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[alsa-devel] [PATCH 2/2] ASoC: wm8960: Fix shared LRCLK support
by Ma Haijun 14 Aug '13

14 Aug '13

Shared LRCLK initialization does not survive wm8960_reset, so place it after the reset. Signed-off-by: Ma Haijun <mahaijuns(a)gmail.com> --- sound/soc/codecs/wm8960.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 368d39f..b606dd3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -990,6 +990,9 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pdata && pdata->shared_lrclk) + snd_soc_update_bits(codec, WM8960_ADDCTL2, 0x4, 0x4); + /* Latch the update bits */ snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); @@ -1041,7 +1044,6 @@ static const struct regmap_config wm8960_regmap = { static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct wm8960_data *pdata = dev_get_platdata(&i2c->dev); struct wm8960_priv *wm8960; int ret; @@ -1054,16 +1056,6 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); - if (pdata && pdata->shared_lrclk) { - ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, - 0x4, 0x4); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", - ret); - return ret; - } - } - i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8960: Fix ADC volume bits
by Ma Haijun 14 Aug '13

14 Aug '13

Signed-off-by: Ma Haijun <mahaijuns(a)gmail.com> --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd..368d39f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -263,8 +263,8 @@ SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0), SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0), -SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH, - 0, 127, 0), +SOC_DOUBLE_R_TLV("ADC PCM Capture Volume", WM8960_LADC, WM8960_RADC, + 0, 255, 0, adc_tlv), SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume", WM8960_BYPASS1, 4, 7, 1, bypass_tlv), -- 1.8.1.2

2 1

[alsa-devel] [PATCH 0/29] simplify use of devm_ioremap_resource
by Julia Lawall 14 Aug '13

14 Aug '13

devm_ioremap_resource often uses the result of a call to platform_get_resource as its last argument. devm_ioremap_resource does appropriate error handling on this argument, so error handling can be removed from the call site. To make the connection between the call to platform_get_resource and the call to devm_ioremap_resource more clear, the former is also moved down to be adjacent to the latter. In many cases, this patch changes the specific error value that is returned on failure of platform_get_resource. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l; @@ ( res = platform_get_resource(pdev, IORESOURCE_MEM, n); - if (res == NULL) { ... $goto l;\|return ret;$ } e = devm_ioremap_resource(e1, res); | - res = platform_get_resource(pdev, IORESOURCE_MEM, n); ... when != res - if (res == NULL) { ... $goto l;\|return ret;$ } ... when != res + res = platform_get_resource(pdev, IORESOURCE_MEM, n); e = devm_ioremap_resource(e1, res); ) // </smpl>

2 3

[alsa-devel] HDMI audio is not working on my new desktop
by Alexander E. Patrakov 14 Aug '13

14 Aug '13

Hello. I built a new desktop PC based on the Gigabyte H87N-WIFI motherboard. It has one DVI and two HDMI outputs. However, I cannot get any sound over HDMI. The same OS image works perfectly on my laptop that also has an HDMI output. This is not a pulseaudio problem, as I don't get any sound even if I test with this command: pasuspender -- speaker-test -c2 -D hdmi:0,2 # or, for that matter, 0,0 or 0,1 Alsa-info.sh has uploaded its output to http://www.alsa-project.org/db/?f=1ad7192a15ff4a70b5ddb2579476dbb98adbd929 Please note this error in dmesg, it may be relevant: [ 1.803595] hda_codec: invalid CONNECT_LIST verb 5[1]:0 [ 1.803719] hda_codec: invalid CONNECT_LIST verb 7[1]:0 -- Alexander E. Patrakov

1 0

Re: [alsa-devel] Query on Audio DMA using DMAEngine
by Lars-Peter Clausen 14 Aug '13

14 Aug '13

Added alsa-devel to Cc. On 06/28/2013 05:27 AM, Fernandes, Joel wrote: > Hi Lars, > > Hope you are doing well. > > I am implementing Cyclic DMA support in the EDMA driver that is used by Davinci and now newer TI SoCs. > I am thinking once I am done I can plug it into the snd_dmaengine framework. > > Currently however, the davinci-pcm code directly programs the EDMA. That is what I am working to replace with a single driver and adapt to the snd dmaengine framework. However, once the current code in davinci-pcm uses internal RAM as an intermediate step in the whole DMA process (First data is TX to IRAM from DRAM and then from DRAM to the audio device). > > Do you have any ideas on how we can adapt to the framework, such that we can till use the IRAM? Are there any existing implementations out there that do something similar? Hm, I guess using the snd_dmaengine_pcm helper functions here shouldn't be too hard. Using the generic snd_dmaengine_pcm driver will require some extensions to it though. The mmp platform (pxa/mmp-pcm.c) is also using some kind of on-chip memory, so having support for this in the generic driver certainly makes sense. For the chaining you'd probably have to extend the dmaengine framework, since this kind of interleaved mem-to-mem and mem-to-dev cyclic transfer is currently not possible. I'm wondering though why do you need to copy the data to RAM first, is it not possible to map the IRAM to userspace? - Lars

6 28

[alsa-devel] [PATCH v5 0/1] ASoc: kirkwood: extend the kirkwood audio subsystem for DT usage
by Jean-Francois Moine 14 Aug '13

14 Aug '13

This patch series adds DT support to the kirkwood/mvebu audio subsystem. History v5 - change the DT name to "marvell,mvebu-audio" v4 - change the DT name to "mrvl,mvebu-audio" - remove previous patches 1..3 which have been applied by Russell King v3 - change the DT name from "kirkwood-i2s" to "mvebu-pcm-audio" - apply to linux-next - let NULL as the internal i2s clock for compatibility - change i2s external clock name to "extclk" instead of "external" - cancel previous patch 1/4. Russell's patch should be applied instead. v2 - fix compilation errors -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] [PATCH] ASoC: pcm1681: Add DAPM support
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Provide DAPM for the device, ensuring operation with DAPM required by the core and making it easier to hook up external hardware to it. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1681.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 51b1866..651ce09 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -206,6 +206,28 @@ static const struct snd_soc_dai_ops pcm1681_dai_ops = { .digital_mute = pcm1681_digital_mute, }; +static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUT1"), +SND_SOC_DAPM_OUTPUT("VOUT2"), +SND_SOC_DAPM_OUTPUT("VOUT3"), +SND_SOC_DAPM_OUTPUT("VOUT4"), +SND_SOC_DAPM_OUTPUT("VOUT5"), +SND_SOC_DAPM_OUTPUT("VOUT6"), +SND_SOC_DAPM_OUTPUT("VOUT7"), +SND_SOC_DAPM_OUTPUT("VOUT8"), +}; + +static const struct snd_soc_dapm_route pcm1681_dapm_routes[] = { + { "VOUT1", NULL, "Playback" }, + { "VOUT2", NULL, "Playback" }, + { "VOUT3", NULL, "Playback" }, + { "VOUT4", NULL, "Playback" }, + { "VOUT5", NULL, "Playback" }, + { "VOUT6", NULL, "Playback" }, + { "VOUT7", NULL, "Playback" }, + { "VOUT8", NULL, "Playback" }, +}; + static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); static const struct snd_kcontrol_new pcm1681_controls[] = { @@ -258,6 +280,10 @@ static const struct regmap_config pcm1681_regmap = { static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { .controls = pcm1681_controls, .num_controls = ARRAY_SIZE(pcm1681_controls), + .dapm_widgets = pcm1681_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1681_dapm_widgets), + .dapm_routes = pcm1681_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1681_dapm_routes), }; static const struct i2c_device_id pcm1681_i2c_id[] = { -- 1.8.4.rc2

1 0

[alsa-devel] [PATCH 2/3 v2] ASoC: wm8997: Initial CODEC driver
by Charles Keepax 13 Aug '13

13 Aug '13

The wm8997 is a compact, high-performance audio hub CODEC with SLIMbus interfacing, for smartphones, tablets and other portable audio devices based on the Arizona platform. This patch adds the wm8997 CODEC driver. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 13 +- sound/soc/codecs/wm8997.c | 1187 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8997.h | 23 + 5 files changed, 1228 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..14d2888 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -120,6 +120,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8996 if I2C + select SND_SOC_WM8997 if MFD_WM8997 select SND_SOC_WM9081 if I2C select SND_SOC_WM9090 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS @@ -143,8 +144,10 @@ config SND_SOC_ARIZONA tristate default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y + default y if SND_SOC_WM8997=y default m if SND_SOC_WM5102=m default m if SND_SOC_WM5110=m + default m if SND_SOC_WM8997=m config SND_SOC_WM_HUBS tristate @@ -492,6 +495,9 @@ config SND_SOC_WM8995 config SND_SOC_WM8996 tristate +config SND_SOC_WM8997 + tristate + config SND_SOC_WM9081 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..2d327ad 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -112,6 +112,7 @@ snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o +snd-soc-wm8997-objs := wm8997.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9090-objs := wm9090.o snd-soc-wm9705-objs := wm9705.o @@ -235,6 +236,7 @@ obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o +obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index de62581..c460fca 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -199,9 +199,16 @@ int arizona_init_spk(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1); - if (ret != 0) - return ret; + switch (arizona->type) { + case WM8997: + break; + default: + ret = snd_soc_dapm_new_controls(&codec->dapm, + &arizona_spkr, 1); + if (ret != 0) + return ret; + break; + } ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN, "Thermal warning", arizona_thermal_warn, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c new file mode 100644 index 0000000..3ba0a98 --- /dev/null +++ b/sound/soc/codecs/wm8997.c @@ -0,0 +1,1187 @@ +/* + * wm8997.c -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.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/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" +#include "wm8997.h" + +struct wm8997_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); + +static const struct reg_default wm8997_sysclk_reva_patch[] = { + { 0x301D, 0x7B15 }, + { 0x301B, 0x0050 }, + { 0x305D, 0x7B17 }, + { 0x305B, 0x0050 }, + { 0x3001, 0x08FE }, + { 0x3003, 0x00F4 }, + { 0x3041, 0x08FF }, + { 0x3043, 0x0005 }, + { 0x3020, 0x0225 }, + { 0x3021, 0x0A00 }, + { 0x3022, 0xE24D }, + { 0x3023, 0x0800 }, + { 0x3024, 0xE24D }, + { 0x3025, 0xF000 }, + { 0x3060, 0x0226 }, + { 0x3061, 0x0A00 }, + { 0x3062, 0xE252 }, + { 0x3063, 0x0800 }, + { 0x3064, 0xE252 }, + { 0x3065, 0xF000 }, + { 0x3116, 0x022B }, + { 0x3117, 0xFA00 }, + { 0x3110, 0x246C }, + { 0x3111, 0x0A03 }, + { 0x3112, 0x246E }, + { 0x3113, 0x0A03 }, + { 0x3114, 0x2470 }, + { 0x3115, 0x0A03 }, + { 0x3126, 0x246C }, + { 0x3127, 0x0A02 }, + { 0x3128, 0x246E }, + { 0x3129, 0x0A02 }, + { 0x312A, 0x2470 }, + { 0x312B, 0xFA02 }, + { 0x3125, 0x0800 }, +}; + +static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 0: + patch = wm8997_sysclk_reva_patch; + patch_size = ARRAY_SIZE(wm8997_sysclk_reva_patch); + break; + default: + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + default: + break; + } + + return 0; +} + +static const char *wm8997_osr_text[] = { + "Low power", "Normal", "High performance", +}; + +static const unsigned int wm8997_osr_val[] = { + 0x0, 0x3, 0x5, +}; + +static const struct soc_enum wm8997_hpout_osr[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), +}; + +#define WM8997_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \ + SOC_SINGLE(name " NG SPKOUT Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0) + +static const struct snd_kcontrol_new wm8997_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), + +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), + +SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), +SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21, + ARIZONA_EQ1_ENA_MASK), +SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21, + ARIZONA_EQ2_ENA_MASK), +SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21, + ARIZONA_EQ3_ENA_MASK), +SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21, + ARIZONA_EQ4_ENA_MASK), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), + +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUT", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_VALUE_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), +SOC_VALUE_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), + +SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), +SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +WM8997_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), +WM8997_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), +WM8997_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L), +WM8997_NG_SRC("SPKOUT", ARIZONA_NOISE_GATE_SELECT_4L), +WM8997_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L), +WM8997_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUT, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); + +static const char *wm8997_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "EPOUT", "SPKOUT", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm8997_aec_loopback_values[] = { + 0, 1, 4, 6, 8, 9, +}; + +static const struct soc_enum wm8997_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(wm8997_aec_loopback_texts), + wm8997_aec_loopback_texts, + wm8997_aec_loopback_values); + +static const struct snd_kcontrol_new wm8997_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm8997_aec_loopback); + +static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8997_aec_loopback_mux), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUT, "SPKOUT"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC", "AEC Loopback" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" } + +static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDD" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, + { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, + + { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, + { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, + + { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, + { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, + + { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, + { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUT"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MUX_ROUTES("ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2"), + + ARIZONA_MUX_ROUTES("ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2"), + + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "AEC Loopback", "EPOUT", "OUT3L" }, + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "AEC Loopback", "SPKOUT", "OUT4L" }, + { "SPKOUTN", NULL, "OUT4L" }, + { "SPKOUTP", NULL, "OUT4L" }, + + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "MICSUPP", NULL, "SYSCLK" }, +}; + +static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm8997_priv *wm8997 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM8997_FLL1: + return arizona_set_fll(&wm8997->fll[0], source, Fref, Fout); + case WM8997_FLL2: + return arizona_set_fll(&wm8997->fll[1], source, Fref, Fout); + case WM8997_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[0], source, Fref, + Fout); + case WM8997_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[1], source, Fref, + Fout); + default: + return -EINVAL; + } +} + +#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm8997_dai[] = { + { + .name = "wm8997-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-slim1", + .id = 3, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim2", + .id = 4, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim3", + .id = 5, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, +}; + +static int wm8997_codec_probe(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = priv->core.arizona->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + if (ret != 0) + return ret; + + arizona_init_spk(codec); + + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); + + priv->core.arizona->dapm = &codec->dapm; + + return 0; +} + +static int wm8997_codec_remove(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->core.arizona->dapm = NULL; + + return 0; +} + +#define WM8997_DIG_VU 0x0200 + +static unsigned int wm8997_digital_vu[] = { + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8997 = { + .probe = wm8997_codec_probe, + .remove = wm8997_codec_remove, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm8997_set_fll, + + .controls = wm8997_snd_controls, + .num_controls = ARRAY_SIZE(wm8997_snd_controls), + .dapm_widgets = wm8997_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8997_dapm_widgets), + .dapm_routes = wm8997_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8997_dapm_routes), +}; + +static int wm8997_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm8997_priv *wm8997; + int i; + + wm8997 = devm_kzalloc(&pdev->dev, sizeof(struct wm8997_priv), + GFP_KERNEL); + if (wm8997 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm8997); + + wm8997->core.arizona = arizona; + wm8997->core.num_inputs = 4; + + for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++) + wm8997->fll[i].vco_mult = 1; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm8997->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm8997->fll[1]); + + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + + for (i = 0; i < ARRAY_SIZE(wm8997_dai); i++) + arizona_init_dai(&wm8997->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm8997_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm8997_digital_vu[i], + WM8997_DIG_VU, WM8997_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997, + wm8997_dai, ARRAY_SIZE(wm8997_dai)); +} + +static int wm8997_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm8997_codec_driver = { + .driver = { + .name = "wm8997-codec", + .owner = THIS_MODULE, + }, + .probe = wm8997_probe, + .remove = wm8997_remove, +}; + +module_platform_driver(wm8997_codec_driver); + +MODULE_DESCRIPTION("ASoC WM8997 driver"); +MODULE_AUTHOR("Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8997-codec"); diff --git a/sound/soc/codecs/wm8997.h b/sound/soc/codecs/wm8997.h new file mode 100644 index 0000000..5e91c6a --- /dev/null +++ b/sound/soc/codecs/wm8997.h @@ -0,0 +1,23 @@ +/* + * wm8997.h -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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 _WM8997_H +#define _WM8997_H + +#include "arizona.h" + +#define WM8997_FLL1 1 +#define WM8997_FLL2 2 +#define WM8997_FLL1_REFCLK 3 +#define WM8997_FLL2_REFCLK 4 + +#endif -- 1.7.2.5

2 3

[alsa-devel] [PATCH 1/3] ASoC: max9877: Convert to use regmap API
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max9877.c | 187 ++++++++++++++++++++++++++++----------------- 1 file changed, 118 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 6b6c74c..7e2fe50 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -14,27 +14,21 @@ #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <sound/soc.h> #include <sound/tlv.h> #include "max9877.h" -static struct i2c_client *i2c; +static struct regmap *regmap; -static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; - -static void max9877_write_regs(void) -{ - unsigned int i; - u8 data[6]; - - data[0] = MAX9877_INPUT_MODE; - for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) - data[i + 1] = max9877_regs[i]; - - if (i2c_master_send(i2c, data, 6) != 6) - dev_err(&i2c->dev, "i2c write failed\n"); -} +static struct reg_default max9877_regs[] = { + { 0, 0x40 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x49 }, +}; static int max9877_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -45,8 +39,14 @@ static int max9877_get_reg(struct snd_kcontrol *kcontrol, unsigned int shift = mc->shift; unsigned int mask = mc->max; unsigned int invert = mc->invert; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = @@ -65,18 +65,21 @@ static int max9877_set_reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int invert = mc->invert; unsigned int val = (ucontrol->value.integer.value[0] & mask); + bool change; + int ret; if (invert) val = mask - val; - if (((max9877_regs[reg] >> shift) & mask) == val) - return 0; - - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_write_regs(); + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change); + if (ret != 0) + return ret; - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_2reg(struct snd_kcontrol *kcontrol, @@ -88,9 +91,18 @@ static int max9877_get_2reg(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; unsigned int mask = mc->max; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[0] = (val >> shift) & mask; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; - ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask; + ret = regmap_read(regmap, reg2, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[1] = (val >> shift) & mask; return 0; } @@ -106,77 +118,99 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 0; - - if (((max9877_regs[reg] >> shift) & mask) != val) - change = 1; - - if (((max9877_regs[reg2] >> shift) & mask) != val2) - change = 1; - - if (change) { - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_regs[reg2] &= ~(mask << shift); - max9877_regs[reg2] |= val2 << shift; - max9877_write_regs(); - } - - return change; + bool change1, change2; + int ret; + + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change1); + if (ret != 0) + return ret; + + ret = regmap_update_bits_check(regmap, reg2, mask << shift, + val2 << shift, &change2); + if (ret != 0) + return ret; + + if (change1 || change2) + return 1; + else + return 0; } static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OUTMODE_MASK; + if (val) + val--; - if (value) - value -= 1; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; + unsigned int val; + bool change; + int ret; - value += 1; + val = ucontrol->value.integer.value[0] + 1; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) - return 0; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OUTMODE_MASK, val, &change); + if (ret != 0) + return ret; - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OSC_MASK; + val >>= MAX9877_OSC_OFFSET; - value = value >> MAX9877_OSC_OFFSET; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; - - value = value << MAX9877_OSC_OFFSET; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + unsigned int val; + bool change; + int ret; + + val = ucontrol->value.integer.value[0] << MAX9877_OSC_OFFSET; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OSC_MASK, val, &change); + if (ret != 0) + return ret; + + if (change) + return 1; + else return 0; - - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; } static const unsigned int max9877_pgain_tlv[] = { @@ -258,19 +292,34 @@ int max9877_add_controls(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(max9877_add_controls); +static const struct regmap_config max9877_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = max9877_regs, + .num_reg_defaults = ARRAY_SIZE(max9877_regs), + .cache_type = REGCACHE_RBTREE, +}; + static int max9877_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - i2c = client; + int i; + + regmap = devm_regmap_init_i2c(client, &max9877_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - max9877_write_regs(); + /* Ensure the device is in reset state */ + for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) + regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def); return 0; } static int max9877_i2c_remove(struct i2c_client *client) { - i2c = NULL; + regmap = NULL; return 0; } -- 1.8.4.rc2

1 2

[alsa-devel] [PATCH v2] extcon: arizona: Clear trig_sts bits on all paths
by Charles Keepax 13 Aug '13

13 Aug '13

We want the trig_sts bits to be cleared in all cases where we consider the jack detection interrupt to have been handled. Specifically, if a duplicate detection event was suppressed these bits were not cleared causing the CODEC to not enter a low power state. This patch clears the bits on the duplicate detection code path. Reported-by: Ryo Tsutsui <ryo.tsutsui(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/extcon/extcon-arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/extcon/extcon-arizona.c b/drivers/extcon/extcon-arizona.c index 7a1b4a7..4df68de 100644 --- a/drivers/extcon/extcon-arizona.c +++ b/drivers/extcon/extcon-arizona.c @@ -1034,6 +1034,7 @@ static irqreturn_t arizona_jackdet(int irq, void *data) else info->micd_timeout = DEFAULT_MICD_TIMEOUT; +out: /* Clear trig_sts to make sure DCVDD is not forced up */ regmap_write(arizona->regmap, ARIZONA_AOD_WKUP_AND_TRIG, ARIZONA_MICD_CLAMP_FALL_TRIG_STS | @@ -1041,7 +1042,6 @@ static irqreturn_t arizona_jackdet(int irq, void *data) ARIZONA_JD1_FALL_TRIG_STS | ARIZONA_JD1_RISE_TRIG_STS); -out: mutex_unlock(&info->lock); pm_runtime_mark_last_busy(info->dev); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] Add PM support to the rme96 driver
by Knut Petersen 13 Aug '13

13 Aug '13

Hi everybody! The attached patch adds PM support to the rme96 driver, it should be applied after v2 of the patch adding pcm stream synchronization support posted on 2013-08-08. It does work perfectly here on an RME Digi96/8 PAD: After connecting analog output and analog input executing #! /bin/sh # arecord -D hw:0,0 -d 62 testout.wav & aplay -D hw:0,0 -d 60 testin.wav produces an almost perfect recording even when the test machine is suspended / hibernated and resumed several times. cu, Knut

2 2

[alsa-devel] [PATCH] ASoC: dapm: Ensure kcontrol list is initialised
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure that the recently added path kcontrol list is initialised otherwise we may crash trying to delete routes that don't have kcontrols. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b78eaa1..1e0e4ce 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2357,6 +2357,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, path->sink = wsink; path->connected = connected; INIT_LIST_HEAD(&path->list); + INIT_LIST_HEAD(&path->list_kcontrol); INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); -- 1.8.4.rc1

