July 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] question about sound/soc/ux500/ux500_msp_dai.c
by Himangi Saraogi 11 Jul '14

11 Jul '14

Hi, The probe function calls prcmu_qos_add_requirement but on failure it does not call prcmu_qos_remove_requirement even though that is done in the remove function. The in-kernel definition of both functions just returns 0. There is a prototype that is in a header file, but it is controlled by an #ifdef for an option CONFIG_DBX500_PRCMU_QOS_POWER that does not seem to be supported by the kernel. I am not sure what the prcmu_qos_remove_requirement does. Should it be added in the probe function? Thanks Himangi

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[alsa-devel] [PATCH] ASoC: Intel: Add Broadwell Machine support
by Jie Yang 11 Jul '14

11 Jul '14

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Add support for Broadwell based machines with SST DSP audio. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/Kconfig | 12 +++ sound/soc/intel/Makefile | 2 + sound/soc/intel/broadwell.c | 251 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/broadwell.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c30fedb..0b305f9 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH help This adds audio driver for Intel Baytrail platform based boards with the MAX98090 audio codec. + +config SND_SOC_INTEL_BROADWELL_MACH + tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_HASWELL + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT286 + help + This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell + Ultrabook platforms. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4bfca79..7acbfc4 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-broadwell-objs := broadwell.o obj-$(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 +obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c new file mode 100644 index 0000000..0e550f1 --- /dev/null +++ b/sound/soc/intel/broadwell.c @@ -0,0 +1,251 @@ +/* + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * 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. + * + * 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 <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "sst-dsp.h" +#include "sst-haswell-ipc.h" + +#include "../codecs/rt286.h" + +static const struct snd_soc_dapm_widget broadwell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route broadwell_rt286_map[] = { + + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphones", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int broadwell_ssp0_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); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int broadwell_rt286_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, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broadwell_rt286_ops = { + .hw_params = broadwell_rt286_hw_params, +}; + +static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + /* always connected - check HP for jack detect */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "DMIC1"); + snd_soc_dapm_enable_pin(dapm, "DMIC2"); + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broadwell_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = broadwell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Capture PCM", + .stream_name = "Capture", + .cpu_dai_name = "Capture Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &broadwell_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* broadwell audio machine driver for WPT + RT286S */ +static struct snd_soc_card broadwell_rt286 = { + .name = "broadwell-rt286", + .owner = THIS_MODULE, + .dai_link = broadwell_rt286_dais, + .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .dapm_widgets = broadwell_widgets, + .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), + .dapm_routes = broadwell_rt286_map, + .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .fully_routed = true, +}; + +static int broadwell_audio_probe(struct platform_device *pdev) +{ + broadwell_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&broadwell_rt286); +} + +static int broadwell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&broadwell_rt286); + return 0; +} + +static struct platform_driver broadwell_audio = { + .probe = broadwell_audio_probe, + .remove = broadwell_audio_remove, + .driver = { + .name = "broadwell-audio", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(broadwell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:broadwell-audio"); -- 1.8.3.2

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[alsa-devel] [PATCH 0/2] add rockchip i2s driver
by jianqun 11 Jul '14

11 Jul '14

From: Jianqun Xu <xjq(a)rock-chips.com> This patch is to add driver for I2S controller in RK3xxx SoCs. The only one change in this patch against which has been reviewed before is to fix randconfig build error with next-20140710 by add "#include <linux/module.h>". Jianqun Xu (2): ASoC: dt-bindings: add rockchip i2s bindings changes since v2: - change the description of dma-names, adviced by Mark - change the description of interrupts, adviced by Mark - list clock-names to be easy to read, adviced by Mark - modify compatible for rk3288 example, adviced by Mark and Heiko changes since v1: - modify the description of clock-names property, adviced by Mark Rutland. - modify the example to fit for rk3288. ASoC: add driver for Rockchip RK3xxx I2S controller changes since v3: - fix randconfig build error with next-20140710 by add "#include <linux/module.h>" tested with config given by robot test. changes since v2: - replace 4 with DMA_SLAVE_BUSWIDTH_4_BYTES - modify supported rate to SNDRV_PCM_RATE_8000_192000 - rename head file to "rockchi_i2s.h" changes since v1: - proper indentation for function arguments, adviced by Varka Bhadram - replace sizeof(struct rk_i2s_dev) with sizeof(*i2s), adviced by Mark Rutland - not to include head files which are unused actually in driver, adviced by Mark Brown - modify i2s_tx_status/i2s_rx_status to more meaningful with tx_start/rx_start, adviced by Mark Brown - use regmap_update_bits to reduce the amount of time spent locked, adviced by Mark Brown - add warning for possible error while waiting for the hardware to be ready, adviced by Mark Brown - error checking for clk_set_rate, adviced by Mark Brown - error checking for i2s_runtime_resume, adviced by Mark Brown - remove set_clkdiv operation, clock setting will do it, adviced by Mark Brown - no need SND_SOC for SND_SOC_ROCKCHIP, also allow the driver to be select when COMPILE_TEST is selected, adviced by Lars-Peter Clausen - not to include head files which are unused actually in driver, adviced by Lars-Peter Clausen - I2S core just use one snd_soc_dai_driver struct, adviced by Lars-Peter Clausen - delete rockchip_pcm.c and related codes, use generic dmaengine API, adviced by Lars-Peter and Mark Brown - a driver should never attempt to change its own device name, just use the name that the kernel set, adviced by Lars-Peter and Mark Brown .../devicetree/bindings/sound/rockchip-i2s.txt | 37 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/rockchip/Kconfig | 12 + sound/soc/rockchip/Makefile | 4 + sound/soc/rockchip/rockchip_i2s.c | 531 ++++++++++++++++++++ sound/soc/rockchip/rockchip_i2s.h | 223 ++++++++ 7 files changed, 809 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt create mode 100644 sound/soc/rockchip/Kconfig create mode 100644 sound/soc/rockchip/Makefile create mode 100644 sound/soc/rockchip/rockchip_i2s.c create mode 100644 sound/soc/rockchip/rockchip_i2s.h -- 1.7.9.5

