG_3A_LVL_HP_R, 0, 31, 0, max98088_hp_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", M98088_REG_3D_LVL_SPK_L, + M98088_REG_3E_LVL_SPK_R, 0, 31, 0, max98088_spk_tlv), + SOC_DOUBLE_R_TLV("Receiver Volume", M98088_REG_3B_LVL_REC_L, + M98088_REG_3C_LVL_REC_R, 0, 31, 0, max98088_spk_tlv), SOC_DOUBLE_R("Headphone Switch", M98088_REG_39_LVL_HP_L, M98088_REG_3A_LVL_HP_R, 7, 1, 1), -- 1.8.0

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[alsa-devel] [PATCH 1/6] ASoC: tegra: assume CONFIG_OF in tegra_asoc_utils_init
by Stephen Warren 20 Feb '13

20 Feb '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra only supports, and always enables, device tree. Remove all runtime checks for DT support from the driver. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_asoc_utils.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index ba419f8..49861c6 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -176,11 +176,7 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else if (of_machine_is_compatible("nvidia,tegra30")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; - else if (!dev->of_node) - /* non-DT is always Tegra20 */ - data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else - /* DT boot, but unknown SoC */ return -EINVAL; data->clk_pll_a = clk_get_sys(NULL, "pll_a"); -- 1.7.10.4

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[alsa-devel] [PATCH 0/8] ASoC: arizona: Add support for directly setting the FLL REFCLK
by Charles Keepax 20 Feb '13

20 Feb '13

This patch series adds support for specifying the source of REFCLK directly for Wolfson Arizona based devices. This should allow for improved performance by using the highest available REFCLK source. Should the REFCLK source not be directly specified then the patches will default to the older behaviour of using the 32kHz clock for REFCLK. REFCLK is configured using a call to snd_soc_codec_set_pll specifying one of the new WM51xx_FLLx_REFCLK defines as the pll_id. It is valid to clear the SYNCCLK source by setting the source to ARIZONA_FLL_SRC_NONE, however once a REFCLK source it can only be replaced with another valid REFCLK source. Charles Keepax (8): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK sound/soc/codecs/arizona.c | 191 ++++++++++++++++++++++++++++++-------------- sound/soc/codecs/arizona.h | 9 ++- sound/soc/codecs/wm5102.c | 6 ++ sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 6 ++ sound/soc/codecs/wm5110.h | 6 +- 6 files changed, 158 insertions(+), 66 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH v5 0/5] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

Just more fixups according to Takashi's comments. David Henningsson (5): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld ALSA: hda - hdmi: Protect ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 52 +++++++++++----------- sound/pci/hda/hda_local.h | 28 +++++++----- sound/pci/hda/patch_hdmi.c | 105 ++++++++++++++++++++++++++++++++++---------- 3 files changed, 127 insertions(+), 58 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH v4 0/5] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

Changes since v3: * snd_hdmi_get_eld now does not take a full hdmi_eld struct David Henningsson (5): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld ALSA: hda - hdmi: Protect ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 52 +++++++++++----------- sound/pci/hda/hda_local.h | 28 +++++++----- sound/pci/hda/patch_hdmi.c | 102 ++++++++++++++++++++++++++++++++++---------- 3 files changed, 124 insertions(+), 58 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH v3 0/5] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

Changes since v2: Added refactoring patch to split hdmi_eld into hdmi_eld and parsed_hdmi_eld. David Henningsson (5): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld ALSA: hda - hdmi: Protect ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 47 ++++++++++++++---------- sound/pci/hda/hda_local.h | 23 +++++++----- sound/pci/hda/patch_hdmi.c | 86 +++++++++++++++++++++++++++++++++----------- 3 files changed, 107 insertions(+), 49 deletions(-) -- 1.7.9.5

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[alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
by wendelin klimann 19 Feb '13