2 1

[alsa-devel] [PATCH] ASoC: dapm: Don't create routes when creating kcontrols
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Attempting to create the route as part of adding a mux control causes us to attempt to add the same route twice since we loop over all sources for the mux after creating the control. Instead do the addition in the callers. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-dapm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 662a904..b78eaa1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -665,7 +665,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, * create it. Either way, add the widget into the control's widget list */ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, - int kci, struct snd_soc_dapm_path *path) + int kci) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; @@ -766,7 +766,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, return ret; w->kcontrols[kci] = kcontrol; - dapm_kcontrol_add_path(kcontrol, path); return 0; } @@ -790,9 +789,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) continue; } - ret = dapm_create_or_share_mixmux_kcontrol(w, i, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, i); if (ret < 0) return ret; + + dapm_kcontrol_add_path(w->kcontrols[i], path); } } @@ -821,7 +822,7 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) path = list_first_entry(&w->sources, struct snd_soc_dapm_path, list_sink); - ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0); if (ret < 0) return ret; -- 1.8.4.rc1

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[alsa-devel] [PATCH 1/2] ALSA: hda - Allow auto_mute_via_amp on bind mute controls
by Takashi Iwai 13 Aug '13

13 Aug '13

The auto-mute using the amp currently works only for a single amp on a pin. Make it working also with HDA_CTL_BIND_MUTE type, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f6c0344..6ed2209 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -816,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx) static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); enum { HDA_CTL_WIDGET_VOL, @@ -833,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = { .put = hda_gen_mixer_mute_put, /* replaced */ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), }, - HDA_BIND_MUTE(NULL, 0, 0, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_bind_switch_get, + .put = hda_gen_bind_mute_put, /* replaced */ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), + }, }; /* add dynamic controls from template */ @@ -940,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx, } /* playback mute control with the software mute bit check */ -static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gen_spec *spec = codec->spec; @@ -952,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] &= enabled; ucontrol->value.integer.value[1] &= enabled; } +} +static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); + return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol); +} + /* any ctl assigned to the path with the given index? */ static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type) { -- 1.8.3.1

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[alsa-devel] [PATCH 1/3] ASoC: wm5102: Add inputs for noise and mic mixers
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> The noise and mic mixer inputs were not connected, do so. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f38c52d..8bbddc1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1423,9 +1423,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -1552,6 +1549,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), -- 1.8.4.rc1

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[alsa-devel] [PATCH v2 0/2] ALSA: Implement core jack support for kcontrol
by Felipe F. Tonello 12 Aug '13

12 Aug '13

From: "Felipe F. Tonello" <eu(a)felipetonello.com> On this second version I made sure that the patch series can be bisected. Also, since HDA codec has it's own jack infrastructure, I just disabled the support of the core ALSA jack in it. Why? Because it's causing duplication of controls, although the input events works fine. A HDA codec developer might give us more information about it. Patch 1) Is the actual implementation and fixes for users that uses the API. Patch 2) Name fixes. Just to make sure no jack has " Jack" duplicated at the end of its name. Felipe F. Tonello (2): ALSA: Added jack detection KControl support ALSA: SoC: Fix jack names according new API include/sound/control.h | 2 +- include/sound/jack.h | 8 ++-- include/sound/soc.h | 2 +- sound/core/Kconfig | 1 + sound/core/ctljack.c | 15 ++++++- sound/core/jack.c | 88 +++++++++++++++++++++++++++++++++++--- sound/pci/hda/Kconfig | 8 ---- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 4 +- sound/soc/mid-x86/mfld_machine.c | 6 +-- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 4 +- sound/soc/omap/omap-twl4030.c | 4 +- sound/soc/omap/rx51.c | 6 +-- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/palm27x.c | 4 +- sound/soc/pxa/ttc-dkb.c | 10 ++--- sound/soc/pxa/z2.c | 4 +- sound/soc/samsung/goni_wm8994.c | 7 +-- sound/soc/samsung/h1940_uda1380.c | 4 +- sound/soc/samsung/littlemill.c | 10 ++--- sound/soc/samsung/lowland.c | 6 +-- sound/soc/samsung/rx1950_uda1380.c | 4 +- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/speyside.c | 6 +-- sound/soc/samsung/tobermory.c | 4 +- sound/soc/soc-jack.c | 5 ++- sound/soc/tegra/tegra_alc5632.c | 4 +- sound/soc/tegra/tegra_rt5640.c | 4 +- sound/soc/tegra/tegra_wm8903.c | 8 ++-- 30 files changed, 163 insertions(+), 77 deletions(-) -- 1.8.3.1

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[alsa-devel] [RFC PATCH] ASoC: omap-mcbsp: Add support for highter rate up to 384000
by Michael Trimarchi 12 Aug '13

12 Aug '13

Add support for highter rate up to 384000Khz. There are codecs in the market that can play up to this frequency. Test was done just with: #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_192000) playing wav file 32bit/192000Khz stereo. Some glitch has observed but this should depend on the selected min buffer bytes that was left untouched in aplay and in the omap-pcm driver. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- sound/soc/omap/omap-mcbsp.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb..1e95f6a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -39,7 +39,9 @@ #include "mcbsp.h" #include "omap-mcbsp.h" -#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS | \ + SNDRV_PCM_RATE_KNOT) #define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ xhandler_get, xhandler_put) \ @@ -574,11 +576,15 @@ static struct snd_soc_dai_driver omap_mcbsp_dai = { .channels_max = 16, .rates = OMAP_MCBSP_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 384000, }, .capture = { .channels_min = 1, .channels_max = 16, .rates = OMAP_MCBSP_RATES, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, }, .ops = &mcbsp_dai_ops, -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] ASoC: dt: Move ALC5632 pin list from Tegra board binding to CODEC binding
by Mark Brown 12 Aug '13

12 Aug '13

From: Mark Brown <broonie(a)linaro.org> The pin list is the same for any board using the CODEC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../devicetree/bindings/sound/alc5632.txt | 20 ++++++++++++++++++ .../bindings/sound/nvidia,tegra-audio-alc5632.txt | 24 ++-------------------- 2 files changed, 22 insertions(+), 22 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt index 8608f74..b913b74 100644 --- a/Documentation/devicetree/bindings/sound/alc5632.txt +++ b/Documentation/devicetree/bindings/sound/alc5632.txt @@ -13,6 +13,26 @@ Required properties: - #gpio-cells : Should be two. The first cell is the pin number and the second cell is used to specify optional parameters (currently unused). +Pins on the device (for linking into audio routes): + + * SPK_OUTP + * SPK_OUTN + * HP_OUT_L + * HP_OUT_R + * AUX_OUT_P + * AUX_OUT_N + * LINE_IN_L + * LINE_IN_R + * PHONE_P + * PHONE_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MICBIAS1 + * DMICDAT + + Example: alc5632: alc5632@1e { diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt index 05ffecb..8b8903e 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt @@ -11,28 +11,8 @@ Required properties: - nvidia,audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources and - sinks are the ALC5632's pins: - - ALC5632 pins: - - * SPK_OUTP - * SPK_OUTN - * HP_OUT_L - * HP_OUT_R - * AUX_OUT_P - * AUX_OUT_N - * LINE_IN_L - * LINE_IN_R - * PHONE_P - * PHONE_N - * MIC1_P - * MIC1_N - * MIC2_P - * MIC2_N - * MICBIAS1 - * DMICDAT - - Board connectors: + sinks are the ALC5632's pins as documented in the binding for the device + and: * Headset Stereophone * Int Spk -- 1.8.4.rc1

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Re: [alsa-devel] [PATCH v3 5/5] ALSA: snd-usb: add quirks handler for DSD streams
by Daniel Mack 12 Aug '13

12 Aug '13

On 12.08.2013 13:27, Jesus R wrote: > On Mon, Aug 12, 2013 at 4:52 AM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > Right now MPD supports DSD/DoP with one selectable on/off option. On > equals DoP output and off equals conversion to PCM. I'm not aware of any > case where it does not work. I would seem to me that there is one > predominant format? Let me clarify that again: I read the above as the two options are a) DSD material is augmented by DOP markers and send as such over a regular PCM interface, or b) convert DSD to PCM in software and then send PCM samples over a regular PCM interface. What applications need to use for my new approach is another is: c) take DSD samples, bit reverse them if necessary, and then send them over the generic DSD data pipe to the Linux kernel. Device-specific details are taken care of by the kernel layer. I'm not aware of any software that uses that new DSD format type directly except for my example program. Daniel

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Re: [alsa-devel] [PATCH v3 5/5] ALSA: snd-usb: add quirks handler for DSD streams
by Daniel Mack 12 Aug '13

12 Aug '13

[please never drop people from the Cc: list but always use 'reply to all'!] On 12.08.2013 13:27, Jesus R wrote: > On Mon, Aug 12, 2013 at 4:52 AM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > Given than none of devices have a chance to report their capabilities to > the host, finding out which modes to use is a trial-and-error game, > unless you have access to the firmware sources. > > > Hence, please just look up the USB IDs and alternate interface settings > of your connected device with lsusb, and then augment the code in > snd_usb_interface_dsd_format_quirks() (file sound/usb/quirks.c) and play > around with different possibilities (bitreverse on/off, DSD_U8 vs > DSD_U16 etc). Follow the example that I provided in that function. > > Right now MPD supports DSD/DoP with one selectable on/off option. On > equals DoP output and off equals conversion to PCM. I'm not aware of any > case where it does not work. I would seem to me that there is one > predominant format? It's more complicated than that. Applications were always free to use a usual PCM stream and stuff it with DOP/DSD samples. This is the whole idea of DOP: allow applications to transport DSD samples via regular audio streams. The downside of this is that applications are in charge of adding the DOP markers manually, and information about bit-reversing and marker format (8 bit vs. 16 bit) is up to the application, which is not the right place for it. The patch series I brought in addressed this in a way that all that information about device-specific details is now known to the kernel, which can hence offer a standard compliant interface to userspace. [There's yet one more level of confusion as some file formats store the samples bit-reversed, but there's nothing we can do about that from kernel space except for defining a stable interface application have to obey to in order to be compatible. That is, samples are expected to always carry the oldest bit in the most significant bit of a byte.] So my question is: are you 100% sure that MPD uses the newly introduces ALSA formats, or are they doing the DOP conversion in manually in their own software layers? > You have to recompile and rmmod/insmod the snd-usb driver every time you > make a change, and then you need some sort of userspace tool to test it. > I published one here: > > This may be more than I can do. Can you add the options (bitreverse > on/off, DSD_U8 vs > DSD_U16 etc) to the alas-dsd-player? This would allow me to quickly test > the devices. There are #if0'ed blocks that you can opt-in for bit reversal, but as the marker format is different for 8 and 16 bits, and up to the kernel space now, the dsd player does not know about them. That was the entire intention of bringing that logic to kernel space in the first place. > Take a look at the DSD database so you can see what we are up against. I > could even add the tests results to the database. > https://docs.google.com/spreadsheet/ccc?key=0AgVhKcl_3lHfdFVyenBBNjNpQ2lieG… Nice list. Only devics with USB interface are interesting at this point, but that's a good start. As I said: you need to add the quirks for the devices to the quirks list. Other than that, there's no chance to test them. Copy'n pasting code inside that function and recompiling the kernel is really not as hard as you might think it is. Just search the internet for "compile kernel git <your-distribution-here>" and follow the links. Thanks, Daniel

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[alsa-devel] [PATCH V4 0/4] Add i2s support on smdk5420
by Padmavathi Venna 12 Aug '13

12 Aug '13

Samsung has different versions of I2S introduced in different platforms. Each version has some new support added for multichannel, secondary fifo, s/w reset control, internal mux for rclk src clk and tdm support. Each newly added change has a quirk. So this patch adds all the required quirks as driver data and based on compatible string from dtsi fetches the quirks. This also adds i2s support on exynos5420. Changes since V3: - Addressed review comments by Tomasz Figa related to const qualifier for samsung_i2s_dai_data - Removed passing quirks as driver data for non-dt platforms. - Separated out adding i2s nodes and enabling audio support on 5420 into different patch set as they are dependent on some of already posted but not yet merged i2c, dwmmc, dma and audss clock controller patches. Changes since V2: - Separated out driver side changes and dts changes in two patch sets. - Replaced samsung,s3c6410-i2s-v4 with samsung,s3c6410-i2s-multi for more clarity as suggested by Tomasz Figa. Changes since V1: - Pass quirks as driver data and fetch the quirks based on compatible string from dtsi file as suggested by Tomasz Figa and Mark Brown - Make the I2S driver more flexible with respect to register access as suggested by Tomasz Figa and Mark Brown - Add 5420 support in the driver. - Modify the dtsi files with the corresponding compatible strings and removed the i2s quirks from 5250 dtsi file. - Updated the i2s Documentation with relevent changes and i2s versioning info. - Add i2s nodes on exynos5420.dtsi - Enable sound support on smdk5420 This patch set is made based on Mark Brown for-next branch on sound.git. Padmavathi Venna (4): ASoC: Samsung: I2S: Add quirks as driver data in I2S ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 ARM: dts: exynos5250: move common i2s properties to exynos5 dtsi ARM: dts: Change i2s compatible string on exynos5250 .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++-- arch/arm/boot/dts/exynos5.dtsi | 21 +++ arch/arm/boot/dts/exynos5250.dtsi | 17 +--- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 15 ++ sound/soc/samsung/i2s.c | 143 +++++++++++++++----- 6 files changed, 157 insertions(+), 62 deletions(-) -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: arizona: Add widget<->mux route into mux route macro
by Charles Keepax 12 Aug '13

12 Aug '13

The routes linking the widget and the input mux were being added manually, rather than by the ARIZONA_MUX_ROUTES macro. This patchs adds the routes to the macro. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 3 ++- sound/soc/codecs/wm5102.c | 41 ++++++++++++----------------------------- sound/soc/codecs/wm5110.c | 8 ++++---- 3 files changed, 18 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b60b08c..bf67d65 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -150,7 +150,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MUX(name_str " Aux 5", &name##_aux5_mux), \ ARIZONA_MUX(name_str " Aux 6", &name##_aux6_mux) -#define ARIZONA_MUX_ROUTES(name) \ +#define ARIZONA_MUX_ROUTES(widget, name) \ + { widget, NULL, name " Input" }, \ ARIZONA_MIXER_INPUT_ROUTES(name " Input") #define ARIZONA_MIXER_ROUTES(widget, name) \ diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 282fd23..9e9a0c1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1499,23 +1499,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "IN3L PGA", NULL, "IN3L" }, { "IN3R PGA", NULL, "IN3R" }, - { "ASRC1L", NULL, "ASRC1L Input" }, - { "ASRC1R", NULL, "ASRC1R Input" }, - { "ASRC2L", NULL, "ASRC2L Input" }, - { "ASRC2R", NULL, "ASRC2R Input" }, - - { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, - { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, - - { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, - { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, - - { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, - { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, - - { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, - { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, - ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -1567,22 +1550,22 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), - ARIZONA_MUX_ROUTES("ISRC1INT1"), - ARIZONA_MUX_ROUTES("ISRC1INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), - ARIZONA_MUX_ROUTES("ISRC1DEC1"), - ARIZONA_MUX_ROUTES("ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), - ARIZONA_MUX_ROUTES("ISRC2INT1"), - ARIZONA_MUX_ROUTES("ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), - ARIZONA_MUX_ROUTES("ISRC2DEC1"), - ARIZONA_MUX_ROUTES("ISRC2DEC2"), + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), ARIZONA_DSP_ROUTES("DSP1"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e7cb4b..0cfc5e6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -979,10 +979,10 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, -- 1.7.2.5

2 1

[alsa-devel] Another possible race in deferred probing...
by Russell King - ARM Linux 12 Aug '13

12 Aug '13

There is no better way to provoke bugs than to show a device to someone who has never seen it before. Obviously, I didn't get much chance to investigate what happened, so all I have are the kernel messages: kirkwood-spdif-audio kirkwood-spdif-audio.1: ASoC: CPU DAI mvebu-audio.1 not registered kirkwood-spdif-audio kirkwood-spdif-audio.1: failed to register card platform kirkwood-spdif-audio.1: Driver kirkwood-spdif-audio requests probe deferral mvebu-audio mvebu-audio.1: found external clock kirkwood-spdif-audio kirkwood-spdif-audio.1: ASoC: CODEC spdif-dit not registered kirkwood-spdif-audio kirkwood-spdif-audio.1: failed to register card platform kirkwood-spdif-audio.1: Driver kirkwood-spdif-audio requests probe deferral kirkwood-spdif-audio kirkwood-spdif-audio.1: ASoC: CODEC spdif-dit not registered kirkwood-spdif-audio kirkwood-spdif-audio.1: failed to register card platform kirkwood-spdif-audio.1: Driver kirkwood-spdif-audio requests probe deferral and then when manually removing the kirkwood-spdif module and re-inserting it: kirkwood-spdif-audio kirkwood-spdif-audio.1: dit-hifi <-> mvebu-audio.1 mapping ok First, remember that Ubuntu loads kernel modules from separate threads, so module loading here is multithreaded. What I suspect happened was the spdif-dit module was inserted and initialized between kirkwood-spdif starting to be re-probed, and returning from its probe function with -EPROBE_DEFER, and I suspect that spdif-dit was the last module. As there was no further module loading or driver activity, there was no attempt to re-probe the kirkwood-spdif module. This seems to be a very rare event (it's the first time I've seen it since the last fix to the deferred probing, and that's many hundreds of reboots.) Nevertheless, the fact that it has happened indicates that deferred probing still has holes that can leave modular drivers unbound after boot. Let's consider this scenario, which from a quick scan of the code looks like it's possible to happen: Thread 0 Thread 1 deferred probe triggered driver A's probe function called (kirkwood_spdif_probe in my case) takes lock L to lookup resource A checks for resource A (the spdif_dit codec) releases lock L resource A is not found ----- context switch ------ module init function called (to register spdif_dit_driver) driver B's probe function called (to register the spdif_dit codec via snd_soc_register_codec) takes lock L for adding resource A registers resource A releases lock L driver B's probe function completes deferred probe triggered, deferred probe list empty module init completes ----- context switch ------ probe returns -EPROBE_DEFER deferred probe mutex taken added to deferred probe list deferred probe mutex released This can happen because there's no locking between looking up the resource and having the driver added to the list of devices waiting to be probed. Given that deferred probing is becoming the "normal" thing for device drivers, these races are only going to get worse. Consider that as we're working on ARM towards having a single zImage bootable on many boards, most drivers will be modules, and if they're inserted into the kernel in a multi-threaded way, the above race is going to continually come back and bite us. I don't see an easy solution to this; we can't serialise device probes, because we have some places where device driver probe functions register devices themselves, which then go on to have their drivers probed (nested probes). Maybe this reflects back into Greg's whole argument against platform devices: maybe we need a struct device which lists _all_ the resources which a device requires, have the bus layer safely check for the presence of all those resources before calling the probe function, and move the deferred probe out of the core into the bus layer(s). Another solution would be to have a way to attach to the device a list of resources which failed, and have the driver core scan that list after the probe returns rechecking to see if all the resources are now present. That to me sounds particularly horrid, invasive and quite disgusting to implement safely.

1 0

[alsa-devel] [PATCH] ALSA: compress: update the MAINTAINER entry
by Vinod Koul 12 Aug '13

12 Aug '13

add missing uapi/ headers and Documentation files Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- MAINTAINERS | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 38a870e..a40e393 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7667,7 +7667,9 @@ M: Vinod Koul <vinod.koul(a)intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Supported +F: Documentation/sound/alsa/compress_offload.txt F: include/sound/compress_driver.h +F: include/uapi/sound/compress_* F: sound/core/compress_offload.c F: sound/soc/soc-compress.c -- 1.7.0.4

2 1

[alsa-devel] [PATCH 2/2] 6fire: make buffers DMA-able (midi)
by Torsten Schenk 12 Aug '13

12 Aug '13

Patch makes midi output buffer DMA-able by allocating it separately. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> --- diff -Nur a/sound/6fire/midi.c b/sound/6fire/midi.c --- a/sound/6fire/midi.c 2013-08-04 22:46:46.000000000 +0200 +++ b/sound/6fire/midi.c 2013-08-11 10:51:05.682482388 +0200 @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff -Nur a/sound/6fire/midi.h b/sound/6fire/midi.h --- a/sound/6fire/midi.h 2013-08-04 22:46:46.000000000 +0200 +++ b/sound/6fire/midi.h 2013-08-11 10:51:05.682482388 +0200 @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length);

2 1

[alsa-devel] [PATCH] 6fire: fix URB transfer buffer for midi output
by Torsten Schenk 12 Aug '13

12 Aug '13

Patch fixes URB transfer buffer allocation for midi output to be DMA-able. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> --- diff -Nur a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c --- a/sound/usb/6fire/midi.c 2013-08-07 16:32:10.579639391 +0200 +++ b/sound/usb/6fire/midi.c 2013-08-07 16:32:31.363378104 +0200 @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff -Nur a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h --- a/sound/usb/6fire/midi.h 2013-08-07 16:32:10.579639391 +0200 +++ b/sound/usb/6fire/midi.h 2013-08-07 16:32:31.363378104 +0200 @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length);

4 7

[alsa-devel] LG LW25 pin fix
by Joonas Saarinen 12 Aug '13

12 Aug '13

Hello, I discovered that the LG LW25 laptop needs a small fix to its pin mapping. The functions of blue and green jacks simply have to be interchanged. The rest of the configuration is correct. Can this fix be made permanent in the driver? The fixed mappings are: 0x1a 0x0181344f 0x1b 0x0321403f Here's the full correct map: [codec] 0x10ec0880 0x00001854 0 [pincfg] 0x14 0xb7031110 0x15 0x411111f0 0x16 0x411111f0 0x17 0x411111f0 0x18 0x01a19c22 0x19 0xb7a31120 0x1a 0x0181344f 0x1b 0x0321403f 0x1c 0x411111f0 0x1d 0x411111f0 0x1e 0x0145311e 0x1f 0x411111f0 Thanks, Joonas

2 5

Re: [alsa-devel] [PATCH v3 5/5] ALSA: snd-usb: add quirks handler for DSD streams
by Jesus R 12 Aug '13

12 Aug '13

Daniel/team, I have access to a few DSD devices and or USB based devices. I can provide any information you need to help with allowing ALSA to identify the device's capabilities. Just let me know how to get the information you need from command line:) I also have a list of known DSD/DoP devices in Google Drive that may be helpful. Let me know if it's okay to post the link. Jesus R sonore.us

2 2

[alsa-devel] [PATCH] ASoC: adau1701: Use gpio_set_value_cansleep()
by Mark Brown 12 Aug '13