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[alsa-devel] wm8974 and sam9g45 atmel
by Michele Da Rold 11 Jul '14

11 Jul '14

Hello everybody, I'm starting a new development with a Atmel SAM9G45 and a WM8974. I'm using kernel 2.6.33.2 I have connected the SSC1 pin TD,TF,TK and the RD pin, so I want to use WM8974 like a slave. I wrote a interface driver based on sam9263_wm8731.c The driver seem to load ok, I recognize the codec but when I start a mp3 playing (I use mpg123) the playback not start. Putting some dedug messagge the driver not call the function "set_bias_level", and call the mute function with mute=0. Testing the MCLK signal I not see any clock on pin because I never call clk_enable!! There is a way to identify the problem? Someone could help me? Thank you in advance Michele Da Rold I put the code of driver (don't care about printk I used it for debug purpose!!) /* * sam9g45_wm8974 -- SoC audio for AT91SAM9G20-based * ATMEL AT91SAM9G20ek board. * * Copyright (C) 2005 SAN People * Copyright (C) 2008 Atmel * * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> * * Based on sam9263_wm8731.c. * Frank Mandarino <fmandarino(a)endrelia.com> * Copyright 2006 Endrelia Technologies Inc. * Based on corgi.c by: * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/clk.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/i2c.h> #include <linux/atmel-ssc.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 <mach/hardware.h> #include <mach/gpio.h> #include "../codecs/wm8974.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" #define MCLK_RATE 12000000 /* * As shipped the board does not have inputs. However, it is relatively * straightforward to modify the board to hook them up so support is left * in the driver. */ #undef ENABLE_MIC_INPUT #define ENABLE_MIC_INPUT 1 static struct clk *mclk; static int at91sam9g45ek_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->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_hw_params() called\n"); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; return 0; } static struct snd_soc_ops at91sam9g45ek_ops = { .hw_params = at91sam9g45ek_hw_params, }; static int at91sam9g45ek_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { static int mclk_on; int ret = 0; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_set_bias_level() called %d\n",level); switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: if (!mclk_on) ret = clk_enable(mclk); if (ret == 0) mclk_on = 1; break; case SND_SOC_BIAS_OFF: case SND_SOC_BIAS_STANDBY: if (mclk_on) clk_disable(mclk); mclk_on = 0; break; } return ret; } #if 0 static const struct snd_soc_dapm_widget at91sam9g45ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), }; static const struct snd_soc_dapm_route intercon[] = { /* Speaker output mixer */ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, /* speaker connected to SPKOUTP */ {"SpkN Out", NULL, "Speaker Mixer"}, {"SpkP Out", NULL, "Speaker Mixer"}, {"SPKOUTN", NULL, "SpkN Out"}, {"SPKOUTP", NULL, "SpkP Out"}, /* Input PGA */ {"Input PGA", "Aux Switch", "Aux Input"}, {"Input PGA", "MicN Switch", "MICN"}, {"Input PGA", "MicP Switch", "MICP"}, }; /* * Logic for a wm974 as connected on a at91sam9g45ek board. */ static int at91sam9g45ek_wm8974_init(struct snd_soc_codec *codec) { struct snd_soc_dai *codec_dai = &codec->dai[0]; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45_wm8974_init() called\n"); /* ret = snd_soc_dai_set_sysclk(codec_dai, 0, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "Failed to set WM8974 SYSCLK: %d\n", ret); return ret; } */ /* Add specific widgets */ snd_soc_dapm_new_controls(codec, at91sam9g45ek_dapm_widgets, ARRAY_SIZE(at91sam9g45ek_dapm_widgets)); /* Set up specific audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ snd_soc_dapm_nc_pin(codec, "MONOOUT"); snd_soc_dapm_nc_pin(codec, "AUX"); #ifdef ENABLE_MIC_INPUT snd_soc_dapm_enable_pin(codec, "Int Mic"); #else snd_soc_dapm_nc_pin(codec, "Int Mic"); #endif /* always connected */ snd_soc_dapm_enable_pin(codec, "Ext Spk"); snd_soc_dapm_sync(codec); return 