19 Feb '13

hello I try to implement an PCM3168 audio codec from TI to the BeagleBoard but actually i am stuck and it would be rally nice if one could help me a little bit. My ALSA driver version is 1.0.24 and i use an 3.2.18 kernel. As a startingpoint i used the files */linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3008.c* */linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3008.h* The files (see end of e-mail) */linux-mainline-3.2.18-r121b/git/sound/soc/omap/pcm3168_soundcard.c /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.c /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.h* could be compiled successfully and also the insmod loads the modules *$: insmod /lib/modules/3.2.18/kernel/sound/soc/omap/snd-soc-pcm3168_soundcard.ko* *$: lsmod* Module Size Used by snd_soc_pcm3168 1886 0 snd_soc_pcm3168_soundcard 2094 0 twl4 030_madc_hwmon 2569 0 snd_soc_omap3beagle 2239 0 rtc_twl 4382 0 snd_soc_twl4030 37616 1 rtc_ds1307 6642 0 snd_soc_omap 3521 1 snd_soc_omap_mcbsp 8374 1 snd_soc_core 107346 6 snd_soc_pcm3168,snd_soc_pcm3168_soundcard,snd_soc_omap3beagle,snd_soc_twl4030,snd_soc_omap,snd_soc_omap_mcbsp regmap_spi 1263 1 snd_soc_core snd_pcm 78039 4 snd_soc_twl4030,snd_soc_omap,snd_soc_omap_mcbsp,snd_soc_core snd_timer 18848 1 snd_pcm snd 57930 3 snd_soc_core,snd_pcm,snd_timer soundcore 6954 1 snd iptable_nat 3819 0 snd_page_alloc 4979 1 snd_pcm nf_nat 16515 1 iptable_nat nf_conntrack_ipv4 12018 3 iptable_nat,nf_nat nf_conntrack 71219 3 iptable_nat,nf_nat,nf_conntrack_ipv4 nf_defrag_ipv4 1379 1 nf_conntrack_ipv4 ip_tables 10537 1 iptable_nat x_tables 16505 2 iptable_nat,ip_tables gpio_keys 7117 0 autofs4 20492 2 but the new ALSA device is not recognized *$:aplay -l* **** List of PLAYBACK Hardware Devices **** card 0: omap3beagle [omap3beagle], device 0: TWL4030 twl4030-hifi-0 [] Subdevices : 1/1 Subdevice #0: subdevice #0 So i tried to get some information in *$: cd /sys/kernel/debug/asoc* */sys/kernel/debug/asoc$: ls* codecs dais omap3beagle pcm3168 platforms */sys/kernel/debug/asoc$: cat codecs* pcm3168-codec.0 twl4030-codec */sys/kernel/debug/asoc$: cat dais* pcm3168-hifi twl4030-voice twl4030-hifi omap-mcbsp-dai.4 omap-mcbsp-dai.3 omap-mcbsp-dai.2 omap-mcbsp-dai.1 omap-mcbsp-dai.0 */sys/kernel/debug/asoc$: cat platforms* omap-pcm-audio snd-soc-dummy */sys/kernel/debug/asoc$:cd omap3beagle* */sys/kernel/debug/asoc/omap3beagle$:ls* dapm dapm_pop_time twl4030-codec */sys/kernel/debug/asoc/omap3beagle$:cd /sys/kernel/debug/asoc/pcm3168* */sys/kernel/debug/asoc/pcm3168$: ls* dapm_pop_time At this point i have no idea how to solve the problem or how to debug further and i would be very grateful for any hint. Thanks a lot Wendelin * *********************************************************************************************** * /linux-mainline-3.2.18-r121b/git/sound/soc/omap/pcm3168_soundcard.c * ** *********************************************************************************************** */* * PCM3168 ASoC driver for BeagleBoard. * * Author: Hugo Villeneuve * Copyright (C) 2008 Lyrtech inc * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow: * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> //#include <asm/dma.h> #include <mach/hardware.h> #include <mach/gpio.h> //#include <mach/edma.h> #include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/pcm3168.h" /* * CLKX and CLKR are the inputs for the Sample Rate Generator. * FSX and FSR are outputs, driven by the sample Rate Generator. */ /* * SND_SOC_DAIFMT_CBS_CFS * for codec in slave mode */ #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | \ SND_SOC_DAIFMT_IB_NF) static int pcm3168_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 *cpu_dai = rtd->cpu_dai; int fs; int ret = 0; printk("in hw_params -> by kli"); /* Fsref can be 8000 to 96000. */ fs = params_rate(params); /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); if (ret < 0) { printk(KERN_ERR "can't set cpu DAI configuration\n"); return ret; } pr_debug("pcm3168_hw_params: rate = %d Hz\n", fs); return 0; } static struct snd_soc_ops pcm3168_ops = { .hw_params = pcm3168_hw_params, }; /* omap-pcm3168 digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link pcm3168_dai = { .name = "PCM3168", /* Codec name */ .stream_name = "PCM3168", .cpu_dai_name = "omap-mcbsp-dai.2", .codec_dai_name = "pcm3168-hifi", .codec_name = "pcm3168-codec", .platform_name = "omap-pcm-audio", .ops = &pcm3168_ops, }; /* omap-pcm3168 audio machine driver */ static struct snd_soc_card snd_soc_pcm3168 = { .name = "pcm3168", // -> "OMAP pcm3168" .owner = THIS_MODULE, .dai_link = &pcm3168_dai, .num_links = 1, }; /* pcm3168 audio private data */ static struct pcm3168_setup_data pcm3168_pcm3168_setup = { // .dem0_pin = GPIO(45), // .dem1_pin = GPIO(46), // .pdad_pin = GPIO(47), // .pdda_pin = GPIO(38), }; struct platform_device pcm3168_codec = { .name = "pcm3168-codec", .id = 0, .dev = { .platform_data = &pcm3168_pcm3168_setup, }, }; static struct platform_device *pcm3168_snd_device; static int __init pcm3168_init(void) { int ret; printk(KERN_ALERT "in init soundcard \n"); if (!