12 Aug '13

From: Mark Brown <broonie(a)linaro.org> The GPIO manipulation done by this driver is never in atomic context so we can use gpio_set_value_cansleep() and support GPIOs that can't be set from atomic context. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/adau1701.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 2c10252..ebff112 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -247,21 +247,21 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) gpio_is_valid(adau1701->gpio_pll_mode[1])) { switch (clkdiv) { case 64: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 256: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; case 384: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 0: /* fallback */ case 512: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; } } @@ -269,10 +269,10 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) adau1701->pll_clkdiv = clkdiv; if (gpio_is_valid(adau1701->gpio_nreset)) { - gpio_set_value(adau1701->gpio_nreset, 0); + gpio_set_value_cansleep(adau1701->gpio_nreset, 0); /* minimum reset time is 20ns */ udelay(1); - gpio_set_value(adau1701->gpio_nreset, 1); + gpio_set_value_cansleep(adau1701->gpio_nreset, 1); /* power-up time may be as long as 85ms */ mdelay(85); } -- 1.8.4.rc1

2 1

[alsa-devel] [PATCH] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 12 Aug '13

12 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Add mute/unmute control in widget event and modify the power on/off sequence to avoid pop noise. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Please ignore the previous patch. I have tested speaker and headphone playback on this patch. --- sound/soc/codecs/rt5640.c | 389 +++++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5640.h | 15 ++ 2 files changed, 352 insertions(+), 52 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..cae3f46 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -378,21 +376,119 @@ static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; +static const char * const rt5640_mute_mode[] = {"mute", "unmute"}; + +static const SOC_ENUM_SINGLE_DECL(rt5640_hp_enum, 0, 0, rt5640_mute_mode); +static const SOC_ENUM_SINGLE_DECL(rt5640_spo_enum, 0, 0, rt5640_mute_mode); + static const SOC_ENUM_SINGLE_DECL( rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); +static int rt5640_spo_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->spo_l_mute; + ucontrol->value.integer.value[1] = !rt5640->spo_r_mute; + + return 0; +} + +static int rt5640_spo_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_DIG1) & RT5640_PWR_CLS_D; + + if (ucontrol->value.integer.value[0]) { + rt5640->spo_l_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->spo_l_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, + RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->spo_r_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->spo_r_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + } + + return 0; +} + +static int rt5640_hp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->hp_l_mute; + ucontrol->value.integer.value[1] = !rt5640->hp_r_mute; + + return 0; +} + +static int rt5640_hp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_ANLG1); + + if (ucontrol->value.integer.value[0]) { + rt5640->hp_l_mute = false; + if (pow & RT5640_PWR_HP_L) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->hp_l_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->hp_r_mute = false; + if (pow & RT5640_PWR_HP_R) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->hp_r_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, RT5640_R_MUTE); + } + + + return 0; +} + static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("Speaker Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_spo_get, rt5640_spo_put), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("HP Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_hp_get, rt5640_hp_put), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -868,33 +964,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +1012,216 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_spo_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_r_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_spo_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_l_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, + RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | + RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(20000, 25000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, + RT5640_PWR_FV1 | RT5640_PWR_FV2); + + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CO_MASK | + RT5640_HP_SG_MASK, + RT5640_HP_CO_EN | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, pow; + + regmap_read(rt5640->regmap, RT5640_PWR_ANLG1, &pow); + if (rt5640->hp_l_mute || !(pow & RT5640_PWR_HP_L)) + val |= RT5640_L_MUTE; + if (rt5640->hp_r_mute || !(pow & RT5640_PWR_HP_R)) + val |= RT5640_R_MUTE; + /* headphone unmute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK, RT5640_SMT_TRIG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK, RT5640_RSTN_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_RSTN_DIS | RT5640_HP_L_SMT_EN | + RT5640_HP_R_SMT_EN); + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, val); + usleep_range(40000, 45000); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | RT5640_HP_R_SMT_DIS); + +} + +static void rt5640_pmd_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* headphone mute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK, RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK, RT5640_RSTP_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_RSTP_DIS | + RT5640_HP_L_SMT_EN | RT5640_HP_R_SMT_EN); + + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, + RT5640_L_MUTE | RT5640_R_MUTE); + usleep_range(40000, 45000); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | + RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | + RT5640_HP_R_SMT_DIS); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK | + RT5640_HP_CD_PD_MASK | + RT5640_HP_CO_MASK | + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_SMT_TRIG_DIS | + RT5640_HP_CD_PD_EN | + RT5640_HP_CO_DIS | RT5640_HP_CP_PD | + RT5640_HP_SG_EN | RT5640_HP_CB_PD); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5640_pmd_depop(codec); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1411,23 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("SPO L Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_l_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("SPO R Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_r_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1396,13 +1683,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"SPO L Amp", NULL, "SPOL MIX"}, + {"SPO R Amp", NULL, "SPOR MIX"}, + {"SPOLP", NULL, "SPO L Amp"}, + {"SPOLN", NULL, "SPO L Amp"}, + {"SPORP", NULL, "SPO R Amp"}, + {"SPORN", NULL, "SPO R Amp"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,7 +1701,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, + {"HPOL", NULL, "HP Amp"}, {"HPOL", NULL, "HP L Amp"}, + {"HPOR", NULL, "HP Amp"}, {"HPOR", NULL, "HP R Amp"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, @@ -1792,17 +2083,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(15000, 20000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +2133,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..cb6bce9 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) @@ -2087,6 +2098,10 @@ struct rt5640_priv { int pll_out; int dmic_en; + bool spo_l_mute; + bool spo_r_mute; + bool hp_l_mute; + bool hp_r_mute; }; #endif -- 1.8.1.1.439.g50a6b54

5 26

[alsa-devel] [PATCH] ASoC: compress: Use power efficient workqueue
by Mark Brown 12 Aug '13

12 Aug '13

From: Mark Brown <broonie(a)linaro.org> There is no need for the power down work to be done on a per CPU workqueue especially considering the fairly long delay before powerdown. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-compress.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index d220150..53c9ecd 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -149,8 +149,9 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } else { rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ -- 1.8.4.rc1

2 1

[alsa-devel] BeagleBoard driver development (PCM3168)
by Michael Jordan 12 Aug '13

12 Aug '13

Okay, before I went too far I thought to download archives and found that Klimann has been working on this, but last post I see is April 2013. I picked through and got what I think are the last discussed of the 3 files omap-pcm3168.c pcm3168.c and pcm3168.h. I did not mean to cross paths with anybody. I will compare these to what I have worked up and see what I can contribute. I am a bit confused about one thing though. When I began this (before I knew I might be stepping on somebody else's work) I downloaded 3.10 sources from kernel.org. I have previously and again grepped through the sources and I find no mention of pcm3168. Has this project just not made it that far? And do I have the latest versions (plucked from the text archives, or are they in a git somewhere that I don't know about? Michael

1 0

[alsa-devel] Hello
by Michael Jordan 11 Aug '13

11 Aug '13

Hello all, I am involved in a university project using a TI OMAPL138 EXPKIT (Logic PD Zoom Board) to create a sound processing box. I am tasked with developing the code to replace the TLV320AIC3106 with a PCM3168A due to the greater ADC/DAC count. We have already accomplished the hardware end, and I have been exercising my linux skills at trying to put this together. I found this list and subscribed, and I am glad I sat and listened for a bit before I said anything. 457 patches in the last 10 days! My fingers would fall off! I will get together what I have been working on (2 files pcm3168a.c and pcm3168a.h) and post them. I have some questions, obviously, but I will follow up with them in a bit. If someone could help me along it would be greatly appreciated. Or if I can just provide some relevant information, I would be happy to do that as well. Thanks, Michael

1 0

[alsa-devel] [PATCH] ASoC: doc: Add documentation to MAINTAINERS patterns
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 427a930..07e2bc8 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7710,6 +7710,7 @@ T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC S: Supported +F: Documentation/sound/alsa/soc/ F: sound/soc/ F: include/sound/soc* -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH 1/16] sound/soc/pxa/mioa701_wm9713.c: Avoid using ARRAY_AND_SIZE(e) as a function argument
by Julia Lawall 11 Aug '13

11 Aug '13

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Replace ARRAY_AND_SIZE(e) in function argument position to avoid hiding the arity of the called function. The semantic match that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e,f; @@ f(..., - ARRAY_AND_SIZE(e) + e,ARRAY_SIZE(e) ,...) // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not compiled. sound/soc/pxa/mioa701_wm9713.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 97b711e..20fdce6 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -133,10 +133,11 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) unsigned short reg; /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets, + ARRAY_SIZE(mioa701_dapm_widgets)); /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG);

2 1

[alsa-devel] [PATCH] ASoC: cs4271: fix dapm routing table
by Daniel Mack 11 Aug '13

11 Aug '13

The widgets are names "AINA" and "AINB", which is also what the datasheets calls the terminals. Fix the routing table to match these names. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 35a6a67..a20f1bb 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -184,8 +184,8 @@ SND_SOC_DAPM_OUTPUT("AOUTB-"), }; static const struct snd_soc_dapm_route cs4271_dapm_routes[] = { - { "Capture", NULL, "AINL" }, - { "Capture", NULL, "AINR" }, + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, { "AOUTA+", NULL, "Playback" }, { "AOUTA-", NULL, "Playback" }, -- 1.8.3.1

2 1

[alsa-devel] [PATCH] usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525
by Maksim A. Boyko 11 Aug '13

11 Aug '13

Add the volume control quirk for avoiding the kernel warning for the Logitech HD Webcam C525 as in the similar commit 36691e1be6ec551eef4a5225f126a281f8c051c2 (ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310) for the Logitech HD Webcam C310. Reported-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Tested-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Cc: <stable(a)vger.kernel.org> # 3.10.5+ Signed-off-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> -- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) --- linux-3.10.5-orig/sound/usb/mixer.c 2013-08-04 12:51:49.000000000 +0400 +++ linux-3.10.5/sound/usb/mixer.c 2013-08-06 21:12:42.274197287 +0400 @@ -888,6 +888,7 @@ static void volume_control_quirks(struct case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. -- WBR, Maksim Boyko

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs4271: Convert to table based control init
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cs4271.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 03036b3..65ad56c 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -576,8 +576,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) CS4271_MODE2_MUTECAEQUB, CS4271_MODE2_MUTECAEQUB); - return snd_soc_add_codec_controls(codec, cs4271_snd_controls, - ARRAY_SIZE(cs4271_snd_controls)); + return 0; } static int cs4271_remove(struct snd_soc_codec *codec) @@ -596,6 +595,9 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, + + .controls = cs4271_snd_controls, + .num_controls = ARRAY_SIZE(cs4271_snd_controls), }; #if defined(CONFIG_SPI_MASTER) -- 1.8.4.rc1

3 5

[alsa-devel] [PATCH] ASoC: max9768: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max9768.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index a6ac231..31f9156 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -118,6 +118,18 @@ static const struct snd_kcontrol_new max9768_mute[] = { SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio), }; +static const struct snd_soc_dapm_widget max9768_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN"), + +SND_SOC_DAPM_OUTPUT("OUT+"), +SND_SOC_DAPM_OUTPUT("OUT-"), +}; + +static const struct snd_soc_dapm_route max9768_dapm_routes[] = { + { "OUT+", NULL, "IN" }, + { "OUT-", NULL, "IN" }, +}; + static int max9768_probe(struct snd_soc_codec *codec) { struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); @@ -148,6 +160,10 @@ static struct snd_soc_codec_driver max9768_codec_driver = { .probe = max9768_probe, .controls = max9768_volume, .num_controls = ARRAY_SIZE(max9768_volume), + .dapm_widgets = max9768_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9768_dapm_widgets), + .dapm_routes = max9768_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9768_dapm_routes), }; static const struct regmap_config max9768_i2c_regmap_config = { -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH] ASoC: ak5386: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak5386.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 1f30398..72e953b 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -22,7 +22,22 @@ struct ak5386_priv { int reset_gpio; }; -static struct snd_soc_codec_driver soc_codec_ak5386; +static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + +static struct snd_soc_codec_driver soc_codec_ak5386 = { + .dapm_widgets = ak5386_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), + .dapm_routes = ak5386_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak5386_dapm_routes), +}; static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH] ASoC: ak5386: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak5386.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 1f30398..fb01f35 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -22,7 +22,22 @@ struct ak5386_priv { int reset_gpio; }; -static struct snd_soc_codec_driver soc_codec_ak5386; +static const struct snd_soc_dapm_widget ak4554_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route ak4554_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + +static struct snd_soc_codec_driver soc_codec_ak5386 = { + .dapm_widgets = ak4554_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets), + .dapm_routes = ak4554_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4554_dapm_routes), +}; static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) -- 1.8.4.rc1

2 2

[alsa-devel] [PATCH] ASoC: cs4270: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cs4270.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8e47798..83c835d 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -139,6 +139,22 @@ struct cs4270_private { struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; +static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, + + { "AOUTA", NULL, "Playback" }, + { "AOUTB", NULL, "Playback" }, +}; + /** * struct cs4270_mode_ratios - clock ratio tables * @ratio: the ratio of MCLK to the sample rate @@ -612,6 +628,10 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .controls = cs4270_snd_controls, .num_controls = ARRAY_SIZE(cs4270_snd_controls), + .dapm_widgets = cs4270_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4270_dapm_widgets), + .dapm_routes = cs4270_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs4270_dapm_routes), }; /* -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH] ASoC: ak4554: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4554.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index 6aed9c4..79e9555 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -29,6 +29,22 @@ * CPU-DAI2 (capture only fmt = LEFT_J) ---+ */ +static const struct snd_soc_dapm_widget ak4554_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route ak4554_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, + + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver ak4554_dai = { .name = "ak4554-hifi", .playback = { @@ -49,6 +65,10 @@ static struct snd_soc_dai_driver ak4554_dai = { }; static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { + .dapm_widgets = ak4554_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets), + .dapm_routes = ak4554_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4554_dapm_routes), }; static int ak4554_soc_probe(struct platform_device *pdev) -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH 1/3] ASoC: bt-sco: enable OF support
by Barry Song 11 Aug '13

11 Aug '13

From: Rongjun Ying <Rongjun.Ying(a)csr.com> this makes users be able to add a bt-sco node in dts instead of hardcoding a platform device in mach. Signed-off-by: Rongjun Ying <Rongjun.Ying(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/bt-sco.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index a081d9f..39e7ebe 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -46,6 +46,14 @@ static int bt_sco_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id bt_sco_of_match[] = { + { .compatible = "bt-sco-codec", }, + {} +}; +MODULE_DEVICE_TABLE(of, bt_sco_of_match); +#endif + static struct platform_device_id bt_sco_driver_ids[] = { { .name = "dfbmcs320", @@ -58,6 +66,9 @@ static struct platform_driver bt_sco_driver = { .driver = { .name = "bt-sco", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = bt_sco_of_match, +#endif }, .probe = bt_sco_probe, .remove = bt_sco_remove, -- 1.8.2.3 Member of the CSR plc group of companies. CSR plc registered in England and Wales, registered number 4187346, registered office Churchill House, Cambridge Business Park, Cowley Road, Cambridge, CB4 0WZ, United Kingdom More information can be found at www.csr.com. Follow CSR on Twitter at http://twitter.com/CSR_PLC and read our blog at www.csr.com/blog

5 15

[alsa-devel] [PATCH 1/2] ASoC: Remove unused snd_soc_info_volsw_ext()
by Lars-Peter Clausen 11 Aug '13

11 Aug '13

The SOC_SINGLE_EXT control has been using snd_soc_info_volsw() for its info callback since commit 1c433fb ("[ALSA] soc - 0.13 ASoC headers"). The snd_soc_info_volsw_ext() function has been unused ever since then, so remove it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 2 -- sound/soc/soc-core.c | 26 -------------------------- 2 files changed, 28 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 13d388a..64eb60a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -499,8 +499,6 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); #define snd_soc_info_bool_ext snd_ctl_boolean_mono_info int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fdddc45..47cb2c0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2594,32 +2594,6 @@ int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); /** - * snd_soc_info_volsw_ext - external single mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a single external mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int max = kcontrol->private_value; - - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); - -/** * snd_soc_info_volsw - single mixer info callback * @kcontrol: mixer control * @uinfo: control element information -- 1.8.0

2 2

[alsa-devel] [PATCH v10 0/7] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 11 Aug '13

11 Aug '13

Hi, this series implements ac97-slave support for fsl-ssi and phycore ac97 DT support. Since v9 there were a lot of changes. I dropped the board specific ac97-link reset functions and added generic functions to soc-core. It uses devicetree parsed pinctrl and gpios. The audmux setup was removed from phycore-ac97-dt. The initial setup can now be defined in DT. Regards, Markus Markus Pargmann (7): ASoC: core: Generic ac97 link reset functions ASoC: codec: wm9712 simple DT bindings ASoC: imx-audmux: Read default configuration from devicetree ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: ac97-slave support ASoC: Add phycore-ac97-dt driver .../devicetree/bindings/sound/fsl,ssi.txt | 12 + .../devicetree/bindings/sound/imx-audmux.txt | 9 + .../bindings/sound/phytec,phycore-ac97.txt | 16 + .../devicetree/bindings/sound/soc-ac97link.txt | 28 ++ Documentation/devicetree/bindings/sound/wm9712.txt | 10 + include/sound/soc.h | 2 + sound/soc/codecs/wm9712.c | 9 + sound/soc/fsl/Kconfig | 15 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 431 +++++++++++++++++---- sound/soc/fsl/imx-audmux.c | 62 +++ sound/soc/fsl/phycore-ac97-dt.c | 124 ++++++ sound/soc/fsl/phycore-ac97.c | 13 +- sound/soc/soc-core.c | 153 ++++++++ 14 files changed, 789 insertions(+), 97 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt create mode 100644 Documentation/devicetree/bindings/sound/wm9712.txt create mode 100644 sound/soc/fsl/phycore-ac97-dt.c

2 19

[alsa-devel] [PATCH] usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525
by Maksim A. Boyko 10 Aug '13

10 Aug '13

Add the volume control quirk for avoiding the kernel warning for the Logitech HD Webcam C525 as in the similar commit 36691e1be6ec551eef4a5225f126a281f8c051c2 for the Logitech HD Webcam C310. Reported-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Tested-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Cc: <stable(a)vger.kernel.org> # 3.10.5+ Signed-off-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> -- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) --- linux-3.10.5-orig/sound/usb/mixer.c 2013-08-04 12:51:49.000000000 +0400 +++ linux-3.10.5/sound/usb/mixer.c 2013-08-06 21:12:42.274197287 +0400 @@ -888,6 +888,7 @@ static void volume_control_quirks(struct case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. -- WBR, Maksim P.S. Please, use this patch due to it should have correct tab symbols.

3 2

[alsa-devel] [PATCH] ALSA: Add MAINTAINERS entry for compressed audio API
by Mark Brown 10 Aug '13

10 Aug '13

From: Mark Brown <broonie(a)linaro.org> Help ensure that Vinod gets included in review of compressed audio patches by adding a MAINTAINERS entry for it. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- MAINTAINERS | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index a367e52..427a930 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7694,6 +7694,15 @@ F: include/sound/ F: include/uapi/sound/ F: sound/ +SOUND - COMPRESSED AUDIO +M: Vinod Koul <vinod.koul(a)intel.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git +S: Supported +F: include/sound/compress_driver.h +F: sound/core/compress_offload.c +F: sound/soc/soc-compress.c + SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)kernel.org> -- 1.8.4.rc1

2 1

[alsa-devel] [PATCH] ASoC: Remove unused soc_pm_waitq
by Lars-Peter Clausen 10 Aug '13

10 Aug '13

The soc_pm_waitq waitqueue has been around as long as the ASoC framework existed, but has never been used so far, so remove it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1d5e84a..fdddc45 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -47,8 +47,6 @@ #define NAME_SIZE 32 -static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); - #ifdef CONFIG_DEBUG_FS struct dentry *snd_soc_debugfs_root; EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); -- 1.8.0

2 1

[alsa-devel] [PATCH v4 1/1] ASoc: kirkwood: add DT support to the mvebu audio subsystem
by Jean-Francois Moine 10 Aug '13

10 Aug '13

This patch adds DT support to the audio subsystem of the mvebu family (Kirkwood, Dove, Armada 370). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- .../devicetree/bindings/sound/mvebu-audio.txt | 29 ++++++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 26 +++++++++++++++++----- 2 files changed, 49 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt new file mode 100644 index 0000000..7e5fd37 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) audio controller + +Required properties: + +- compatible: "mrvl,mvebu-audio" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "extclk" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "mrvl,mvebu-audio"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c.next b/sound/soc/kirkwood/kirkwood-i2s.c index e5f3f7a..a4170b4 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c.next +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -22,6 +22,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" #define DRV_NAME "mvebu-audio" @@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np = pdev->dev.of_node; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); return -EINVAL; } - priv->burst = data->burst; - - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); if (IS_ERR(priv->clk)) { dev_err(&pdev->dev, "no clock\n"); return PTR_ERR(priv->clk); @@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static struct of_device_id kirkwood_i2s_of_match[] = { + { .compatible = "mrvl,mvebu-audio" }, + { } +}; +MODULE_DEVICE_TABLE(of, kirkwood_i2s_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(kirkwood_i2s_of_match), }, }; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

6 20

[alsa-devel] [patch] sound: oss/dmabuf: remove an unneeded temporary variable
by Dan Carpenter 09 Aug '13

09 Aug '13

We don't actually use the "go" variable so it can be removed. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index a59c888..461d94c 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -557,7 +557,6 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock) unsigned long flags; int err = 0, n = 0; struct dma_buffparms *dmap = adev->dmap_in; - int go; if (!(adev->open_mode & OPEN_READ)) return -EIO; @@ -584,7 +583,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock) spin_unlock_irqrestore(&dmap->lock,flags); return -EAGAIN; } - if ((go = adev->go)) + if (adev->go) timeout = dmabuf_timeout(dmap); spin_unlock_irqrestore(&dmap->lock,flags);

2 2

[alsa-devel] [PATCH V3 0/3] Add audio support on smdk5420
by Padmavathi Venna 09 Aug '13