0; } #endif static struct snd_soc_dai_link at91sam9g45ek_dai = { .name = "WM8974", .stream_name = "WM8974 PCM", .cpu_dai = &atmel_ssc_dai[1], .codec_dai = &wm8974_dai, #if 0 .init = at91sam9g45ek_wm8974_init, #endif .ops = &at91sam9g45ek_ops, }; static struct snd_soc_card snd_soc_at91sam9g45ek = { .name = "AT91SAMG45", .platform = &atmel_soc_platform, .dai_link = &at91sam9g45ek_dai, .num_links = 1, .set_bias_level = at91sam9g45ek_set_bias_level, }; static struct snd_soc_device at91sam9g45ek_snd_devdata = { .card = &snd_soc_at91sam9g45ek, .codec_dev = &soc_codec_dev_wm8974, }; static struct platform_device *at91sam9g45ek_snd_device; static int __init at91sam9g45ek_init(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc = NULL; struct clk *main; int ret; at91_set_A_periph(AT91_PIN_PD27, 0); /* * Codec MCLK is supplied by PCK1 - set it up. */ mclk = clk_get(NULL, "pck1"); if (IS_ERR(mclk)) { printk(KERN_ERR "ASoC: Failed to get MCLK\n"); ret = PTR_ERR(mclk); goto err; } main = clk_get(NULL, "main"); if (IS_ERR(main)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); ret = PTR_ERR(mclk); goto err_mclk; } ret = clk_set_parent(mclk, main); clk_put(main); if (ret != 0) { printk(KERN_ERR "ASoC: Failed to set MCLK parent\n"); goto err_mclk; } clk_set_rate(mclk, MCLK_RATE); /* * Request SSC device */ ssc = ssc_request(1); if (IS_ERR(ssc)) { printk(KERN_ERR "ASoC: Failed to request SSC 1\n"); ret = PTR_ERR(ssc); ssc = NULL; goto err_ssc; } ssc_p->ssc = ssc; at91sam9g45ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g45ek_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; } platform_set_drvdata(at91sam9g45ek_snd_device, &at91sam9g45ek_snd_devdata); at91sam9g45ek_snd_devdata.dev = &at91sam9g45ek_snd_device->dev; ret = platform_device_add(at91sam9g45ek_snd_device); if (ret) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); platform_device_put(at91sam9g45ek_snd_device); } return ret; err_ssc: ssc_free(ssc); ssc_p->ssc = NULL; err_mclk: clk_put(mclk); mclk = NULL; err: return ret; } static void __exit at91sam9g45ek_exit(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc; if (ssc_p != NULL) { ssc = ssc_p->ssc; if (ssc != NULL) ssc_free(ssc); ssc_p->ssc = NULL; } platform_device_unregister(at91sam9g45ek_snd_device); at91sam9g45ek_snd_device = NULL; clk_put(mclk); mclk = NULL; } module_init(at91sam9g45ek_init); module_exit(at91sam9g45ek_exit); /* Module information */ MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); MODULE_DESCRIPTION("ALSA SoC AT91SAM9G45_WM8974"); MODULE_LICENSE("GPL");

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[alsa-devel] [PATCH] alsa-lib: Provide a CLOCK_MONOTONIC_RAW timestamp type
by Mark Brown 10 Jul '14

10 Jul '14

From: Mark Brown <broonie(a)linaro.org> For applications which need to synchronise with external timebases such as broadcast TV applications the kernel monotonic time is not optimal as it includes adjustments from NTP and so may still include discontinuities due to that. A raw monotonic time which does not include any adjustments is available in the kernel from getrawmonotonic() so provide userspace with a new timestamp type SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW which provides timestamps based on this as an option. Reported-by: Daniel Thompson <daniel.thompson(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/asound.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 1774a5c..9061cdd 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -457,7 +457,8 @@ struct snd_xfern { enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ - SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; /* channel positions */ -- 2.0.0

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[alsa-devel] [PATCH alsa-lib 0/5] Add monotonic raw timestamp support
by Takashi Iwai 10 Jul '14

10 Jul '14

Hi, this is a series of patches to add the monotonic raw timestamp support to alsa-lib. A couple of new API functions and the new type are introduced while the old hw_params monotonic function still remains. Maybe we should mark it depecated? Takashi