(machine_is_omap3_beagle())) return -ENODEV; platform_device_register(&pcm3168_codec); pcm3168_snd_device = platform_device_alloc("soc-audio", 0); if (!pcm3168_snd_device) { printk(KERN_ERR "platform device allocation failed\n"); return -ENOMEM; } platform_set_drvdata(pcm3168_snd_device, &snd_soc_pcm3168); // platform_device_add_data(pcm3168_snd_device, &pcm3168_snd_data, // sizeof(pcm3168_snd_data)); /* ret = platform_device_add_resources(pcm3168_snd_device, pcm3168_snd_resources, ARRAY_SIZE(pcm3168_snd_resources)); if (ret) { printk(KERN_ERR "platform device add resources failed\n"); goto error; } */ ret = platform_device_add(pcm3168_snd_device); if (ret) goto error; return ret; error: platform_device_put(pcm3168_snd_device); return ret; } static void __exit pcm3168_exit(void) { platform_device_unregister(pcm3168_snd_device); platform_device_unregister(&pcm3168_codec); } module_init(pcm3168_init); module_exit(pcm3168_exit); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_DESCRIPTION("PCM3168 Beagle ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:pcm3168-soc-audio"); * *********************************************************************************************** * /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.c * ** *********************************************************************************************** */* * ALSA Soc PCM3168 codec support * * Author: Hugo Villeneuve * Copyright (C) 2008 Lyrtech inc * * adapted by Klimann Wendelin * * * Based on AC97 Soc codec, original copyright follow: * Copyright 2005 Wolfson Microelectronics PLC. * * 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; either version 2 of the License, or (at your * option) any later version. * * Generic PCM3168 support. */ #include <linux/init.h> #include <linux/kernel.h> #include <linux/device.h> #include <linux/gpio.h> #include <linux/slab.h> #include <linux/module.h> #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> #include <sound/pcm_params.h> //#include <sound/soc-dapm.h> #include <sound/soc.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> #include <plat/mcbsp.h> #include "../omap/omap-mcbsp.h" #include "../omap/omap-pcm.h" #include "pcm3168.h" #define PCM3168_VERSION "0.1" #define PCM3168_RATES SNDRV_PCM_RATE_8000_96000 /*(SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000)*/ static struct snd_soc_dai_driver pcm3168_dai = { .name = "pcm3168-hifi", .playback = { .stream_name = "PCM3168 Playback", .channels_min = 2, .channels_max = 8, .rates = PCM3168_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE }, .capture = { .stream_name = "PCM3168 Capture", .channels_min = 2, .channels_max = 6, .rates = PCM3168_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE }, }; static void pcm3168_gpio_free(struct pcm3168_setup_data *setup) { // gpio_free(setup->dem0_pin); // gpio_free(setup->dem1_pin); // gpio_free(setup->pdad_pin); // gpio_free(setup->pdda_pin); } static int pcm3168_soc_probe(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; int ret = 0; printk(KERN_ALERT "in SOC Probe -> kli \n"); printk(KERN_INFO "PCM3168 SoC Audio Codec %s\n", PCM3168_VERSION); return ret; gpio_err: // pcm3168_gpio_free(setup); return ret; } static int pcm3168_soc_remove(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; // pcm3168_gpio_free(setup); return 0; } #define pcm3168_soc_suspend NULL #define pcm3168_soc_resume NULL static struct snd_soc_codec_driver soc_codec_dev_pcm3168 = { .probe = pcm3168_soc_probe, .remove = pcm3168_soc_remove, .suspend = pcm3168_soc_suspend, .resume = pcm3168_soc_resume, }; static int __devinit pcm3168_codec_probe(struct platform_device *pdev) { int ret; printk(KERN_ALERT "probe pcm3168 -> kli \n"); ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm3168, &pcm3168_dai, 1); printk(KERN_ALERT "probe_after pcm3168 -> kli = %d \n", ret); return ret; } static int __devexit pcm3168_codec_remove(struct platform_device *pdev) { printk(KERN_ALERT "remove pcm3168 -> kli \n"); snd_soc_unregister_codec(&pdev->dev); return 0; } MODULE_ALIAS("platform:pcm3168-codec"); static struct platform_driver pcm3168_codec_driver = { .probe = pcm3168_codec_probe, .remove = __devexit_p(pcm3168_codec_remove), .driver = { .name = "pcm3168-codec", .owner = THIS_MODULE, }, }; static int __init pcm3168_modinit(void) { printk(KERN_ALERT "in init of 3168 -> kli \n"); return platform_driver_register(&pcm3168_codec_driver); } module_init(pcm3168_modinit); static void __exit pcm3168_exit(void) { printk(KERN_ALERT "in exit of 3168 -> kli \n"); platform_driver_unregister(&pcm3168_codec_driver); } module_exit(pcm3168_exit); MODULE_DESCRIPTION("Soc PCM3168 driver"); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_LICENSE("GPL"); * *********************************************************************************************** * /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.h * *********************************************************************************************** * /* * PCM3168 ALSA SoC Layer * * Author: Hugo Villeneuve * Copyright (C) 2008 Lyrtech inc * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_PCM3168_2_H #define __LINUX_SND_SOC_PCM3168_2_H struct pcm3168_setup_data { // unsigned dem0_pin; // unsigned dem1_pin; // unsigned pdad_pin; // unsigned pdda_pin; }; #endif