09 Aug '13

Samsung has different versions of I2S introduced in different platforms. Each version has some new support added for multichannel, secondary fifo, s/w reset control, internal mux for rclk src clk and tdm support. Each newly added change has a quirk. So this patch adds all the required quirks as driver data and based on compatible string from dtsi fetches the quirks. This also adds i2s support on exynos5420. Changes since V2: - Seperated out driver side changes and dts changes in two patch sets. - Replaced samsung,s3c6410-i2s-v4 with samsung,s3c6410-i2s-multi for more clarity as suggested by Tomasz Figa. Changes since V1: - Pass quirks as driver data and fetch the quirks based on compatible string from dtsi file as suggested by Tomasz Figa and Mark Brown - Make the I2S driver more flexible with respect to register access as suggested by Tomasz Figa and Mark Brown - Add 5420 support in the driver. - Modify the dtsi files with the corresponding compatible strings and removed the i2s quirks from 5250 dtsi file. - Updated the i2s Documentation with relevent changes and i2s versioning info. - Add i2s nodes on exynos5420.dtsi - Enable sound support on smdk5420 This patch set is made based on Mark Brown for-next branch on sound.git. Padmavathi Venna (3): platform: Increase platform name size ASoC: Samsung: I2S: Add quirks as driver data in I2S ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 .../devicetree/bindings/sound/samsung-i2s.txt | 25 ++-- include/linux/mod_devicetable.h | 2 +- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 15 ++ sound/soc/samsung/i2s.c | 163 +++++++++++++++----- 5 files changed, 158 insertions(+), 48 deletions(-) -- 1.7.4.4

3 5

[alsa-devel] Found a typo
by Torino 09 Aug '13

09 Aug '13

Dear Alsa developers, within Saucy (13.10), "alsa-utils", string nr. 124 it says "Channel %d doesn't match with hw_parmas" whereas in other strings like nr. 84 or 126 it says "hw_params". I hope it was correct informing you via this mailing list. Best Regads, Torino (from the German Übersetzer-Team)

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: do not trust too-big wMaxPacketSize values
by Clemens Ladisch 08 Aug '13

08 Aug '13

The driver used to assume that the streaming endpoint's wMaxPacketSize value would be an indication of how much data the endpoint expects or sends, and compute the number of packets per URB using this value. However, the Focusrite Scarlett 2i4 declares a value of 1024 bytes, while only about 88 or 44 bytes are be actually used. This discrepancy would result in URBs with far too few packets, which would not work correctly on the EHCI driver. To get correct URBs, use wMaxPacketSize only as an upper limit on the packet size. Reported-by: James Stone <jamesmstone(a)gmail.com> Tested-by: James Stone <jamesmstone(a)gmail.com> Cc: <stable(a)vger.kernel.org> # 2.6.35+ Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/endpoint.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -591,17 +591,16 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; - /* calculate max. frequency */ - if (ep->maxpacksize) { + /* assume max. frequency is 25% higher than nominal */ + ep->freqmax = ep->freqn + (ep->freqn >> 2); + maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - ep->datainterval); + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ maxsize = ep->maxpacksize; ep->freqmax = (maxsize / (frame_bits >> 3)) << (16 - ep->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); } if (ep->fill_max)

3 2

[alsa-devel] [PATCH 1/1] ASoC: s6105-ipcam: Fix incorrect placement of __initdata
by Sachin Kamat 08 Aug '13

08 Aug '13

__initdata should be placed between the variable name and equal sign for the variable to be placed in the intended section. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Daniel Gloeckner <dg(a)emlix.com> --- sound/soc/s6000/s6105-ipcam.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 58cfb1e..945e8ab 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -192,7 +192,7 @@ static struct snd_soc_card snd_soc_card_s6105 = { .num_links = 1, }; -static struct s6000_snd_platform_data __initdata s6105_snd_data = { +static struct s6000_snd_platform_data s6105_snd_data __initdata = { .wide = 0, .channel_in = 0, .channel_out = 1, -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: ad73311: Add DAPM support
by Mark Brown 08 Aug '13

08 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook up other devices in boards and is required by removal of support for non-DAPM CODECs in the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ad73311.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index b1f2baf..5fac8ad 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -23,6 +23,21 @@ #include "ad73311.h" +static const struct snd_soc_dapm_widget ad73311_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINP"), +SND_SOC_DAPM_INPUT("VINN"), +SND_SOC_DAPM_OUTPUT("VOUTN"), +SND_SOC_DAPM_OUTPUT("VOUTP"), +}; + +static const struct snd_soc_dapm_route ad73311_dapm_routes[] = { + { "Capture", NULL, "VINP" }, + { "Capture", NULL, "VINN" }, + + { "VOUTN", NULL, "Playback" }, + { "VOUTP", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver ad73311_dai = { .name = "ad73311-hifi", .playback = { @@ -39,7 +54,12 @@ static struct snd_soc_dai_driver ad73311_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; -static struct snd_soc_codec_driver soc_codec_dev_ad73311; +static struct snd_soc_codec_driver soc_codec_dev_ad73311 = { + .dapm_widgets = ad73311_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad73311_dapm_widgets), + .dapm_routes = ad73311_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad73311_dapm_routes), +}; static int ad73311_probe(struct platform_device *pdev) { -- 1.8.4.rc1

2 3

[alsa-devel] [PATCH] ASoC: pcm1681: Staticise DAI driver
by Mark Brown 08 Aug '13

08 Aug '13

From: Mark Brown <broonie(a)linaro.org> It is not exported so doesn't need to be in the global namespace and sparse warns on this. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 27da41b..51b1866 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -225,7 +225,7 @@ static const struct snd_kcontrol_new pcm1681_controls[] = { pcm1681_get_deemph, pcm1681_put_deemph), }; -struct snd_soc_dai_driver pcm1681_dai = { +static struct snd_soc_dai_driver pcm1681_dai = { .name = "pcm1681-hifi", .playback = { .stream_name = "Playback", -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH v4 0/1] ASoc: kirkwood: extend the kirkwood audio subsystem for DT usage
by Jean-Francois Moine 08 Aug '13

08 Aug '13

This patch series adds DT support to the kirkwood/mvebu audio subsystem. History v4 - change the DT name to "mrvl,mvebu-audio" - remove previous patches 1..3 which have been applied by Russell King v3 - change the DT name from "kirkwood-i2s" to "mvebu-pcm-audio" - apply to linux-next - let NULL as the internal i2s clock for compatibility - change i2s external clock name to "extclk" instead of "external" - cancel previous patch 1/4. Russell's patch should be applied instead. v2 - fix compilation errors -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] bug
by Mattia Bertoni 08 Aug '13

08 Aug '13

finally I can report a bug which I encountered some time ago; I send you the important part of the entire log, which is to much big to be posted: Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 420736 bytes (2191 ms). Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: snd_pcm_dump(): Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Hardware PCM card 0 'NVidia CK804' device 0 subdevice 0 Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Its setup is: Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: stream : PLAYBACK Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: access : MMAP_INTERLEAVED Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: format : S16_LE Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: subformat : STD Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: channels : 2 Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: rate : 48000 Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: exact rate : 48000 (48000/1) Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: msbits : 16 Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: buffer_size : 16384 Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_size : 16384 Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_time : 341333 Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: tstamp_mode : ENABLE Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_step : 1 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: avail_min : 16384 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_event : 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: start_threshold : -1 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: stop_threshold : 4611686018427387904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: silence_threshold: 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: silence_size : 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: boundary : 4611686018427387904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: appl_ptr : 26406104 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: hw_ptr : 26494904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -465056 bytes (-2422 ms). Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: snd_pcm_dump(): Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Hardware PCM card 0 'NVidia CK804' device 0 subdevice 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Its setup is: Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: stream : PLAYBACK Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: access : MMAP_INTERLEAVED Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: format : S16_LE Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: subformat : STD Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: channels : 2 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: rate : 48000 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: exact rate : 48000 (48000/1) Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: msbits : 16 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: buffer_size : 16384 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_size : 16384 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_time : 341333 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: tstamp_mode : ENABLE Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_step : 1 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: avail_min : 16384 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_event : 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: start_threshold : -1 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: stop_threshold : 4611686018427387904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: silence_threshold: 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: silence_size : 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: boundary : 4611686018427387904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: appl_ptr : 26569944 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: hw_ptr : 26686208

2 6

[alsa-devel] [PATCH] [alsa-lib] UCM: Document some standard values
by Arun Raghavan 08 Aug '13

08 Aug '13

Playback/CaptureChannels has been around for a bit. Playback/CaptureRate is new and is intended to be used to specify the sample rate at which the Playback/CapturePCM should be opened. --- include/use-case.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/use-case.h b/include/use-case.h index 93b3456..4e13249 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -259,9 +259,13 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * TQ - Tone Quality * PlaybackPCM - full PCM playback device name * CapturePCM - full PCM capture device name + * PlaybackRate - playback device sample rate + * PlaybackChannels - playback device channel count * PlaybackCTL - playback control device name * PlaybackVolume - playback control volume ID string * PlaybackSwitch - playback control switch ID string + * CaptureRate - capture device sample rate + * CaptureChannels - capture device channel count * CaptureCTL - capture control device name * CaptureVolume - capture control volume ID string * CaptureSwitch - capture control switch ID string -- 1.8.2.1

2 1

Re: [alsa-devel] [PATCH V3 2/3] ASoC: Samsung: I2S: Add quirks as driver data in I2S
by Padma Venkat 08 Aug '13

08 Aug '13

Hi Tomasz, On Wed, Aug 7, 2013 at 4:43 PM, Tomasz Figa <t.figa(a)samsung.com> wrote: > Ahh, one more thing inline. > [snip] >> >> +static struct samsung_i2s_dai_data i2sv3_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv4_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_c100_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR | QUIRK_SEC_DAI | >> + QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data samsung_dai_type_sec = { >> + .dai_type = TYPE_SEC, >> +}; >> + >> static struct platform_device_id samsung_i2s_driver_ids[] = { >> { >> - .name = "samsung-i2s", >> - .driver_data = TYPE_PRI, >> + .name = "samsung,s3c6410-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv3_dai_type, >> + }, { >> + .name = "samsung,s3c6410-i2s-multi", >> + .driver_data = (kernel_ulong_t)&i2sv4_dai_type, >> + }, { >> + .name = "samsung,s5pc100-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_c100_dai_type, >> }, { >> - .name = "samsung-i2s-sec", >> - .driver_data = TYPE_SEC, >> + .name = "samsung,s5pv210-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_dai_type, >> + }, { >> + .name = "samsung-i2s-sec", >> + .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, > > I don't think you need to change the legacy platform IDs at all. If legacy platforms not required to change then I need to introduce a new samsung_i2s_dai_data structure which holds only dai_type for non-dt platforms. If I change legacy platforms it breaks the older platforms now. Is it okay adding a samsung_i2s_dai_data structure for non-dt platforms which will be removed later? > > IMHO it would be better to keep the old way of quirk retrieval from > platform_data (which I think this patch does anyway, because the probe path > without DT is unchanged), as it will be dropped in some point in time > anyway. > > This would also eliminate the need for patch 1. > > Best regards, > Tomasz > Thanks Padma

3 2

[alsa-devel] snd_pcm_set_params typo patch
by Laurentiu Nicola 08 Aug '13

08 Aug '13

Please, somebody apply this one. I lost quite a bit of time because of it :). Also, is there anything wrong with the bug tracker? It doesn't load for me. diff -urpN alsa-lib-1.0.27.2/src/pcm/pcm.c alsa-lib-1.0.27.2-mine/src/pcm/pcm.c --- alsa-lib-1.0.27.2/src/pcm/pcm.c 2013-07-08 15:31:36.000000000 +0300 +++ alsa-lib-1.0.27.2-mine/src/pcm/pcm.c 2013-08-07 22:17:14.908673700 +0300 @@ -7932,7 +7932,7 @@ int snd_pcm_set_params(snd_pcm_t *pcm, return err; } if (rrate != rate) { - SNDERR("Rate doesn't match (requested %iHz, get %iHz)", rate, err); + SNDERR("Rate doesn't match (requested %iHz, got %iHz)", rate, rrate); return -EINVAL; } /* set the buffer time */ -- Because there are many ways to skin a cat, but you only get to choose one.

2 1

[alsa-devel] [PATCH] ALSA: don't push static constants on stack for %*ph
by Andy Shevchenko 08 Aug '13

08 Aug '13

There is no need to pass constants via stack. The width may be explicitly specified in the format. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/isa/gus/interwave.c | 3 +-- sound/usb/6fire/firmware.c | 4 ++-- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 9942691..afef0d7 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) for (i = 0; i < 8; ++i) iwave[i] = snd_gf1_peek(gus, bank_pos + i); #ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos, - 8, iwave); + printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave); #endif if (strncmp(iwave, "INTRWAVE", 8)) continue; /* first check */ diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index b9defcd..780bf3f 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -346,10 +346,10 @@ static int usb6fire_fw_check(u8 *version) if (!memcmp(version, known_fw_versions + i, 2)) return 0; - snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. " + snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. " "please reconnect to power. if this failure " "still happens, check your firmware installation.", - 4, version); + version); return -EINVAL; } -- 1.8.4.rc1

2 1

[alsa-devel] Alsa bug/bugtracker
by Mattia Bertoni 08 Aug '13

08 Aug '13

hi, under Mark Brown directions I submit to you my problem: recently I've gotten a crash on my machine running Archlinux x86_64 the log suggest me to report it in the Alsa bugtracker but the link on the official project (https://bugtrack.alsa-project.org/alsa-bug/) is down.... how can I do ??? thank you

3 4

[alsa-devel] [PATCH] Fix invalid volume resolution for Logitech HD Webcam C525
by Бойко Максим 08 Aug '13

08 Aug '13

From: Maksim Boyko <maksim.a.boyko(a)gmail.com> Add the volume control quirk for avoiding the kernel warning for Logitech HD Webcam C525. The similar patch was previously reported for Logitech HD Webcam C310 (see 36691e1be6ec551eef4a5225f126a281f8c051c2). --- linux-3.10.5-orig/sound/usb/mixer.c 2013-08-04 12:51:49.000000000 +0400 +++ linux-3.10.5/sound/usb/mixer.c 2013-08-06 21:12:42.274197287 +0400 @@ -888,6 +888,7 @@ case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384.

3 2

[alsa-devel] [PATCH] ASoC: dt: Move WM8903 pin list from Tegra board binding to CODEC binding
by Mark Brown 08 Aug '13

08 Aug '13

From: Mark Brown <broonie(a)linaro.org> The pin list is the same for any board using the CODEC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 24 ++-------------------- Documentation/devicetree/bindings/sound/wm8903.txt | 19 +++++++++++++++++ 2 files changed, 21 insertions(+), 22 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt index 3bf722d..4b44dfb 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt @@ -11,28 +11,8 @@ Required properties: - nvidia,audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources and - sinks are the WM8903's pins, and the jacks on the board: - - WM8903 pins: - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS - - Board connectors: + sinks are the WM8903's pins (documented in the WM8903 binding document), + and the jacks on the board: * Headphone Jack * Int Spk diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt index f102cbc..94ec32c 100644 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ b/Documentation/devicetree/bindings/sound/wm8903.txt @@ -28,6 +28,25 @@ Optional properties: performed. If any entry has the value 0xffffffff, that GPIO's configuration will not be modified. +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + Example: codec: wm8903@1a { -- 1.8.4.rc1

3 2

[alsa-devel] [PATCH] ALSA: hda - Fix missing mute controls for CX5051
by Takashi Iwai 08 Aug '13

08 Aug '13

We've added a fake mute control (setting the amp volume to zero) for CX5051 at commit [3868137e: ALSA: hda - Add a fake mute feature], but this feature was overlooked in the generic parser implementation. Now the driver lacks of mute controls on these codecs. The fix is just to check both AC_AMPCAP_MUTE and AC_AMPCAP_MIN_MUTE bits in each place checking the amp capabilities. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59001 Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbb..e3c7ba8 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -522,7 +522,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, } #define nid_has_mute(codec, nid, dir) \ - check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE) + check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) #define nid_has_volume(codec, nid, dir) \ check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS) @@ -624,7 +624,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, if (enable) val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; } - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (!enable) val |= HDA_AMP_MUTE; } @@ -648,7 +648,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec, { unsigned int mask = 0xff; - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL)) mask &= ~0x80; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: ak4104: Add stub DAPM support
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it easer to integrate the device with other on-board components and ensures correct operation following removal of support for non-DAPM CODECs. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4104.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index c7cfdf9..9a7c89b 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -51,6 +51,14 @@ struct ak4104_private { struct regmap *regmap; }; +static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ak4104_dapm_routes[] = { + { "TX", NULL, "Playback" }, +}; + static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { @@ -214,6 +222,11 @@ static int ak4104_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, + + .dapm_widgets = ak4104_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4104_dapm_widgets), + .dapm_routes = ak4104_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4104_dapm_routes), }; static const struct regmap_config ak4104_regmap = { -- 1.8.4.rc1

1 1

[alsa-devel] [PATCH] ASoC: dt: Move WM8903 pin list from Tegra board binding to CODEC binding
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> The pin list is the same for any board using the CODEC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 22 ++-------------------- Documentation/devicetree/bindings/sound/wm8903.txt | 19 +++++++++++++++++++ 2 files changed, 21 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt index 3bf722d..0efe58c 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt @@ -11,26 +11,8 @@ Required properties: - nvidia,audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources and - sinks are the WM8903's pins, and the jacks on the board: - - WM8903 pins: - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS + sinks are the WM8903's pins (documented in the WM8903 binding document), + and the jacks on the board: Board connectors: diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt index f102cbc..94ec32c 100644 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ b/Documentation/devicetree/bindings/sound/wm8903.txt @@ -28,6 +28,25 @@ Optional properties: performed. If any entry has the value 0xffffffff, that GPIO's configuration will not be modified. +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + Example: codec: wm8903@1a { -- 1.8.4.rc1

2 4

[alsa-devel] [PATCH] ASoC: ads711x: Add DAPM support
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it easier to hook into boards and ensures the driver continues to work with support for non-DAPM CODECs removed. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ads117x.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 506d474..8f388ed 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -23,6 +23,28 @@ #define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000) #define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) +static const struct snd_soc_dapm_widget ads117x_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("Input1"), +SND_SOC_DAPM_INPUT("Input2"), +SND_SOC_DAPM_INPUT("Input3"), +SND_SOC_DAPM_INPUT("Input4"), +SND_SOC_DAPM_INPUT("Input5"), +SND_SOC_DAPM_INPUT("Input6"), +SND_SOC_DAPM_INPUT("Input7"), +SND_SOC_DAPM_INPUT("Input8"), +}; + +static const struct snd_soc_dapm_route ads117x_dapm_routes[] = { + { "Capture", NULL, "Input1" }, + { "Capture", NULL, "Input2" }, + { "Capture", NULL, "Input3" }, + { "Capture", NULL, "Input4" }, + { "Capture", NULL, "Input5" }, + { "Capture", NULL, "Input6" }, + { "Capture", NULL, "Input7" }, + { "Capture", NULL, "Input8" }, +}; + static struct snd_soc_dai_driver ads117x_dai = { /* ADC */ .name = "ads117x-hifi", @@ -34,7 +56,12 @@ static struct snd_soc_dai_driver ads117x_dai = { .formats = ADS117X_FORMATS,}, }; -static struct snd_soc_codec_driver soc_codec_dev_ads117x; +static struct snd_soc_codec_driver soc_codec_dev_ads117x = { + .dapm_widgets = ads117x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets), + .dapm_routes = ads117x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ads117x_dapm_routes), +}; static int ads117x_probe(struct platform_device *pdev) { -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH] 6fire: fix DMA issues with URB transfer_buffer usage
by Jussi Kivilinna 07 Aug '13

07 Aug '13

Patch fixes 6fire not to use stack as URB transfer_buffer. URB buffers need to be DMA-able, which stack is not. Furthermore, transfer_buffer should not be allocated as part of larger device structure because DMA coherency issues and patch fixes this issue too. Patch is only compile tested. Cc: stable(a)vger.kernel.org Signed-off-by: Jussi Kivilinna <jussi.kivilinna(a)iki.fi> --- sound/usb/6fire/comm.c | 38 +++++++++++++++++++++++++++++++++----- sound/usb/6fire/comm.h | 2 +- 2 files changed, 34 insertions(+), 6 deletions(-) diff --git a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c index 9e6e3ff..23452ee 100644 --- a/sound/usb/6fire/comm.c +++ b/sound/usb/6fire/comm.c @@ -110,19 +110,37 @@ static int usb6fire_comm_send_buffer(u8 *buffer, struct usb_device *dev) static int usb6fire_comm_write8(struct comm_runtime *rt, u8 request, u8 reg, u8 value) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, value, 0x00); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } static int usb6fire_comm_write16(struct comm_runtime *rt, u8 request, u8 reg, u8 vl, u8 vh) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, vl, vh); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } int usb6fire_comm_init(struct sfire_chip *chip) @@ -135,6 +153,12 @@ int usb6fire_comm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->receiver_buffer = kzalloc(COMM_RECEIVER_BUFSIZE, GFP_KERNEL); + if (!rt->receiver_buffer) { + kfree(rt); + return -ENOMEM; + } + urb = &rt->receiver; rt->serial = 1; rt->chip = chip; @@ -153,6 +177,7 @@ int usb6fire_comm_init(struct sfire_chip *chip) urb->interval = 1; ret = usb_submit_urb(urb, GFP_KERNEL); if (ret < 0) { + kfree(rt->receiver_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create comm data receiver."); return ret; @@ -171,6 +196,9 @@ void usb6fire_comm_abort(struct sfire_chip *chip) void usb6fire_comm_destroy(struct sfire_chip *chip) { - kfree(chip->comm); + struct comm_runtime *rt = chip->comm; + + kfree(rt->receiver_buffer); + kfree(rt); chip->comm = NULL; } diff --git a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h index 6a0840b..780d5ed 100644 --- a/sound/usb/6fire/comm.h +++ b/sound/usb/6fire/comm.h @@ -24,7 +24,7 @@ struct comm_runtime { struct sfire_chip *chip; struct urb receiver; - u8 receiver_buffer[COMM_RECEIVER_BUFSIZE]; + u8 *receiver_buffer; u8 serial; /* urb serial */

3 3

Re: [alsa-devel] [PATCH V3 2/3] ASoC: Samsung: I2S: Add quirks as driver data in I2S
by Padma Venkat 07 Aug '13