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[alsa-devel] [PATCH RESEND v7 0/3] Add ES8328 audio codec
by Sean Cross 10 Jul '14

10 Jul '14

This patchset adds support for the Everest Semi ES8328 audio codec, used in the Novena open source laptop. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks, in addition to the i.MX audmux routing. Changes since v6: - Minor clarification of AUDMUX in imx-audio-es8328.txt - Use u32 instead of int in imx-es8328.c for int-mux and ext-mux - Bounds checking on mux-int-port and mux-ext-port in imx-es8328.c - Remove NULL checks before calling of_node_put() Changes since v5: - Removed empty imx_set_frequency function from imx-es8328.c - Move the clock from imx-audio-es8328 to the es8328 codec itself - Add support for 2x (22.5792 MHz) and 1x (11.2896 MHz) source clocks - Restore registers on resume - Document pins for ES8328, including input pins Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/imx-audio-es8328.txt | 61 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 719 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 233 +++++++ 12 files changed, 1510 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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[alsa-devel] [PATCH] ASoC: rt286: make rt286_i2c_driver static
by bardliao＠realtek.com 10 Jul '14

10 Jul '14

From: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- This is reported by kbuild test robot. I don't know if my sob line is needed or not. --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8103315..218f86e 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1206,7 +1206,7 @@ static int rt286_i2c_remove(struct i2c_client *i2c) } -struct i2c_driver rt286_i2c_driver = { +static struct i2c_driver rt286_i2c_driver = { .driver = { .name = "rt286", .owner = THIS_MODULE, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: omap-dmic: use managed interfaces
by Himangi Saraogi 10 Jul '14

10 Jul '14

This patch moves data allocated using clk_put and snd_soc_register_component to the corresponding managed interfaces and does away with the functions to free memory in the probe and remove functions. The probe function is no longer required and is completely removed. Signed-off-by: Himangi Saraogi <himangi774(a)gmail.com> Acked-by: Julia Lawall <julia.lawall(a)lip6.fr> --- sound/soc/omap/omap-dmic.c | 35 ++++++++--------------------------- 1 file changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 6925d71..b315b27 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -466,7 +466,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) mutex_init(&dmic->mutex); - dmic->fclk = clk_get(dmic->dev, "fck"); + dmic->fclk = devm_clk_get(dmic->dev, "fck"); if (IS_ERR(dmic->fclk)) { dev_err(dmic->dev, "cant get fck\n"); return -ENODEV; @@ -475,8 +475,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); if (!res) { dev_err(dmic->dev, "invalid dma memory resource\n"); - ret = -ENODEV; - goto err_put_clk; + return -ENODEV; } dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG; @@ -484,34 +483,17 @@ static int asoc_dmic_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); dmic->io_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(dmic->io_base)) { - ret = PTR_ERR(dmic->io_base); - goto err_put_clk; - } - + if (IS_ERR(dmic->io_base)) + return PTR_ERR(dmic->io_base); - ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component, - &omap_dmic_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &omap_dmic_component, + &omap_dmic_dai, 1); if (ret) - goto err_put_clk; + return ret; ret = omap_pcm_platform_register(&pdev->dev); if (ret) - goto err_put_clk; - - return 0; - -err_put_clk: - clk_put(dmic->fclk); - return ret; -} - -static int asoc_dmic_remove(struct platform_device *pdev) -{ - struct omap_dmic *dmic = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - clk_put(dmic->fclk); + return ret; return 0; } @@ -529,7 +511,6 @@ static struct platform_driver asoc_dmic_driver = { .of_match_table = omap_dmic_of_match, }, .probe = asoc_dmic_probe, - .remove = asoc_dmic_remove, }; module_platform_driver(asoc_dmic_driver); -- 1.9.1

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[alsa-devel] [PATCH 0/7] Arizona FLL Bug Fixes
by Charles Keepax 10 Jul '14

10 Jul '14

Hi, This patch series contains several small bug fixes and improves the handling of the situation in which an already running FLL is reconfigured. In the case of a running FLL only some tranisitions are supported whereas previously all transitions were allowed. Thanks, Charles Charles Keepax (7): ASoC: arizona: Do not test ratio zero as it is not a valid setting ASoC: arizona: Correct checking of FLL ratio limitations ASoC: arizona: Correct relationship between VCO corner and Fref ASoC: arizona: Coding standards, remove unneeded brackets ASoC: arizona: Correct return value of arizona_is_enabled_fll ASoC: arizona: FLL freerun only required whilst disabling ASoC: arizona: Update handling for input change on an active FLL sound/soc/codecs/arizona.c | 63 ++++++++++++++++++++++++++++++-------------- 1 files changed, 43 insertions(+), 20 deletions(-) -- 1.7.2.5

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