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[alsa-devel] [PATCH v2 0/4] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

New day, new patches. This time there is a proper lock around access to the ELD buffer. I hope this satisfies the previous comments. I've also tested the patches together with some PulseAudio patches I'm writing and the notification works just fine. The ELD data shows up 300 ms after hotplug, so it's likely the first repoll giving me new ELD data). David Henningsson (4): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: protect access to ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 12 ++++++-- sound/pci/hda/hda_local.h | 3 ++ sound/pci/hda/patch_hdmi.c | 67 +++++++++++++++++++++++++++++++++++--------- 3 files changed, 66 insertions(+), 16 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH 0/3] Make HDMI ELD usable with hotplug
by David Henningsson 19 Feb '13

19 Feb '13

I'm experimenting with using ELD information in PulseAudio and found that it didn't work very well with hotplugging. Neither the proc file nor the ELD kcontrol was cleared correctly after unplug, and there was no way (except polling) to find out when ELD has become available. Could use advice on which ones of these that should or should not be sent to stable. David Henningsson (3): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/patch_hdmi.c | 35 ++++++++++++++++++++++++++++------- 1 file changed, 28 insertions(+), 7 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Jack: add configurable option for irq_flag
by Omair Mohammed Abdullah 18 Feb '13

18 Feb '13

From: Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> For a jack using GPIO based detection, the framework sets it to trigger on both edges (rising and falling). Some codecs may provide only a falling or rising edge interrupt but not both. This patch provides a configurable option for irq_flag during snd_soc_jack_add_gpios(). If not configured, it sets the interrupt line to trigger on both edges. Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)linux.intel.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-jack.c | 7 +++++-- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059c..d0286ad 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -585,6 +585,7 @@ struct snd_soc_jack_zone { * @report: value to report when jack detected * @invert: report presence in low state * @debouce_time: debouce time in ms + * @irq_flag: Interrupt flags for GPIO-Irq line * @wake: enable as wake source * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an @@ -597,6 +598,7 @@ struct snd_soc_jack_gpio { int report; int invert; int debounce_time; + unsigned long irq_flags; bool wake; struct snd_soc_jack *jack; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..c9f7953 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -318,10 +318,13 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, INIT_DELAYED_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; + if (!gpios[i].irq_flags) + gpios[i].irq_flags = + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING; + ret = request_any_context_irq(gpio_to_irq(gpios[i].gpio), gpio_handler, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING, + gpios[i].irq_flags, gpios[i].name, &gpios[i]); if (ret < 0) -- 1.7.4.1

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