07 Aug '13

Hi Tomasz, On Wed, Aug 7, 2013 at 4:32 PM, Tomasz Figa <t.figa(a)samsung.com> wrote: > Hi Padmavathi, > > [Ccing DT maintainers with a little comment about contents of this patch > for them: > > This is a rework of Samsung i2s bindings to make them more reasonable than > they are at the moment. I know this was supposed to be stable, fixed, ABI, > etc., but as we discussed a lot the whole topic of DT bindings, we need to > sanitize existing bindings and this patch is a part of this work.] > [snip] >> >> Required SoC Specific Properties: >> >> -- compatible : "samsung,i2s-v5" >> +- compatible : should be one of the following. >> + - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. Previous versions >> + has only 8/16bit support. > > The comment about previous versions seems a bit enigmatic. > > If there is another variant supported by this driver that supports only > 8/16bit audio data, then it should have a separate compatible value. For previous i2s controllers there are two other i2s drivers available. I think I can delete this comment here. > >> + - samsung,s3c6410-i2sv4: for 8/16/24bit multichannel(5.1 channel) >> I2S. + Introduced in s3c6410. This also applicable for s5p64x0 >> platforms. + - samsung,s5pc100-i2s: for 8/16/24bit multichannel(5.1 >> channel) I2S + with secondary fifo and s/w reset control. >> + - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with >> + secondary fifo, s/w reset control and internal mux for root clk >> src. + >> - reg: physical base address of the controller and length of memory >> mapped region. >> - dmas: list of DMA controller phandle and DMA request line ordered >> pairs. @@ -21,13 +30,6 @@ Required SoC Specific Properties: >> >> Optional SoC Specific Properties: >> >> -- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel >> - support, this flag is enabled. >> -- samsung,supports-rstclr: This flag should be set if I2S software reset >> bit - control is required. When this flag is set I2S software reset bit >> will be - enabled or disabled based on need. >> -- samsung,supports-secdai:If I2S block has a secondary FIFO and internal >> DMA, - then this flag is enabled. >> - samsung,idma-addr: Internal DMA register base address of the audio >> sub system(used in secondary sound source). >> - pinctrl-0: Should specify pin control groups used for this controller. >> @@ -46,9 +48,6 @@ i2s0: i2s@03830000 { > > The example has also a compatible value set to "samsung,i2s-v5". I don't > think this is compliant to the new bindings. Yes. It's my mistake. Will change this to new one. > >> <&clock_audss EXYNOS_I2S_BUS>, >> <&clock_audss EXYNOS_SCLK_I2S>; >> clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; >> - samsung,supports-6ch; >> - samsung,supports-rstclr; >> - samsung,supports-secdai; >> samsung,idma-addr = <0x03000000>; >> pinctrl-names = "default"; >> pinctrl-0 = <&i2s0_bus>; >> diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c >> index 47e08dd..8a5504c 100644 >> --- a/sound/soc/samsung/i2s.c >> +++ b/sound/soc/samsung/i2s.c >> @@ -40,6 +40,7 @@ enum samsung_dai_type { >> >> struct samsung_i2s_dai_data { > > const struct samsung_i2s_dai_data OK. > >> int dai_type; >> + u32 quirks; >> }; >> >> struct i2s_dai { >> @@ -1032,18 +1033,18 @@ static struct i2s_dai *i2s_alloc_dai(struct >> platform_device *pdev, bool sec) >> >> static const struct of_device_id exynos_i2s_match[]; >> >> -static inline int samsung_i2s_get_driver_data(struct platform_device >> *pdev) +static inline struct samsung_i2s_dai_data >> *samsung_i2s_get_driver_data( + struct platform_device > *pdev) >> { >> #ifdef CONFIG_OF >> - struct samsung_i2s_dai_data *data; >> if (pdev->dev.of_node) { >> const struct of_device_id *match; >> match = of_match_node(exynos_i2s_match, pdev->dev.of_node); >> - data = (struct samsung_i2s_dai_data *) match->data; >> - return data->dai_type; >> + return (struct samsung_i2s_dai_data *) match->data; > > If you make the dai_data const this cast will be no longer necessary. > >> } else >> #endif >> - return platform_get_device_id(pdev)->driver_data; >> + return (struct samsung_i2s_dai_data *) >> + platform_get_device_id(pdev)->driver_data; >> } >> >> #ifdef CONFIG_PM_RUNTIME >> @@ -1074,13 +1075,13 @@ static int samsung_i2s_probe(struct >> platform_device *pdev) struct resource *res; >> u32 regs_base, quirks = 0, idma_addr = 0; >> struct device_node *np = pdev->dev.of_node; >> - enum samsung_dai_type samsung_dai_type; >> + struct samsung_i2s_dai_data *i2s_dai_data; > > const OK. > >> int ret = 0; >> >> /* Call during Seconday interface registration */ >> - samsung_dai_type = samsung_i2s_get_driver_data(pdev); >> + i2s_dai_data = samsung_i2s_get_driver_data(pdev); >> >> - if (samsung_dai_type == TYPE_SEC) { >> + if (i2s_dai_data->dai_type == TYPE_SEC) { >> sec_dai = dev_get_drvdata(&pdev->dev); >> if (!sec_dai) { >> dev_err(&pdev->dev, "Unable to get drvdata\n"); >> @@ -1129,15 +1130,7 @@ static int samsung_i2s_probe(struct >> platform_device *pdev) idma_addr = i2s_cfg->idma_addr; >> } >> } else { >> - if (of_find_property(np, "samsung,supports-6ch", NULL)) >> - quirks |= QUIRK_PRI_6CHAN; >> - >> - if (of_find_property(np, "samsung,supports-secdai", NULL)) >> - quirks |= QUIRK_SEC_DAI; >> - >> - if (of_find_property(np, "samsung,supports-rstclr", NULL)) >> - quirks |= QUIRK_NEED_RSTCLR; >> - >> + quirks = i2s_dai_data->quirks; >> if (of_property_read_u32(np, "samsung,idma-addr", >> &idma_addr)) { >> if (quirks & QUIRK_SEC_DAI) { >> @@ -1250,27 +1243,60 @@ static int samsung_i2s_remove(struct >> platform_device *pdev) return 0; >> } >> >> +static struct samsung_i2s_dai_data i2sv3_dai_type = { > > const OK. > >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv4_dai_type = { > > ditto > >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_c100_dai_type = { > > ditto > >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR | QUIRK_SEC_DAI | >> + QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_dai_type = { > > ditto > >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data samsung_dai_type_sec = { > > ditto > >> + .dai_type = TYPE_SEC, >> +}; >> + >> static struct platform_device_id samsung_i2s_driver_ids[] = { >> { >> - .name = "samsung-i2s", >> - .driver_data = TYPE_PRI, >> + .name = "samsung,s3c6410-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv3_dai_type, >> + }, { >> + .name = "samsung,s3c6410-i2s-multi", >> + .driver_data = (kernel_ulong_t)&i2sv4_dai_type, >> + }, { >> + .name = "samsung,s5pc100-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_c100_dai_type, >> }, { >> - .name = "samsung-i2s-sec", >> - .driver_data = TYPE_SEC, >> + .name = "samsung,s5pv210-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_dai_type, >> + }, { >> + .name = "samsung-i2s-sec", >> + .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, >> }, >> {}, >> }; >> MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids); >> >> #ifdef CONFIG_OF >> -static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = { >> - [TYPE_PRI] = { TYPE_PRI }, >> - [TYPE_SEC] = { TYPE_SEC }, >> -}; >> - >> static const struct of_device_id exynos_i2s_match[] = { >> - { .compatible = "samsung,i2s-v5", >> - .data = &samsung_i2s_dai_data_array[TYPE_PRI], >> + { >> + .compatible = "samsung,s3c6410-i2s", >> + .data = &i2sv3_dai_type, >> + }, { >> + .compatible = "samsung,s5pv210-i2s", >> + .data = &i2sv5_dai_type, > > Hmm, documentation mentions 4 compatible values (and possibly one is > missing there), but this patch adds only 2 of them. I don't think this is > correct, especially that the driver already supports handling all of them. Documentation mentions about all the compatible strings which are all supported by this driver for reference. But some of the platforms haven't converted to DT yet so here I added the compatible strings for DT compliant platforms and compatible strings for non-DT compliant platforms I added in the platform_device_id structure. Any way we have to make all legacy platforms DT compliant so we need this "samsung,s3c6410-i2s-multi" in future. > > Best regards, > Tomasz > Thanks Padma

1 0

[alsa-devel] [PATCH] switch: arizona: Clear trig_sts bits on all paths
by Charles Keepax 07 Aug '13

07 Aug '13

We want the trig_sts bits to be cleared in all cases where we consider the jack detection interrupt to have been handled. Specifically, if a duplicate detection event was suppressed these bits were not cleared causing the CODEC to not enter a low power state. This patch clears the bits on the duplicate detection code path. Reported-by: Ryo Tsutsui <ryo.tsutsui(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/extcon/extcon-arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/extcon/extcon-arizona.c b/drivers/extcon/extcon-arizona.c index 7a1b4a7..4df68de 100644 --- a/drivers/extcon/extcon-arizona.c +++ b/drivers/extcon/extcon-arizona.c @@ -1034,6 +1034,7 @@ static irqreturn_t arizona_jackdet(int irq, void *data) else info->micd_timeout = DEFAULT_MICD_TIMEOUT; +out: /* Clear trig_sts to make sure DCVDD is not forced up */ regmap_write(arizona->regmap, ARIZONA_AOD_WKUP_AND_TRIG, ARIZONA_MICD_CLAMP_FALL_TRIG_STS | @@ -1041,7 +1042,6 @@ static irqreturn_t arizona_jackdet(int irq, void *data) ARIZONA_JD1_FALL_TRIG_STS | ARIZONA_JD1_RISE_TRIG_STS); -out: mutex_unlock(&info->lock); pm_runtime_mark_last_busy(info->dev); -- 1.7.2.5

2 2

[alsa-devel] Linux 3.10.4 and Logitech HD Webcam C525/C310
by Бойко Максим 07 Aug '13

07 Aug '13

Hi, I have Logitech HD Webcam 525 (046d:0826) but its mic works strange. The mic is turned off while video works fine. In dmesg I see the following messages [ 472.079738] cannot submit urb (err = -18) [ 472.079748] cannot submit urb (err = -18) [ 472.079754] cannot submit urb (err = -18) [ 472.079759] cannot submit urb (err = -18) [ 472.079783] cannot submit urb (err = -18) [ 472.079787] cannot submit urb (err = -18) [ 472.079792] cannot submit urb (err = -18) [ 472.079797] cannot submit urb (err = -18) I have did the following modifications to try to fix it (it works for some other cams) add to the sound/usb/mixer.c case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ add to the sound/usb/quirks-table.h (see http://forums.gentoo.org/viewtopic-p-6886404.html) { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .idVendor = 0x046d, .idProduct = 0x0826, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Logitech, Inc.", .product_name = "HD Webcam C525", .ifnum = QUIRK_NO_INTERFACE } }, But all these don't fix the mic. I have not enough experience to fix it myself but I can help with finding a bug and testing fixes. I have also found that the same problem exists for Logitech Webcam C310 (046d:081b). The bug is not specific for my laptop. I have the same issue on my desktop with both cameras. See attachments for additional info.

2 3

[alsa-devel] [PATCH 1/4] mfd: wm5110: Mark register containing FLL gains as readable
by Charles Keepax 07 Aug '13

07 Aug '13

The FLL gains are present on all devices and controlled from common code shared between them. This patch adds these registers into defaults list for wm5110 and marks them as readable. Reported-by: D.J. Barrow <dbarrow(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 2a79723..c04324d 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -468,12 +468,14 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000176, 0x0000 }, /* R374 - FLL1 Control 6 */ { 0x00000177, 0x0281 }, /* R375 - FLL1 Loop Filter Test 1 */ { 0x00000178, 0x0000 }, /* R376 - FLL1 NCO Test 0 */ + { 0x00000179, 0x0000 }, /* R376 - FLL1 Control 7 */ { 0x00000181, 0x0000 }, /* R385 - FLL1 Synchroniser 1 */ { 0x00000182, 0x0000 }, /* R386 - FLL1 Synchroniser 2 */ { 0x00000183, 0x0000 }, /* R387 - FLL1 Synchroniser 3 */ { 0x00000184, 0x0000 }, /* R388 - FLL1 Synchroniser 4 */ { 0x00000185, 0x0000 }, /* R389 - FLL1 Synchroniser 5 */ { 0x00000186, 0x0000 }, /* R390 - FLL1 Synchroniser 6 */ + { 0x00000187, 0x0001 }, /* R390 - FLL1 Synchroniser 7 */ { 0x00000189, 0x0000 }, /* R393 - FLL1 Spread Spectrum */ { 0x0000018A, 0x0004 }, /* R394 - FLL1 GPIO Clock */ { 0x00000191, 0x0000 }, /* R401 - FLL2 Control 1 */ @@ -484,12 +486,14 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000196, 0x0000 }, /* R406 - FLL2 Control 6 */ { 0x00000197, 0x0000 }, /* R407 - FLL2 Loop Filter Test 1 */ { 0x00000198, 0x0000 }, /* R408 - FLL2 NCO Test 0 */ + { 0x00000199, 0x0000 }, /* R408 - FLL2 Control 7 */ { 0x000001A1, 0x0000 }, /* R417 - FLL2 Synchroniser 1 */ { 0x000001A2, 0x0000 }, /* R418 - FLL2 Synchroniser 2 */ { 0x000001A3, 0x0000 }, /* R419 - FLL2 Synchroniser 3 */ { 0x000001A4, 0x0000 }, /* R420 - FLL2 Synchroniser 4 */ { 0x000001A5, 0x0000 }, /* R421 - FLL2 Synchroniser 5 */ { 0x000001A6, 0x0000 }, /* R422 - FLL2 Synchroniser 6 */ + { 0x000001A7, 0x0001 }, /* R422 - FLL2 Synchroniser 7 */ { 0x000001A9, 0x0000 }, /* R425 - FLL2 Spread Spectrum */ { 0x000001AA, 0x0004 }, /* R426 - FLL2 GPIO Clock */ { 0x00000200, 0x0006 }, /* R512 - Mic Charge Pump 1 */ @@ -1392,6 +1396,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL1_CONTROL_4: case ARIZONA_FLL1_CONTROL_5: case ARIZONA_FLL1_CONTROL_6: + case ARIZONA_FLL1_CONTROL_7: case ARIZONA_FLL1_LOOP_FILTER_TEST_1: case ARIZONA_FLL1_NCO_TEST_0: case ARIZONA_FLL1_SYNCHRONISER_1: @@ -1400,6 +1405,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL1_SYNCHRONISER_4: case ARIZONA_FLL1_SYNCHRONISER_5: case ARIZONA_FLL1_SYNCHRONISER_6: + case ARIZONA_FLL1_SYNCHRONISER_7: case ARIZONA_FLL1_SPREAD_SPECTRUM: case ARIZONA_FLL1_GPIO_CLOCK: case ARIZONA_FLL2_CONTROL_1: @@ -1408,6 +1414,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL2_CONTROL_4: case ARIZONA_FLL2_CONTROL_5: case ARIZONA_FLL2_CONTROL_6: + case ARIZONA_FLL2_CONTROL_7: case ARIZONA_FLL2_LOOP_FILTER_TEST_1: case ARIZONA_FLL2_NCO_TEST_0: case ARIZONA_FLL2_SYNCHRONISER_1: @@ -1416,6 +1423,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL2_SYNCHRONISER_4: case ARIZONA_FLL2_SYNCHRONISER_5: case ARIZONA_FLL2_SYNCHRONISER_6: + case ARIZONA_FLL2_SYNCHRONISER_7: case ARIZONA_FLL2_SPREAD_SPECTRUM: case ARIZONA_FLL2_GPIO_CLOCK: case ARIZONA_MIC_CHARGE_PUMP_1: -- 1.7.2.5

2 4

[alsa-devel] [PATCH RESEND] mfd: arizona: Move regulator disable to after marking cache only
by Charles Keepax 07 Aug '13

07 Aug '13

If we disable DCVDD before we mark the cache as cache only, we might attempt to write to the chip whilst it is powered down and lose a write. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/arizona-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index 89a1153..5ac3aa4 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -438,9 +438,9 @@ static int arizona_runtime_suspend(struct device *dev) } } - regulator_disable(arizona->dcvdd); regcache_cache_only(arizona->regmap, true); regcache_mark_dirty(arizona->regmap); + regulator_disable(arizona->dcvdd); return 0; } -- 1.7.2.5

2 1

[alsa-devel] [PATCH v6 0/8] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 07 Aug '13

07 Aug '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) As the at91 chips are on their way to the common clock framework, I thought it may be better to retrieve the codec MCLK via this framework. So it needs Boris Brezillon patch set "ARM: at91: move to common clk framework" I've separated this patch because I don't know what is better: * applying the patchset with a hard coded clock, and once at91 has moved to common clock framework, update the code and change the DT binding. * wait for the common clock framework. Outstanding issue: If this machine driver will support more SoCs maybe we should name it differently. Maybe atmel_wm8731.c ? Since it could be merged with sam9g20_wm8731.c when the g20 is DT only (There will be some more work though). It has been tested on at91sam9g35ek, and should work on g15, g25, x25 and x35 also. patches applies on next-20130729 + "ARM: at91: move to common clk framework" Changes from v5: * Update the wm8731 DT binding documentation with available audio endpoints. * Remove Uwe's signed-off since it seems to be "an integration signed-off" * Remove rate constrains if freq==0 in wm8731. * Allocate dynamically card, dai and private structures. * Add a separate patch to set the oscillator value via DT. NB: this last patch needs Boris Brezillon patch set "ARM: at91: move to common clk framework" Changes from v4: * Move the rates constraints into the codec * Hard code the DAI format since it cannot be changed anyway (and remove the DT bindings accordingly) * Add some consistency in the functions naming (sam9x5ek/at91sam9x5ek/sam9x5) * Retrieve the SSC id from the SSC controller phandle * Clarify the atmel-ssc.txt documentation * Develop audio-routing documentation * Tidy up the includes in the machine driver Changes from v3: * Remove the use of macros in device tree documentation. Changes from v2: * Change atmel,sam9x5-audio-wm8731 to atmel,sam9x5-wm8731-audio for consistency with sam9g20 machine driver. * Use the snd_pcm_hw_constraint() API as suggested by Mark and Lars-Peter. * Remove the unnecessary snd_soc_dapm_sync() call as Lars-Peter suggested. * Add documentation for machine driver DT binding. * Update atmel-ssc DT binding documentation. * Add GPL license instead of just the "GPL" word (hope that's ok !) Changes from v1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): ASoC: atmel: machine driver for at91sam9x5-wm8731 boards Richard Genoud (7): ASoC: wm8731: add rates constraints Documentation: DT: update atmel SSC with DMA binding ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: enable SSC ARM: AT91: DTS: sam9x5ek: add sound configuration ASoC: sam9x5: get codec MCLK via device tree .../devicetree/bindings/misc/atmel-ssc.txt | 23 ++- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 39 ++++ Documentation/devicetree/bindings/sound/wm8731.txt | 9 + arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 32 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 218 ++++++++++++++++++++ sound/soc/codecs/wm8731.c | 60 +++++- 9 files changed, 392 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

3 20

[alsa-devel] [PATCH] ASoC: ad1980: Provide stub DAPM support
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> Since non-DAPM devices are not going to be supported provide DAPM input and output widgets and hook them up to the DAIs. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ad1980.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 43 insertions(+) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 89fcf7d..7257a88 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -96,6 +96,44 @@ SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; +static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_INPUT("CD_L"), +SND_SOC_DAPM_INPUT("CD_R"), +SND_SOC_DAPM_INPUT("AUX_L"), +SND_SOC_DAPM_INPUT("AUX_R"), +SND_SOC_DAPM_INPUT("LINE_IN_L"), +SND_SOC_DAPM_INPUT("LINE_IN_R"), + +SND_SOC_DAPM_OUTPUT("LFE_OUT"), +SND_SOC_DAPM_OUTPUT("CENTER_OUT"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_L"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_R"), +SND_SOC_DAPM_OUTPUT("MONO_OUT"), +SND_SOC_DAPM_OUTPUT("HP_OUT_L"), +SND_SOC_DAPM_OUTPUT("HP_OUT_R"), +}; + +static const struct snd_soc_dapm_route ad1980_dapm_routes[] = { + { "Capture", NULL, "MIC1" }, + { "Capture", NULL, "MIC2" }, + { "Capture", NULL, "CD_L" }, + { "Capture", NULL, "CD_R" }, + { "Capture", NULL, "AUX_L" }, + { "Capture", NULL, "AUX_R" }, + { "Capture", NULL, "LINE_IN_L" }, + { "Capture", NULL, "LINE_IN_R" }, + + { "LFE_OUT", NULL, "Playback" }, + { "CENTER_OUT", NULL, "Playback" }, + { "LINE_OUT_L", NULL, "Playback" }, + { "LINE_OUT_R", NULL, "Playback" }, + { "MONO_OUT", NULL, "Playback" }, + { "HP_OUT_L", NULL, "Playback" }, + { "HP_OUT_R", NULL, "Playback" }, +}; + static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -253,6 +291,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1980 = { .reg_cache_step = 2, .write = ac97_write, .read = ac97_read, + + .dapm_widgets = ad1980_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets), + .dapm_routes = ad1980_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes), }; static int ad1980_probe(struct platform_device *pdev) -- 1.8.4.rc1

2 1

[alsa-devel] horribly wrong code when --with-versioned is active
by John Spencer 07 Aug '13

07 Aug '13

if --with-versioned is active (default), a couple of macros in pcm.c start generating some completely broken, __old-prefixed wrapper functions, which then are getting used whenever the actual function is called. for example: snd_pcm_hw_params_set_buffer_time_near __OLD_NEAR1(snd_pcm_hw_params_set_buffer_time_near, unsigned int); -> #define __OLD_NEAR1(name, ret_type) __P_OLD_NEAR1(__old_, name, ret_type) -> #define __P_OLD_NEAR1(pfx, name, ret_type) \ ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type val, int *dir) \ { \ if (INTERNAL(name)(pcm, params, &val, dir) < 0) \ return 0; \ return (ret_type)val; \ } this will lead to generating a function __old_snd_pcm_hw_params_set_buffer_time_near which expands to unsigned int __old_snd_pcm_hw_params_set_buffer_time_near(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type val, int *dir) { if snd1_pcm_hw_params_set_buffer_time_near(pcm, params, &val, dir) < 0) return 0; return (ret_type)val; } there 2 bugs in there, 1) the real function gets passed a pointer to a pointer of unsigned, which is then happily dereferenced and the original pointer used as an int, and 2) the return type logic is wrong, in case of a non-error, the original pointer will be returned instead of 0 to indicate success. the right fix would look something like this: @@ -7190,19 +7192,15 @@ __OLD_GET1(snd_pcm_hw_params_get_tick_time_max, unsigned int, unsigned int); #define __P_OLD_NEAR(pfx, name, ret_type) \ -ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type val) \ +ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type *val) \ { \ - if (INTERNAL(name)(pcm, params, &val) < 0) \ - return 0; \ - return (ret_type)val; \ + return(INTERNAL(name)(pcm, params, val)); \ } #define __P_OLD_NEAR1(pfx, name, ret_type) \ -ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type val, int *dir) \ +ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type *val, int *dir) \ { \ - if (INTERNAL(name)(pcm, params, &val, dir) < 0) \ - return 0; \ - return (ret_type)val; \ + return (INTERNAL(name)(pcm, params, val, dir) < 0); \ } #define __OLD_NEAR(name, ret_type) __P_OLD_NEAR(__old_, name, ret_type) this is only half of the fix though, the "old" getter functions seem to misbehave as well. the misbehaviour can be inspected by using a small openal-soft (version 1.14) example code, and breaking on alsa_reset_playback and single stepping through the invocation of CHECK(snd_pcm_hw_params_set_buffer_time_near(data->pcmHandle, hp, &bufferLen, NULL));

2 3

[alsa-devel] [PATCH] ASoC: imx-mc13783: Make SND_SOC_IMX_MC13783 visible again
by Fabio Estevam 07 Aug '13

07 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Commit 02502da45 (ASoC: imx-mc13783: Depend on ARCH_ARM) introduced 'ARCH_ARM' as a dependency for SND_SOC_IMX_MC13783, but this is a non-existent symbol. This makes the selection of SND_SOC_IMX_MC13783 to be impossible. Use the correct 'ARM' symbol instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index c26449b..e15f771 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -194,7 +194,7 @@ config SND_SOC_IMX_SGTL5000 config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" - depends on MFD_MC13783 && ARCH_ARM + depends on MFD_MC13783 && ARM select SND_SOC_IMX_SSI select SND_SOC_IMX_AUDMUX select SND_SOC_MC13783 -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume
by brian.austin＠cirrus.com 06 Aug '13

06 Aug '13

From: Brian Austin <brian.austin(a)cirrus.com> Beep Volume Min/Max was backwards. Change to SOC_SONGLE_SX_TLV for correct volume representation Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 987f728..ee25f32 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -451,7 +451,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- 1.7.10.4

2 2

[alsa-devel] [PATCH 0/5] ASoC: R-Car sound DMA transfer support
by Kuninori Morimoto 06 Aug '13

06 Aug '13

Hi Mark These are DMAEngine support patches for R-Car sound Last patch cleanups empty SCU functions. I will work for regmap support soon, but I need more time for it. This patch doesn't use soc-dmaengine-pcm.c method, becouse current R-Car chip doesn't suuport dmaengine_prep_dma_cyclic(). Of course I can re-work to use it if chip/rcar-dmaengine supports it. these are based on mark/topic/rcar branch Kuninori Morimoto (5): ASoC: rsnd: remove platform dai and add dai_id on platform setting ASoC: rsnd: add common DMAEngine method ASoC: rsnd: SSI supports DMA transfer ASoC: rsnd: SSI supports DMA transfer via BUSIF ASoC: rsnd: scu: cleanup empty functions include/sound/rcar_snd.h | 25 ++++-- sound/soc/sh/rcar/core.c | 192 +++++++++++++++++++++++++++++++++++++++++----- sound/soc/sh/rcar/gen.c | 20 +++-- sound/soc/sh/rcar/rsnd.h | 44 +++++++++++ sound/soc/sh/rcar/scu.c | 163 ++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/ssi.c | 150 ++++++++++++++++++++++++++++++++++-- 6 files changed, 529 insertions(+), 65 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Fix keyclick feature
by Lars-Peter Clausen 06 Aug '13

06 Aug '13

The tlv320aic26 contains a embedded snd_soc_codec struct which is referenced in the keyclick code. That struct is never initialized though, replace the embedded struct with a pointer and use that in the keyclick code. Cc: Grant Likely <grant.likely(a)secretlab.ca> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/tlv320aic26.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b1f6982..b192cd4 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -29,7 +29,7 @@ MODULE_LICENSE("GPL"); /* AIC26 driver private data */ struct aic26 { struct spi_device *spi; - struct snd_soc_codec codec; + struct snd_soc_codec *codec; int master; int datfm; int mclk; @@ -330,7 +330,7 @@ static ssize_t aic26_keyclick_show(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); amp = (val >> 12) & 0x7; freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); @@ -346,9 +346,9 @@ static ssize_t aic26_keyclick_set(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + aic26_reg_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -360,8 +360,11 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); */ static int aic26_probe(struct snd_soc_codec *codec) { + struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, err, i, reg; + aic26->codec = codec; + dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); /* Reset the codec to power on defaults */ -- 1.8.0

3 6

[alsa-devel] [PATCH] ASoC: wm5110: Correct input OSR bits for wm5110
by Charles Keepax 06 Aug '13

06 Aug '13

The input OSR bits are specified differently for wm5110 than for current revs of wm5102. This patch corrects support for this on wm5110. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 20 ++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 12 ++++-------- 3 files changed, 25 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8dc6881..779a0ee 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -553,6 +553,26 @@ const struct soc_enum arizona_ng_hold = 4, arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); +static const char * const arizona_in_dmic_osr_text[] = { + "1.536MHz", "3.072MHz", "6.144MHz", +}; + +const struct soc_enum arizona_in_dmic_osr[] = { + SOC_ENUM_SINGLE(ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN2L_CONTROL, ARIZONA_IN2_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), +}; +EXPORT_SYMBOL_GPL(arizona_in_dmic_osr); + static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index fe1b794..b6b6d70 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -198,6 +198,7 @@ extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; extern const struct soc_enum arizona_ng_hold; +extern const struct soc_enum arizona_in_dmic_osr[]; extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fc41037..77fd531 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -58,14 +58,10 @@ static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); SOC_SINGLE(name " NG SPKDAT2R Switch", base, 11, 1, 0) static const struct snd_kcontrol_new wm5110_snd_controls[] = { -SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, - ARIZONA_IN4_OSR_SHIFT, 1, 0), +SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]), +SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), +SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]), +SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]), SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Fix class W controls
by Mark Brown 06 Aug '13

06 Aug '13

From: Mark Brown <broonie(a)linaro.org> Commit 6e0650 (ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it) went too far and converted a DAPM control to use SOC_SINGLE_EXT() which crashes. Revert that portion of the patch. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d76b056..29ad480 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1432,8 +1432,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, }; #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ - SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ - snd_soc_get_volsw, wm8994_put_class_w) + SOC_SINGLE_EXT(xname, reg, shift, 1, 0, \ + snd_soc_dapm_get_volsw, wm8994_put_class_w) static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- 1.8.4.rc0

2 2

[alsa-devel] [PATCH v2 0/9] ALSA: usb-audio: Refactor set_format()
by Eldad Zack 06 Aug '13

06 Aug '13

Hi Takashi, This is a low-priority series (applies against current for-next). This series refactors set_format() and moves the sync endpoint initilization code into separate function. The quirks for implicit feedback are moved to a separate function as well. The intended result is code that is easier to read without function changes. Changes from v2: * Cleaned up the patchset, indentations and whitespace fixes. * Patch #2: removed early returns when subs->sync_endpoint is already set. This change now follows original semantics. * Patch #9: I first corrected it thanks to Clemens Ladisch's feedback, but then I wasn't sure if it improves anything, so I dropped it entirely. * Dropped patch #10. Instead, added patch (now #9) to WARN_ON and return NULL in snd_usb_add_endpoint. Otherwise, the check should be done at each call site, and it "shouldn't happen" anyway. Cheers, Eldad v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-July/064268.html Eldad Zack (9): ALSA: usb-audio: remove disabled debug code in set_format ALSA: usb-audio: remove assignment from if condition ALSA: usb-audio: separate sync endpoint setting from set_format ALSA: usb-audio: move implicit fb quirks to separate function ALSA: usb-audio: reverse condition logic in set_sync_endpoint ALSA: usb-audio: do not initialize and check implicit_fb ALSA: usb-audio: remove is_playback from implicit feedback quirks ALSA: usb-audio: remove implicit_fb from quirk ALSA: usb-audio: WARN_ON when alts is passed as NULL sound/usb/endpoint.c | 3 + sound/usb/pcm.c | 243 +++++++++++++++++++++++++++++---------------------- 2 files changed, 141 insertions(+), 105 deletions(-) -- 1.8.1.5

2 8

[alsa-devel] [PATCH 1/2] ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume
by Brian Austin 06 Aug '13

06 Aug '13

Beep Volume Min/Max was backwards. Change to SOC_SONGLE_SX_TLV for correct volume representation Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 987f728..ee25f32 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -451,7 +451,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- 1.7.10.4

4 4

[alsa-devel] [PATCH] ASoC: wm8994: Fix class W controls
by Mark Brown 05 Aug '13

05 Aug '13

From: Mark Brown <broonie(a)linaro.org> Commit 6e0650 (ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it) went too far and converted a DAPM control to use SOC_SINGLE_EXT() which crashes. Revert that portion of the patch. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8994.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d76b056..f32998a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1432,8 +1432,10 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, }; #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ - SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ - snd_soc_get_volsw, wm8994_put_class_w) +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = wm8994_put_class_w, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- 1.8.4.rc0

2 2

[alsa-devel] [PATCH 2/2] ASoC: cs42l52: Add new TLV for Beep Volume
by Brian Austin 05 Aug '13

05 Aug '13

CS42L52 Beep control uses 2dB scale from -56dB Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index ee25f32..1e7d03f 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -195,6 +195,8 @@ static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -451,7 +453,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, + 0, 0x07, 0x1f, beep_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: cs42l52: Fixes for Beep Volume kcontrol (fwd)
by Brian Austin 05 Aug '13

05 Aug '13

This patch reorders the min/max values for BEEP Volume. Also changes SOC_SINGLE_TLV to SOC_SINGLE_SX_TLV Add DB_SCALE beep_tlv for volume control Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0f6f481..66d1e42 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(beep_tlv, -5600, 200, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -449,7 +451,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", + CS42L52_BEEP_VOL, 0, 0x07, 0x1f, beep_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- 1.7.10.4

2 6

[alsa-devel] [PATCH 1/3] ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture
by Lothar Waßmann 05 Aug '13

05 Aug '13

When a sound capture/playback is terminated while a playback/capture is running, power_vag_event() will clear SGTL5000_CHIP_ANA_POWER in the SND_SOC_DAPM_PRE_PMD event, thus muting the respective other channel. Don't clear SGTL5000_CHIP_ANA_POWER when both DAC and ADC are active to prevent this. Signed-off-by: Lothar Waßmann <LW(a)KARO-electronics.de> --- sound/soc/codecs/sgtl5000.c | 16 +++++++++++++--- 1 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c8b9090..a4b2973 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -153,6 +153,8 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP; + switch (event) { case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, @@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); - msleep(400); + /* + * Don't clear VAG_POWERUP, when both DAC and ADC are + * operational to prevent inadvertently starving the + * other one of them. + */ + if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) & + mask) != mask) { + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + } break; default: break; -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: pcm1792a: Fix build with !OF
by Mark Brown 05 Aug '13

05 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1792a.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 3f83bf9..72cf835 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -169,13 +169,11 @@ static struct snd_soc_dai_driver pcm1792a_dai = { .ops = &pcm1792a_dai_ops, }; -#ifdef CONFIG_OF static const struct of_device_id pcm1792a_of_match[] = { { .compatible = "ti,pcm1792a", }, { } }; MODULE_DEVICE_TABLE(of, pcm1792a_of_match); -#endif static const struct regmap_config pcm1792a_regmap = { .reg_bits = 8, @@ -231,7 +229,7 @@ static struct spi_driver pcm1792a_codec_driver = { .driver = { .name = "pcm1792a", .owner = THIS_MODULE, - .of_match_table = pcm1792a_of_match, + .of_match_table = of_match_ptr(pcm1792a_of_match), }, .id_table = pcm1792a_spi_ids, .probe = pcm1792a_spi_probe, -- 1.8.4.rc0

1 0

[alsa-devel] [RFC PATCH] ASoC: Add PCM1792A spi mode codec support
by Michael Trimarchi 05 Aug '13

05 Aug '13

Add PCM1792A spi mode codec support. This version implements only a subset of functionalities. Tested connect to a pandaboard ES device and based on recently pcm1681 codec. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- .../devicetree/bindings/sound/pcm1792a.txt | 18 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1792a.c | 245 ++++++++++++++++++++ sound/soc/codecs/pcm1792a.h | 26 +++ 5 files changed, 295 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 sound/soc/codecs/pcm1792a.c create mode 100644 sound/soc/codecs/pcm1792a.h diff --git a/Documentation/devicetree/bindings/sound/pcm1792a.txt b/Documentation/devicetree/bindings/sound/pcm1792a.txt new file mode 100644 index 0000000..b8e48e8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm1792a.txt @@ -0,0 +1,18 @@ +Texas Instruments pcm1792a DT bindings + +This driver supports the SPI bus. + +Required properties: + + - compatible: "ti,pcm1792a" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Examples: + + codec_spi: 1792a@0 { + compatible = "ti,pcm1792a"; + spi-max-frequency = <600000>; + }; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fb..45a9e3c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,6 +55,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM3008 + select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -295,6 +296,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_PCM3008 tristate +config SND_SOC_PCM1792A + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd806..1f8ee0b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -43,6 +43,7 @@ snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -172,6 +173,7 @@ obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c new file mode 100644 index 0000000..99a4ee2 --- /dev/null +++ b/sound/soc/codecs/pcm1792a.c @@ -0,0 +1,245 @@ +/* + * PCM1792A ASoC codec driver + * + * Copyright (c) Amarula Solutions B.V. 2013 + * + * Michael Trimarchi <michael(a)amarulasolutions.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; 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. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <linux/spi/spi.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/of_device.h> + +#include "pcm1792a.h" + +#define PCM1792A_DAC_VOL_LEFT 0x10 +#define PCM1792A_DAC_VOL_RIGHT 0x11 +#define PCM1792A_FMT_CONTROL 0x12 +#define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL + +#define PCM1792A_FMT_MASK 0x70 +#define PCM1792A_FMT_SHIFT 4 +#define PCM1792A_MUTE_MASK 0x01 +#define PCM1792A_MUTE_SHIFT 0 +#define PCM1792A_ATLD_ENABLE (1 << 7) + +static const struct reg_default pcm1792a_reg_defaults[] = { + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x50 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x01 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, +}; + +static bool pcm1792a_accessible_reg(struct device *dev, unsigned int reg) +{ + return reg >= 0x10 && reg <= 0x17; +} + +static bool pcm1792a_writeable_reg(struct device *dev, unsigned register reg) +{ + bool accessible; + + accessible = pcm1792a_accessible_reg(dev, reg); + + return accessible && reg != 0x16 && reg != 0x17; +} + +struct pcm1792a_private { + struct regmap *regmap; + unsigned int format; + unsigned int rate; +}; + +static int pcm1792a_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->format = format; + + return 0; +} + +static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, + PCM1792A_MUTE_MASK, !!mute); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1792a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x02; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x05; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x04; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE; + + ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, + PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val); + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_soc_dai_ops pcm1792a_dai_ops = { + .set_fmt = pcm1792a_set_dai_fmt, + .hw_params = pcm1792a_hw_params, + .digital_mute = pcm1792a_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1792a_dac_tlv, -12000, 50, 1); + +static const struct snd_kcontrol_new pcm1792a_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT, + PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0, + pcm1792a_dac_tlv), +}; + +static struct snd_soc_dai_driver pcm1792a_dai = { + .name = "pcm1792a-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PCM1792A_RATES, + .formats = PCM1792A_FORMATS, }, + .capture = { + .channels_min = 0, + .channels_max = 0, + }, + .ops = &pcm1792a_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1792a_of_match[] = { + { .compatible = "ti,pcm1792a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1792a_of_match); +#endif + +static const struct regmap_config pcm1792a_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 24, + .reg_defaults = pcm1792a_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1792a_reg_defaults), + .writeable_reg = pcm1792a_writeable_reg, + .readable_reg = pcm1792a_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = { + .controls = pcm1792a_controls, + .num_controls = ARRAY_SIZE(pcm1792a_controls), +}; + +static int pcm1792a_spi_probe(struct spi_device *spi) +{ + struct pcm1792a_private *pcm1792a; + int ret; + + pcm1792a = devm_kzalloc(&spi->dev, sizeof(struct pcm1792a_private), + GFP_KERNEL); + if (!pcm1792a) + return -ENOMEM; + + spi_set_drvdata(spi, pcm1792a); + + pcm1792a->regmap = devm_regmap_init_spi(spi, &pcm1792a_regmap); + if (IS_ERR(pcm1792a->regmap)) { + ret = PTR_ERR(pcm1792a->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&spi->dev, + &soc_codec_dev_pcm1792a, &pcm1792a_dai, 1); +} + +static int pcm1792a_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm1792a_spi_ids[] = { + { "pcm1792a", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm1792a_spi_ids); + +static struct spi_driver pcm1792a_codec_driver = { + .driver = { + .name = "pcm1792a", + .owner = THIS_MODULE, + .of_match_table = pcm1792a_of_match, + }, + .id_table = pcm1792a_spi_ids, + .probe = pcm1792a_spi_probe, + .remove = pcm1792a_spi_remove, +}; + +module_spi_driver(pcm1792a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM1792A driver"); +MODULE_AUTHOR("Michael Trimarchi <michael(a)amarulasolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h new file mode 100644 index 0000000..7a83d1f --- /dev/null +++ b/sound/soc/codecs/pcm1792a.h @@ -0,0 +1,26 @@ +/* + * definitions for PCM1792A + * + * Copyright 2013 Amarula Solutions + * + * 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. + */ + +#ifndef __PCM1792A_H__ +#define __PCM1792A_H__ + +#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S16_LE) + +#endif -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: core: init delayed_work for codec-codec links
by Richard Fitzgerald 05 Aug '13

05 Aug '13

If soc_probe_link_dais() finds a codec-codec link it skips creating a compress or pcm stream and links the DAIs together. But it must also init the delayed_work otherwise shutting down the DAI chain will fault when calling flush_delayed_work_sync() on the linked DAI. Pointing it to a dummy work callback is cleaner than taking special cases in the code to bypass the flush_delayed_work_sync(). Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4489c5b..bbe136c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -199,6 +199,10 @@ static ssize_t pmdown_time_set(struct device *dev, return count; } +static void dummy_delayed_work(struct work_struct *work) +{ +} + static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); #ifdef CONFIG_DEBUG_FS @@ -1428,6 +1432,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return ret; } } else { + INIT_DELAYED_WORK(&rtd->delayed_work, dummy_delayed_work); + /* link the DAI widgets */ play_w = codec_dai->playback_widget; capture_w = cpu_dai->capture_widget; -- 1.7.2.5

2 11

[alsa-devel] [PATCH] sound/sgtl5000: fix codec clock source transition to avoid clockless moment
by Oskar Schirmer 05 Aug '13

05 Aug '13

Powering down PLL before switching to a mode that does not use it is a bad idea. It would cause the SGTL5000 be without internal clock supply, especially on the I2C interface, which would make subsequent access to it fail. Thus, in case of not using PLL any longer, first set the mode control, then power down PLL. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/codecs/sgtl5000.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6c8a9e7..8ecb201 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -644,16 +644,19 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); } else { + /* otherwise, clk_ctrl must be set before pll power down */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + /* power down pll */ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, 0); } - /* if using pll, clk_ctrl must be set after pll power up */ - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); - return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: dapm: Implement mixer input auto-disable
by Lars-Peter Clausen 05 Aug '13

05 Aug '13

Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Rebased onto the latest asoc/topic/dapm --- include/sound/soc-dapm.h | 17 +++++++- include/sound/soc.h | 28 ++++++------ sound/soc/soc-dapm.c | 108 +++++++++++++++++++++++++++++++++++++++++------ 3 files changed, 125 insertions(+), 28 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3575721..c728d28 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -280,14 +280,26 @@ struct device; { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_AUTODISABLE(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_AUTODISABLE(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ @@ -484,6 +496,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_in, /* link to DAI structure */ snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ + snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ }; enum snd_soc_dapm_subclass { diff --git a/include/sound/soc.h b/include/sound/soc.h index b1e1f96..6201c6e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -30,13 +30,13 @@ /* * Convenience kcontrol builders */ -#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert) \ +#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert, xautodisable) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, .shift = shift_left, \ .rshift = shift_right, .max = xmax, .platform_max = xmax, \ - .invert = xinvert}) -#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ - SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert) + .invert = xinvert, .autodisable = xautodisable}) +#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \ + SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert}) @@ -52,7 +52,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ @@ -68,7 +68,7 @@ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -97,7 +97,7 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ - max, invert) } + max, invert, 0) } #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw, \ @@ -119,7 +119,7 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ - max, invert) } + max, invert, 0) } #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -190,14 +190,14 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } #define SOC_DOUBLE_EXT(xname, reg, shift_left, shift_right, max, invert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = \ - SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert) } + SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert, 0) } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -206,7 +206,7 @@ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -216,7 +216,7 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, \ - xmax, xinvert) } + xmax, xinvert, 0) } #define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -1088,7 +1088,9 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift, invert; + unsigned int reg, rreg, shift, rshift; + unsigned int invert:1; + unsigned int autodisable:1; }; struct soc_bytes { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5f64c16..0944bc4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -47,6 +47,15 @@ #define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; +static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink, + const char *control, + int (*connected)(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink)); +static struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget); + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -73,16 +82,18 @@ static int dapm_up_seq[] = { [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, [snd_soc_dapm_line] = 10, - [snd_soc_dapm_post] = 11, + [snd_soc_dapm_kcontrol] = 11, + [snd_soc_dapm_post] = 12, }; static int dapm_down_seq[] = { [snd_soc_dapm_pre] = 0, - [snd_soc_dapm_adc] = 1, - [snd_soc_dapm_hp] = 2, - [snd_soc_dapm_spk] = 2, - [snd_soc_dapm_line] = 2, - [snd_soc_dapm_out_drv] = 2, + [snd_soc_dapm_kcontrol] = 1, + [snd_soc_dapm_adc] = 2, + [snd_soc_dapm_hp] = 3, + [snd_soc_dapm_spk] = 3, + [snd_soc_dapm_line] = 3, + [snd_soc_dapm_out_drv] = 3, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, @@ -176,6 +187,7 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( struct dapm_kcontrol_data { unsigned int value; + struct snd_soc_dapm_widget *widget; struct list_head paths; struct snd_soc_dapm_widget_list *wlist; }; @@ -184,6 +196,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data; + struct soc_mixer_control *mc; data = kzalloc(sizeof(*data), GFP_KERNEL); if (!data) { @@ -195,6 +208,39 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, INIT_LIST_HEAD(&data->paths); + switch (widget->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + mc = (struct soc_mixer_control *)kcontrol->private_value; + + if (mc->autodisable) { + struct snd_soc_dapm_widget template; + + memset(&template, 0, sizeof(template)); + template.reg = mc->reg; + template.mask = (1 << fls(mc->max)) - 1; + template.shift = mc->shift; + if (mc->invert) + template.off_val = mc->max; + else + template.off_val = 0; + template.on_val = template.off_val; + template.id = snd_soc_dapm_kcontrol; + template.name = kcontrol->id.name; + + data->widget = snd_soc_dapm_new_control(widget->dapm, + &template); + if (!data->widget) { + kfree(data); + return -ENOMEM; + } + } + break; + default: + break; + } + kcontrol->private_data = data; return 0; @@ -203,6 +249,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); + kfree(data->widget); kfree(data->wlist); kfree(data); } @@ -246,6 +293,21 @@ static void dapm_kcontrol_add_path(const struct snd_kcontrol *kcontrol, struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); list_add_tail(&path->list_kcontrol, &data->paths); + + if (data->widget) { + snd_soc_dapm_add_path(data->widget->dapm, data->widget, + path->source, NULL, NULL); + } +} + +static bool dapm_kcontrol_is_powered(const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + if (!data->widget) + return true; + + return data->widget->power; } static struct list_head *dapm_kcontrol_get_path_list( @@ -275,6 +337,9 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, if (data->value == value) return false; + if (data->widget) + data->widget->on_val = value; + data->value = value; return true; @@ -515,6 +580,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_spk: case snd_soc_dapm_line: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: p->connect = 1; break; /* does affect routing - dynamically connected */ @@ -880,6 +946,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -975,6 +1042,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -1523,7 +1591,7 @@ static void dapm_widget_update(struct snd_soc_card *card) unsigned int wi; int ret; - if (!update) + if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) return; wlist = dapm_kcontrol_get_wlist(update->kcontrol); @@ -1668,6 +1736,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -2335,6 +2404,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -2717,6 +2787,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -2724,17 +2795,24 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + unsigned int val; if (snd_soc_volsw_is_stereo(mc)) dev_warn(codec->dapm.dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); - ucontrol->value.integer.value[0] = - (snd_soc_read(codec, reg) >> shift) & mask; + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + if (dapm_kcontrol_is_powered(kcontrol)) + val = (snd_soc_read(codec, reg) >> shift) & mask; + else + val = dapm_kcontrol_get_value(kcontrol); + mutex_unlock(&card->dapm_mutex); + if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; + ucontrol->value.integer.value[0] = max - val; + else + ucontrol->value.integer.value[0] = val; return 0; } @@ -2775,11 +2853,14 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - mask = mask << shift; - val = val << shift; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm_kcontrol_set_value(kcontrol, val); + + mask = mask << shift; + val = val << shift; + change = snd_soc_test_bits(codec, reg, mask, val); if (change) { update.kcontrol = kcontrol; @@ -3179,6 +3260,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: w->power_check = dapm_supply_check_power; break; default: -- 1.8.0

2 1

[alsa-devel] [PATCH v3 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 05 Aug '13

05 Aug '13

Changelog: v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 62 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1311 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 227 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1781 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

2 4

[alsa-devel] ASoC: omap-mcbsp 192000 freq
by Michael Trimarchi 05 Aug '13

05 Aug '13

Hi I'm working on having pcm1792a and omap platform. Right now I have already sent and rfc for the codec and I have a local build that change this file too to support 192000. What I don't understand is the meaning of: .period_bytes_min = 32, What is the relation between fifo_size and period_bytes_min? because at this frequency 32 bytes are not so much and the fifo of the port are >= 512 bytes. Is it possible to increase the period_bytes_min without violating the hw constraints? Michael .period_bytes_max = 64 * 1024, .periods_min = 2, .periods_max = 255, .buffer_bytes_max = 128 * 1024, >From f1e6fcfd81e6642ba5695c848ac6aa729e7467aa Mon Sep 17 00:00:00 2001 From: Michael Trimarchi <michael(a)amarulasolutions.com> Date: Mon, 5 Aug 2013 12:26:35 +0200 Subject: [PATCH] OMAP MCBSP can support 192000 frame rate. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- sound/soc/omap/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb..d285ddd 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -39,7 +39,7 @@ #include "mcbsp.h" #include "omap-mcbsp.h" -#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_192000) #define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ xhandler_get, xhandler_put) \ -- 1.7.9.5

1 0

[alsa-devel] (no subject)
by Chih-Chung Chang 05 Aug '13

05 Aug '13

This is a revised patch which also checks if the mic is external.

2 2

[alsa-devel] [PATCH 1/1] ALSA: hda - Fix jack gating when auto_{mute, mic} is suppressed.
by Chih-Chung Chang 05 Aug '13

05 Aug '13

The snd_hda_jack_set_gating_jack() call didn't work when auto_{mute,mic} is suppressed because (1) am_entry is not filled with nid of the mic pin. (2) The jacks are not created (by snd_hda_jack_detect_enable_callback) before the snd_hda_jack_set_gating_jack call. Now we use the first input pin nid directly, and create the jack if it doesn't exist yet. Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3fd2973..aa02c0f 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -247,8 +247,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid) { - struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid); - struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid); + struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid); + struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid); if (!gated || !gating) return -EINVAL; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..2b64de5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3258,11 +3258,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PROBE) { - if (snd_BUG_ON(!spec->gen.am_entry[1].pin || - !spec->gen.autocfg.hp_pins[0])) + int mic_pin = spec->gen.autocfg.inputs[0].pin; + int hp_pin = spec->gen.autocfg.hp_pins[0]; + + if (snd_BUG_ON(!mic_pin || !hp_pin)) return; - snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin, - spec->gen.autocfg.hp_pins[0]); + snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin); } } -- 1.8.3

3 2

[alsa-devel] [PATCH v2 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 05 Aug '13

05 Aug '13

Changelog: v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 62 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1311 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 227 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1781 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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Re: [alsa-devel] hda: sink info in case of AMD Radeon GPU card (no ELD?)
by Pierre Ossman 05 Aug '13

05 Aug '13

On Wed, 17 Apr 2013 14:04:00 +0200 Rafał Miłecki <zajec5(a)gmail.com> wrote: > > I tried hacking this sink info problem again. I've installed "fglrx" > on my AMD Radeon HD 6970M (BARTS == DCE5) and verified it sets SAD > registers: Any progress on this? I fiddled around with providing the ELD stuff out-of-band, but that's apparently not sufficient. I guess the widget capabilities property also needs to change. Did you check to see if that happened with fglrx? Rgds -- -- Pierre Ossman WARNING: This correspondence is being monitored by FRA, a Swedish intelligence agency. Make sure your server uses encryption for SMTP traffic and consider using PGP for end-to-end encryption.

2 1

[alsa-devel] [PATCH] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 05 Aug '13

05 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Add mute/unmute control in widget event and modify the power on/off sequence to avoid pop noise. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 387 ++++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/rt5640.h | 15 ++ 2 files changed, 352 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..3490f38 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -378,21 +378,119 @@ static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; +static const char * const rt5640_mute_mode[] = {"mute", "unmute"}; + +static const SOC_ENUM_SINGLE_DECL(rt5640_hp_enum, 0, 0, rt5640_mute_mode); +static const SOC_ENUM_SINGLE_DECL(rt5640_spo_enum, 0, 0, rt5640_mute_mode); + static const SOC_ENUM_SINGLE_DECL( rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); +static int rt5640_spo_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->spo_l_mute; + ucontrol->value.integer.value[1] = !rt5640->spo_r_mute; + + return 0; +} + +static int rt5640_spo_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_DIG1) & RT5640_PWR_CLS_D; + + if (ucontrol->value.integer.value[0]) { + rt5640->spo_l_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->spo_l_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, + RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->spo_r_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->spo_r_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + } + + return 0; +} + +static int rt5640_hp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->hp_l_mute; + ucontrol->value.integer.value[1] = !rt5640->hp_r_mute; + + return 0; +} + +static int rt5640_hp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_ANLG1); + + if (ucontrol->value.integer.value[0]) { + rt5640->hp_l_mute = false; + if (pow & RT5640_PWR_HP_L) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->hp_l_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->hp_r_mute = false; + if (pow & RT5640_PWR_HP_R) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->hp_r_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, RT5640_R_MUTE); + } + + + return 0; +} + static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("Speaker Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_spo_get, rt5640_spo_put), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("HP Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_hp_get, rt5640_hp_put), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -868,33 +966,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +1014,216 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_spo_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_r_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_spo_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_l_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, + RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | + RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(20000, 25000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, + RT5640_PWR_FV1 | RT5640_PWR_FV2); + + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CO_MASK | + RT5640_HP_SG_MASK, + RT5640_HP_CO_EN | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, pow; + + regmap_read(rt5640->regmap, RT5640_PWR_ANLG1, &pow); + if (rt5640->hp_l_mute || !(pow & RT5640_PWR_HP_L)) + val |= RT5640_L_MUTE; + if (rt5640->hp_r_mute || !(pow & RT5640_PWR_HP_R)) + val |= RT5640_R_MUTE; + /* headphone unmute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK, RT5640_SMT_TRIG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK, RT5640_RSTN_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_RSTN_DIS | RT5640_HP_L_SMT_EN | + RT5640_HP_R_SMT_EN); + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, val); + usleep_range(40000, 45000); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | RT5640_HP_R_SMT_DIS); + +} + +static void rt5640_pmd_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* headphone mute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK, RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK, RT5640_RSTP_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_RSTP_DIS | + RT5640_HP_L_SMT_EN | RT5640_HP_R_SMT_EN); + + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, + RT5640_L_MUTE | RT5640_R_MUTE); + usleep_range(40000, 45000); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | + RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | + RT5640_HP_R_SMT_DIS); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK | + RT5640_HP_CD_PD_MASK | + RT5640_HP_CO_MASK | + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_SMT_TRIG_DIS | + RT5640_HP_CD_PD_EN | + RT5640_HP_CO_DIS | RT5640_HP_CP_PD | + RT5640_HP_SG_EN | RT5640_HP_CB_PD); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5640_pmd_depop(codec); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1413,23 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("SPO L Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_l_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("SPO R Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_r_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1396,13 +1685,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"SPO L Amp", NULL, "SPOL MIX"}, + {"SPO R Amp", NULL, "SPOR MIX"}, + {"SPOLP", NULL, "SPO L Amp"}, + {"SPOLN", NULL, "SPO L Amp"}, + {"SPORP", NULL, "SPO R Amp"}, + {"SPORN", NULL, "SPO R Amp"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,7 +1703,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, + {"HPOL", NULL, "HP Amp"}, {"HPOL", NULL, "HP L Amp"}, + {"HPOR", NULL, "HP Amp"}, {"HPOR", NULL, "HP R Amp"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, @@ -1792,17 +2085,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(15000, 20000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +2135,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..cb6bce9 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) @@ -2087,6 +2098,10 @@ struct rt5640_priv { int pll_out; int dmic_en; + bool spo_l_mute; + bool spo_r_mute; + bool hp_l_mute; + bool hp_r_mute; }; #endif -- 1.8.1.1.439.g50a6b54

3 2

Re: [alsa-devel] [PATCH v3 4/4] ASoC: kirkwood: change kirkwood-i2s to mvebu-pcm-audio and DT doc
by Russell King - ARM Linux 04 Aug '13

04 Aug '13

On Wed, Jul 31, 2013 at 08:18:58AM +0200, Jean-Francois Moine wrote: > diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c > index 9844010..4f817a2 100644 > --- a/sound/soc/kirkwood/kirkwood-i2s.c > +++ b/sound/soc/kirkwood/kirkwood-i2s.c > @@ -25,7 +25,7 @@ > > #include "kirkwood.h" > > -#define DRV_NAME "kirkwood-i2s" > +#define DRV_NAME "mvebu-pcm-audio" This breaks non-DT users of this driver. The device in arch/arm/mach-kirkwood is left as "kirkwood-i2s" but the platform device is now called something different. > static struct of_device_id kirkwood_i2s_of_match[] = { > - { .compatible = "marvell,kirkwood-i2s" }, > + { .compatible = "marvell,mvebu-pcm-audio" }, You've only just introduced "marvell,kirkwood-i2s" in the previous patch. Why introduce something and then immediately change its name? This makes no sense what so ever.

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Re: [alsa-devel] [PATCH v3 2/4] ASoc: kirkwood: merge kirkwood-i2c and kirkwood-dma
by Russell King - ARM Linux 04 Aug '13

04 Aug '13

On Wed, Jul 31, 2013 at 08:18:06AM +0200, Jean-Francois Moine wrote: > To avoid the declaration of a 'kirkwood-pcm-audio' device in the DT, > this patch merges the kirkwood-i2c and kirkwood-dma drivers into one > module associated with 'kirkwood-i2s'. I suggest holding off on this stuff at the moment. I think Mark and Liam (who now have a whole raft of emails from me today) have some work to do to fix ASoC to work how they're saying it should work, because to me ASoC seems to contradict what they're telling me. To put it another way, it must be buggy. The DAPM stuff seems to be the worst thing since mouldy bread. I'm chasing what seems to be multiple bugs through this stuff, and many of them are not particularly nice. For example - we register multiple copies of DAPM widgets for the same set of declarations if both a CPU DAI and a platform share the same struct device, and thereby end up overwriting some pointers in the DAI structure. It seems that CPU DAIs themselves aren't supposed to have DAPM widgets, but the core creates some... but there's no explicit cleanup of them unlike the other DAPMs, and there's no debugfs for them. The bias levels are stuck at off/standby all the time, which makes any kind of PM with spdif-dit impossible - or even any routing between stream input and output. Basically, I'm tearing my hair out today looking at this stuff and getting nowhere fast - all I'm doing is finding more and more problems. For example, where a codec has no input/output widgets declared, it used to be powered up automatically by ASoC as a matter of course. Things like UDA134x and other things. Things like spdif-dit. That "mysteriously" stopped happening. ASoC: dapm: Treat DAI widgets like AIF widgets for power seems to be the cause, this results in such setups having zero connected inputs/outputs reported, which causes them to remain powered down - because what used to happen before was the DAI links would report their powered state depending on whether they were active, and they are set active in soc_dapm_stream_event(). Now, when playback widget for the Codec and CPU DAIs get marked as active. The playback widget is created as a snd_soc_dapm_dai_in. It's power check function is set to dapm_dac_check_power. Since this widget is active, it checks for connected outputs via is_connected_output_ep(). We drop through to the second switch statement (why do we have two there? They're both switching on the same damned variable and its not like a widget can change its ID.) This is where the different behaviour has appeared - when these were just a simple snd_soc_dapm_dai, we used to just do the snd_soc_dapm_suspend_check() here, but we don't anymore. This is a snd_soc_dapm_dai_in type of widget, so we fall through this switch statement now and start searching the paths. As these codecs have no paths, this ultimately ends up returning no connections. Hence why these codecs with no DAPM widgets declared but have PM support via the bias level stuff have stopped working. Now, about the spdif-dit, if we're going to have to add "pin" widgets to it, what the output of a SPDIF in terms of DAPM widgets? At a guess, it's a "codec pin" despite there being no codec and no "pin" on that codec in reality, and that "pin" is always active. With that "fixed" (rather, altered to a state where it behaves the same way as it used to before the above commit) if I set up a DAPM route between the CPU DAI playback stream, an AIF (for spdif output) and the spdif-dit playback stream, I see the playback streams marked active and powered up, but the AIF stream remains powered down: spdif-dit/dapm/Playback:Playback: On in 1 out 1 spdif-dit/dapm/Playback: stream Playback active spdif-dit/dapm/Playback: in "static" "spdif-playback" spdif-dit/dapm/bias_level:On kirkwood-audio.1/dapm/spdif-playback:spdif-playback: Off in 1 out 0 kirkwood-audio.1/dapm/spdif-playback: in "static" "dma-playback" kirkwood-audio.1/dapm/spdif-playback: out "static" "Playback" kirkwood-audio.1/dapm/dma-playback:dma-playback: On in 1 out 1 kirkwood-audio.1/dapm/dma-playback: stream dma-playback active kirkwood-audio.1/dapm/dma-playback: out "static" "spdif-playback" kirkwood-audio.1/dapm/dma-playback: out "static" "i2s-playback" It looks to me like the DAPM stuff is - in one plain and simple word - buggered. I've no idea what the right fixes are in this area. It needs someone like Mark or Liam who supposedly understand this to spend time checking that it actually operates as they _think_ it should operate, because at the moment it plainly doesn't.

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[alsa-devel] [PATCH V2 0/8] Add audio support on smdk5420
by Padmavathi Venna 04 Aug '13

04 Aug '13

Samsung has different versions of I2S introduced in different platforms. Each version has some new support added for multichannel, secondary fifo, s/w reset control, internal mux for rclk src clk and tdm support. Each newly added change has a quirk. So this patch adds all the required quirks as driver data and based on compatible string from dtsi fetches the quirks. This also adds i2s support on exynos5420 and make relevent changes in the dtsi files. Changes since V1: - Pass quirks as driver data and fetch the quirks based on compatible string from dtsi file as suggested by Tomasz Figa and Mark Brown - Make the I2S driver more flexible with respect to register access as suggested by Tomasz Figa and Mark Brown - Add 5420 support in the driver. - Modify the dtsi files with the corresponding compatible strings and removed the i2s quirks from 5250 dtsi file. - Updated the i2s Documentation with relevent changes and i2s versioning info. - Add i2s nodes on exynos5420.dtsi - Enable sound support on smdk5420 This patch set is dependent on the following dma and audio subsystem clk controller patches. http://comments.gmane.org/gmane.linux.kernel.samsung-soc/20661 http://comments.gmane.org/gmane.linux.kernel.samsung-soc/20668 This patch set is made based on Kukjin Kim for-next branch. Andrew Bresticker (1): ARM: dts: exynos5420: add i2s controllers Padmavathi Venna (7): platform: Increase platform name size ASoC: Samsung: I2S: Add quirks as driver data in I2S ARM: dts: Change i2s compatible string on exynos5250 ASoC: Samsung: I2S: Modify driver to give more flexibility ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 ARM: dts: wm8994: Add wm8994 support on smdk5420 ARM: dts: Enable sound support on smdk5420 .../devicetree/bindings/sound/samsung-i2s.txt | 25 ++- arch/arm/boot/dts/exynos5250.dtsi | 9 +- arch/arm/boot/dts/exynos5420-smdk5420.dts | 60 ++++++ arch/arm/boot/dts/exynos5420.dtsi | 44 +++++ include/linux/mod_devicetable.h | 2 +- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 51 ++++-- sound/soc/samsung/i2s.c | 205 +++++++++++++++----- 8 files changed, 312 insertions(+), 85 deletions(-) -- 1.7.4.4

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Re: [alsa-devel] [PATCH v3 1/4] ASoc: kirkwood: simplify probe error
by Russell King - ARM Linux 03 Aug '13

03 Aug '13

On Wed, Jul 31, 2013 at 08:17:39AM +0200, Jean-Francois Moine wrote: > The function kirkwood_i2s_dev_remove() may be used when probe fails. Looking at this deeper, I'm not happy with this. > +static int kirkwood_i2s_dev_remove(struct platform_device *pdev) > +{ > + struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); > + > + snd_soc_unregister_component(&pdev->dev); ... > @@ -519,30 +532,17 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) > > err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, > soc_dai, 1); > + if (err) { > + dev_err(&pdev->dev, "snd_soc_register_component failed\n"); > + goto fail; > + } > + return 0; > > +fail: > + kirkwood_i2s_dev_remove(pdev); What this means is that if snd_soc_register_component() fails, we end up calling snd_soc_unregister_component(). This may be fine with the way snd_soc_unregister_component() is currently implemented, but you're making the assumption that it's fine to call snd_soc_unregister_component() for a device which hasn't been registered. Technically, this is a layering violation, which makes this change fragile if the behaviour of snd_soc_unregister_component() changes in the future. For the sake of two calls in the error path, I don't think the benefits of this patch outweigh the risk.

2 1

Re: [alsa-devel] [PATCH v3 1/4] ASoc: kirkwood: simplify probe error
by Russell King - ARM Linux 03 Aug '13

03 Aug '13

On Wed, Jul 31, 2013 at 08:17:39AM +0200, Jean-Francois Moine wrote: > @@ -519,30 +532,17 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) > > err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, > soc_dai, 1); > - if (!err) > - return 0; > - dev_err(&pdev->dev, "snd_soc_register_component failed\n"); > - > - if (!IS_ERR(priv->extclk)) > - clk_disable_unprepare(priv->extclk); > - clk_disable_unprepare(priv->clk); > + if (err) { > + dev_err(&pdev->dev, "snd_soc_register_component failed\n"); > + goto fail; > + } > + return 0; This doesn't apply to mainline as there's a clk_put() there.

3 3

Re: [alsa-devel] [PATCH v3 3/4] ASoc: kirkwood: add DT support
by Russell King - ARM Linux 03 Aug '13

03 Aug '13

On Wed, Jul 31, 2013 at 08:18:32AM +0200, Jean-Francois Moine wrote: > The kirkwood audio driver is used without DT in the Kirkwood machine. > This patch adds a DT compatible definition for use in other Marvell > machines such as the Armada 88AP510 (Dove). > > Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> > --- > sound/soc/kirkwood/kirkwood-i2s.c | 27 ++++++++++++++++------ > 1 file changed, 20 insertions(+), 7 deletions(-) This misses documentation for the DT conversion. Documentation should be created at the same time as a driver is converted. I think you should combine patches 3 and 4 together given that you add the documentation and change the DT compatible name you've given this driver in this patch in patch 4. > > diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c > index a13fffb..234b5a1 100644 > --- a/sound/soc/kirkwood/kirkwood-i2s.c > +++ b/sound/soc/kirkwood/kirkwood-i2s.c > @@ -12,7 +12,6 @@ > > #include <linux/init.h> > #include <linux/module.h> > -#include <linux/platform_device.h> Why are you getting rid of this include? It still makes use of platform devices and the like, so it _should_ continue to include this. Don't rely on implicit includes.

1 0

Re: [alsa-devel] [PATCH v3 2/4] ASoc: kirkwood: merge kirkwood-i2c and kirkwood-dma
by Russell King - ARM Linux 03 Aug '13

03 Aug '13

On Wed, Jul 31, 2013 at 08:18:06AM +0200, Jean-Francois Moine wrote: > To avoid the declaration of a 'kirkwood-pcm-audio' device in the DT, > this patch merges the kirkwood-i2c and kirkwood-dma drivers into one > module associated with 'kirkwood-i2s'. This is broken unless you use my patch. You can't just stick the two drivers together because they both make use of the platform device's driver data field for two incompatible uses. I've already explained that, and I've even sent you a patch to fix that, but you seem to ignore anything that you didn't write. So... I'm going to work on this issue this weekend and merge the two drivers myself, and send Mark my own patches to do this.

1 1

[alsa-devel] [PATCH v2 7/9] ALSA: usb-audio: remove is_playback from implicit feedback quirks
by Eldad Zack 03 Aug '13

03 Aug '13

An implicit feedback endpoint can only be a capture source. The consumer (sink) of the implicit feedback endpoint is therefore limited to playback EPs. Check if the target endpoint is a playback first and remove redundant checks. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/pcm.c | 40 +++++++++++++++++++--------------------- 1 file changed, 19 insertions(+), 21 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bb2e0f5..af30e08 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -334,41 +334,39 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, { struct usb_host_interface *alts; struct usb_interface *iface; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; int implicit_fb = 0; unsigned int ep; + /* Implicit feedback sync EPs consumers are always playback EPs */ + if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ - if (is_playback) { - implicit_fb = 1; - ep = 0x81; - iface = usb_ifnum_to_if(dev, 3); + implicit_fb = 1; + ep = 0x81; + iface = usb_ifnum_to_if(dev, 3); - if (!iface || iface->num_altsetting == 0) - return -EINVAL; + if (!iface || iface->num_altsetting == 0) + return -EINVAL; - alts = &iface->altsetting[1]; - goto add_sync_ep; - } + alts = &iface->altsetting[1]; + goto add_sync_ep; break; case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ case USB_ID(0x0763, 0x2081): - if (is_playback) { - implicit_fb = 1; - ep = 0x81; - iface = usb_ifnum_to_if(dev, 2); + implicit_fb = 1; + ep = 0x81; + iface = usb_ifnum_to_if(dev, 2); - if (!iface || iface->num_altsetting == 0) - return -EINVAL; + if (!iface || iface->num_altsetting == 0) + return -EINVAL; - alts = &iface->altsetting[1]; - goto add_sync_ep; - } + alts = &iface->altsetting[1]; + goto add_sync_ep; } - if (is_playback && - attr == USB_ENDPOINT_SYNC_ASYNC && + if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && altsd->bInterfaceProtocol == 2 && altsd->bNumEndpoints == 1 && -- 1.8.1.5

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[alsa-devel] pcm1792a codec driver
by Michael Trimarchi 03 Aug '13

03 Aug '13

Hi Mark Yesterday I have done the pcm1792a codec driver and tested connected to the pandaboard es. The driver is working using spi/i2s connection with the codec and support l/r attenuation and digital mute. Other fmt are supported but not tested. Anyway it's a base of the complete driver. Hope that I can take again the hardware for feature release. Can I submit this initial support patch of it? Michael

1 0

[alsa-devel] end of playback ... design question
by Knut Petersen 02 Aug '13

02 Aug '13

Hi everybody! Although I looked at quite a bit of source code and documentation I have to admit that I am confused about one question: How should alsa playback be stopped in a way that _guarantees_ that no artifacts remaining in the drivers ringbuffer are played back again? Let´s assume we play stereo 16bit audio data using the rme96 driver, period size set to 0x800 frames. At the end of the penultimate frame playing will not be stopped even if there is only one frame of audio data remaining. Obviously that means 0x7ff frames of already played data is played again after that one valid frame, plus possibly a few that are played during the short time from interrupt request to the real stop of the playback hardware (about 5 frames on an old Pentium-M 1.2GHz). Now that could be fixed in different ways: A: Always ensure that there is a bit of silence (at least hardware buffer size plus a few frames) at the end of all audio data. Easy for the user , but only a workaround. B: At the rme96 driver level that condition can be detected and the relevant part of the buffer can be filled with zeros. I did that in an experimental change. It does work, but it´s definitely only a hack as it fixes the problem for only one device. C: The application playing audio data could pad all audio data to the period size with zeros. That would prevent playback of that max 0x7ff frames of garbage, but the few samples that are played during the time from interrupt request to the execution of the stop trigger do escape D: Don´t try to drain the buffer but pad with zeros and set a stop threshold. Workaround for applications should also work with older kernels. E: The alsa core could detect that condition and solve it equivalent to solution D I think that´s the proper solution. Did I miss something or do you agree? cu, Knut

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Re: [alsa-devel] HBR audio with intel HDA
by Anssi Hannula 02 Aug '13

02 Aug '13

02.08.2013 19:14, Ashecrow kirjoitti: > > Le 2 août 2013 à 15:42, Anssi Hannula a écrit : > >> 02.08.2013 16:11, Ashecrow kirjoitti: >>> Hi Anssi, >>> Thanks for your answer >>> >>> Test on kernel 3.8 + non-pcm bit in AES0: >>> aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 dts.spdif >>> => same result: no signal >> >> Hmh, OK. >> >>> With kernel upgrade to 3.9: same result >>> >>> >>> Note that I have tested with alsa* 1.025 and intel-hda driver included in kernel >>> >>> Do you have any other idea? >> >> You could try DEV=1 instead of DEV=0 (according to your log DEV=0 is >> correct, though), and truehd (thd.spdif from ffmpeg samples). >> >> If neither of those work, could you provide another alsa-info while >> playing back using AES0=0x06 (with alsa debug enabled again)? >> > I have already tried DEV=1 but it does the same. Moreover, let me note that I play DTS and dolby fine with DEV=0 in xbmc. So I guess it is the right device. OK. Note that in some cases you can hear stereo audio or AC-3/DTS even when using wrong DEV=x (but HBR or multichannel audio would not work). > see attached alsa-info.txt_dev0_kernel-3.8_aes0_dts-hd which is result of playing dts-hd spdif encapsulated with kernel 3.8 in debug mode and AES0=0x06 through command bellow > > aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 dts.spdif > > see attached alsa-info.txt_dev0_kernel-3.8_aes0_true-hd which is result of playing dts-hd spdif encapsulated with kernel 3.8 in debug mode and AES0=0x06 through command bellow > > aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 thd.spdif > Looks like your kernel is definitely too old to have working HBR on Intel (the "ASP channel x => slot y" messages should have x=y), so you'll need 3.9+. Unfortunately I don't see anything else wrong (and HBR+intel works for me), so I don't have anything else to suggest at the moment. Sorry. > Thanks for your help > >> >> >>> Thanks >>> >>> >>> Le 1 août 2013 à 18:37, Anssi Hannula a écrit : >>> >>>> 01.08.2013 19:29, ashecrow42(a)free.fr kirjoitti: >>>>> Hi there, >>>> >>>> Hi! >>>> >>>>> I have troubles using HBR audio (DTS-HD and Dolby TrueHD). With every >>>>> test, I get no signal in the AV receiver. I have tested using a dts-hd >>>>> encapsulated in spdif sample file (found there >>>>> http://samples.mplayerhq.hu/hdmi/) and commands bellow >>>>> >>>>> - aplay -D hdmi devices >>>>> - aplay -D hw devices >>>>> - aplay -D plughw devices >>>> >>>> You need to set the non-pcm bit in AES0. Use e.g. "-D hdmi:AES0=0x06". >>>> >>>>> Note that I use ubuntu raring with 3.8 kernel that includes Xingchao's >>>>> HBR kernel patches. >>>>> >>>>> >>>>> A. Configurations I have tested: >>>>> >>>>> >>>>> - hda driver upgrade (alsa devel ubuntu team daily builds) >>>>> >>>>> - upgrade alsa-lib & alsa-utils (alsa-testing launchpad) >>>>> >>>>> - kernel upgrade to 3.9 >>>> >>>> IIRC 3.9 (or newer) may indeed be needed since you have both hdmi and spdif. >>>> >>>>> - snd_hda_codec_realtek module backlist (to try to disable SPDIF) >>>>> /etc/modprobe.d/blacklist.conf >>>>> >>>>> - snd_hda_codec_realtek index = -2 in /etc/modprobe.d/alsa-base.conf >>>>> >>>>> - fresh new ubuntu raring install with no modifications at all: still >>>>> no signal >>>>> >>>>> B. Info: >>>>> >>>>> cat /proc/asound/card0/codec* | grep Codec >>>>> Codec: Realtek ALC892 >>>>> Codec: Intel CougarPoint HDMI >>>>> >>>>> I use zotac ZBOX ID82 (intel HM65 chipset with hdmi and spdif) >>>>> >>>>> with alsa debug and alsa debug verbose enabled in kernel I got the >>>>> alsa-info.txt attached file produced on raring with no modification >>>>> while playing dts-hd sample file with command: >>>>> >>>>> >>>>> aplay -D hdmi:CARD=PCH,DEV=0 -c8 -fs16_le -r192000 dts.spdif >>>> >>>> As per above you need to use >>>> aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 dts.spdif >>>> >>>>> >>>>> C. I add that: >>>>> >>>>> - In every test I have non HD audio protocols (DTS and dolby) working >>>>> well in the same configuration >>>>> >>>>> - I have tried with windows just to see if my hardware is broken and I >>>>> get DTS-HD audio with no problem >>>>> >>>>> - Takashi, I've read your HD-Audio.txt file and as far as I see, I have >>>>> no known error message in kernel log >>>>> >>>>> >>>>> => I and Wingchao Wang have conclued that it is driver or codec >>>>> related. Can you help me please? >>>>> >>>>> Thanks a lot >>>>> PS: sorry if you receive this email twice, I have sent it again as I >>>>> was not member of the list the first time. >>>> >>>> >>>> -- >>>> Anssi Hannula >>> >> >> >> -- >> Anssi Hannula > -- Anssi Hannula

2 1

[alsa-devel] [PATCH] ASoC: dapm: Fix empty list check in dapm_new_mux()
by Lars-Peter Clausen 02 Aug '13

02 Aug '13

list_first_entry() will always return a valid pointer, even if the list is empty. So the check whether path is NULL will always be false. So we end up calling dapm_create_or_share_mixmux_kcontrol() with a path struct that points right in the middle of the widget struct and by trying to modify the path the widgets memory will become corrupted. Fix this by using list_emtpy() to check if the widget doesn't have any paths. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This is the real fix for the rt5640 crash. --- sound/soc/soc-dapm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5f64c16..196cea7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -747,13 +747,14 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - path = list_first_entry(&w->sources, struct snd_soc_dapm_path, - list_sink); - if (!path) { + if (list_empty(&w->sources)) { dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name); return -EINVAL; } + path = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); if (ret < 0) return ret; -- 1.8.0

3 4

[alsa-devel] HBR audio with intel HDA
by ashecrow42＠free.fr 02 Aug '13

02 Aug '13

Hi there, I have troubles using HBR audio (DTS-HD and Dolby TrueHD). With every test, I get no signal in the AV receiver. I have tested using a dts-hd encapsulated in spdif sample file (found there http://samples.mplayerhq.hu/hdmi/) and commands bellow - aplay -D hdmi devices - aplay -D hw devices - aplay -D plughw devices Note that I use ubuntu raring with 3.8 kernel that includes Xingchao's HBR kernel patches. A. Configurations I have tested: - hda driver upgrade (alsa devel ubuntu team daily builds) - upgrade alsa-lib & alsa-utils (alsa-testing launchpad) - kernel upgrade to 3.9 - snd_hda_codec_realtek module backlist (to try to disable SPDIF) /etc/modprobe.d/blacklist.conf - snd_hda_codec_realtek index = -2 in /etc/modprobe.d/alsa-base.conf - fresh new ubuntu raring install with no modifications at all: still no signal B. Info: cat /proc/asound/card0/codec* | grep Codec Codec: Realtek ALC892 Codec: Intel CougarPoint HDMI I use zotac ZBOX ID82 (intel HM65 chipset with hdmi and spdif) with alsa debug and alsa debug verbose enabled in kernel I got the alsa-info.txt attached file produced on raring with no modification while playing dts-hd sample file with command: aplay -D hdmi:CARD=PCH,DEV=0 -c8 -fs16_le -r192000 dts.spdif C. I add that: - In every test I have non HD audio protocols (DTS and dolby) working well in the same configuration - I have tried with windows just to see if my hardware is broken and I get DTS-HD audio with no problem - Takashi, I've read your HD-Audio.txt file and as far as I see, I have no known error message in kernel log => I and Wingchao Wang have conclued that it is driver or codec related. Can you help me please? Thanks a lot PS: sorry if you receive this email twice, I have sent it again as I was not member of the list the first time.

3 3

[alsa-devel] [PATCH 01/10] ASoC: dapm: Move snd_soc_dapm_update from dapm context to card
by Lars-Peter Clausen 02 Aug '13

02 Aug '13

The update field of a DAPM context is only assigned while the card's dapm_mutex is locked, the field is also cleared again while the mutex is stil locked. So there will only ever be one DAPM context at a time with a non-NULL update field. So it is safe to move the update field from the DAPM context struct to the card struct. Doing so will allow further cleanups in this area. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 2 -- include/sound/soc.h | 1 + sound/soc/soc-dapm.c | 22 +++++++++++----------- 3 files changed, 12 insertions(+), 13 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e77c6f5..3397292 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -575,8 +575,6 @@ struct snd_soc_dapm_context { struct delayed_work delayed_work; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ - struct snd_soc_dapm_update *update; - void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7..b1e1f96 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1042,6 +1042,7 @@ struct snd_soc_card { /* Generic DAPM context for the card */ struct snd_soc_dapm_context dapm; struct snd_soc_dapm_stats dapm_stats; + struct snd_soc_dapm_update *update; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_card_root; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 366daef..7449e27 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1425,7 +1425,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, static void dapm_widget_update(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_update *update = dapm->update; + struct snd_soc_dapm_update *update = dapm->card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; unsigned int wi; @@ -1959,9 +1959,9 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm->update = update; + card->update = update; ret = soc_dapm_mux_update_power(dapm, kcontrol, mux, e); - dapm->update = NULL; + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2002,9 +2002,9 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm->update = update; + card->update = update; ret = soc_dapm_mixer_update_power(dapm, kcontrol, connect); - dapm->update = NULL; + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2693,11 +2693,11 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2775,11 +2775,11 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.reg = e->reg; update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2928,11 +2928,11 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.reg = e->reg; update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); -- 1.8.0

2 15

[alsa-devel] [PATCH] ALSA: hda - Fix missing fixup for Mac Mini with STAC9221
by Takashi Iwai 02 Aug '13

02 Aug '13

A fixup for Apple Mac Mini was lost during the adaption to the generic parser because the fallback for the generic ID 8384:7680 was dropped, and it resulted in the silence output (and maybe other problems). Unfortunately, just adding the missing subsystem ID wasn't enough, in this case. The subsystem ID of this machine is 0000:0100 (what Apple thought...?), and since snd_hda_pick_fixup() doesn't take the vendor id zero into account, the driver ignored this entry. Now it's fixed to regard the vendor id zero as a valid value. Reported-and-tested-by: Linus Torvalds <torvalds(a)linux-foundation.org> Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/patch_sigmatel.c | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7c11d46..48a9d00 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -860,7 +860,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } if (id < 0 && quirk) { - for (q = quirk; q->subvendor; q++) { + for (q = quirk; q->subvendor || q->subdevice; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); unsigned int mask = 0xffff0000 | q->subdevice_mask; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 92b9b43..6d1924c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2819,6 +2819,7 @@ static const struct hda_pintbl ecs202_pin_configs[] = { /* codec SSIDs for Intel Mac sharing the same PCI SSID 8384:7680 */ static const struct snd_pci_quirk stac922x_intel_mac_fixup_tbl[] = { + SND_PCI_QUIRK(0x0000, 0x0100, "Mac Mini", STAC_INTEL_MAC_V3), SND_PCI_QUIRK(0x106b, 0x0800, "Mac", STAC_INTEL_MAC_V1), SND_PCI_QUIRK(0x106b, 0x0600, "Mac", STAC_INTEL_MAC_V2), SND_PCI_QUIRK(0x106b, 0x0700, "Mac", STAC_INTEL_MAC_V2), -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: dapm: Check return value of snd_soc_cnew()
by Lars-Peter Clausen 01 Aug '13

01 Aug '13

snd_soc_cnew() can return NULL, so we should check the result before trying to use it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d74c356..b4fae87 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -671,8 +671,10 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], NULL, name, prefix); - kcontrol->private_free = dapm_kcontrol_free; kfree(long_name); + if (!kcontrol) + return -ENOMEM; + kcontrol->private_free = dapm_kcontrol_free; ret = dapm_kcontrol_data_alloc(w, kcontrol); if (ret) { -- 1.8.0

2 4

[alsa-devel] [PATCH] ASoC: rt5640: remove unused mux
by bardliao＠realtek.com 01 Aug '13

01 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Remove unused "INL Mux" and "INR Mux". Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- I meet a kernel panic issue if I don't remove the unused widget and modify something in rt5640_dapm_widgets[] and/or rt5640_dapm_routes[]. The message is "Unable to handle kernel paging request at virtual address 6b6b6bd3". PC is at dapm_clear_walk_output+0x18/0x7c LR is at dapm_clear_walk_output+0x18/0x7c If I remove those unused widget, ie. not appear in rt5640_dapm_routes[], everything is fine. So I think they should be removed. --- sound/soc/codecs/rt5640.c | 26 -------------------------- 1 file changed, 26 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index ce585e3..4db7314 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -737,29 +737,6 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; -/* INL/R source */ -static const char * const rt5640_inl_src[] = { - "IN2P", "MONOP" -}; - -static const SOC_ENUM_SINGLE_DECL( - rt5640_inl_enum, RT5640_INL_INR_VOL, - RT5640_INL_SEL_SFT, rt5640_inl_src); - -static const struct snd_kcontrol_new rt5640_inl_mux = - SOC_DAPM_ENUM("INL source", rt5640_inl_enum); - -static const char * const rt5640_inr_src[] = { - "IN2N", "MONON" -}; - -static const SOC_ENUM_SINGLE_DECL( - rt5640_inr_enum, RT5640_INL_INR_VOL, - RT5640_INR_SEL_SFT, rt5640_inr_src); - -static const struct snd_kcontrol_new rt5640_inr_mux = - SOC_DAPM_ENUM("INR source", rt5640_inr_enum); - /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -1005,9 +982,6 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { RT5640_PWR_IN_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR VOL", RT5640_PWR_VOL, RT5640_PWR_IN_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL Mux", SND_SOC_NOPM, 0, 0, &rt5640_inl_mux), - SND_SOC_DAPM_MUX("INR Mux", SND_SOC_NOPM, 0, 0, &rt5640_inr_mux), /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5640_PWR_MIXER, RT5640_PWR_RM_L_BIT, 0, rt5640_rec_l_mix, ARRAY_SIZE(rt5640_rec_l_mix)), -- 1.8.1.1.439.g50a6b54

4 3

[alsa-devel] [PATCH] ASoC: wm_adsp: Sanitize parameter passing
by Dimitris Papastamos 01 Aug '13

01 Aug '13

No need to hold on to the `codec' pointer. We can use the `dsp' pointer and grab all the information we need from there. This makes the parameters for the functions a bit more sane and idiomatic. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 31 ++++++++++++++++--------------- sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 17 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3168224..b38f350 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -227,7 +227,6 @@ struct wm_coeff_ctl_ops { struct wm_coeff_ctl { const char *name; - struct snd_soc_card *card; struct wm_adsp_alg_region region; struct wm_coeff_ctl_ops ops; struct wm_adsp *adsp; @@ -484,7 +483,7 @@ static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) struct snd_kcontrol_new *kcontrol; int ret; - if (!ctl || !ctl->name || !ctl->card) + if (!ctl || !ctl->name) return -EINVAL; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); @@ -498,14 +497,14 @@ static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) kcontrol->put = wm_coeff_put; kcontrol->private_value = (unsigned long)ctl; - ret = snd_soc_add_card_controls(ctl->card, + ret = snd_soc_add_card_controls(adsp->card, kcontrol, 1); if (ret < 0) goto err_kcontrol; kfree(kcontrol); - ctl->kcontrol = snd_soc_card_get_kcontrol(ctl->card, + ctl->kcontrol = snd_soc_card_get_kcontrol(adsp->card, ctl->name); list_add(&ctl->list, &adsp->ctl_list); @@ -777,11 +776,10 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } -static int wm_adsp_create_control(struct snd_soc_codec *codec, +static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *region) { - struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); struct wm_coeff_ctl *ctl; struct wmfw_ctl_work *ctl_work; char *name; @@ -840,7 +838,6 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, ctl->set = 0; ctl->ops.xget = wm_coeff_get; ctl->ops.xput = wm_coeff_put; - ctl->card = codec->card; ctl->adsp = dsp; ctl->len = region->len; @@ -877,7 +874,7 @@ err_name: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) +static int wm_adsp_setup_algs(struct wm_adsp *dsp) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -1065,7 +1062,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].dm); region->len -= be32_to_cpu(adsp1_alg[i].dm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region DM with ID %x\n", be32_to_cpu(adsp1_alg[i].alg.id)); @@ -1082,7 +1079,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].zm); region->len -= be32_to_cpu(adsp1_alg[i].zm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", be32_to_cpu(adsp1_alg[i].alg.id)); @@ -1111,7 +1108,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].xm); region->len -= be32_to_cpu(adsp2_alg[i].xm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region XM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1128,7 +1125,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].ym); region->len -= be32_to_cpu(adsp2_alg[i].ym); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region YM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1145,7 +1142,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].zm); region->len -= be32_to_cpu(adsp2_alg[i].zm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1365,6 +1362,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, int ret; int val; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, @@ -1399,7 +1398,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; @@ -1492,6 +1491,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, unsigned int val; int ret; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: /* @@ -1554,7 +1555,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 64087fb..d018dea 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -39,6 +39,7 @@ struct wm_adsp { int type; struct device *dev; struct regmap *regmap; + struct snd_soc_card *card; int base; int sysclk_reg; -- 1.8.3.4

2 1

[alsa-devel] [PATCH] ASoC: samsung: Make secondary I2S DAI device a child of primary
by Mark Brown 01 Aug '13

01 Aug '13

From: Mark Brown <broonie(a)linaro.org> More for neatness than for any great utility. Really we shouldn't be creating the child device at all, refactoring will follow. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a3e4b49..47e08dd 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1019,6 +1019,8 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) if (IS_ERR(i2s->pdev)) return NULL; + i2s->pdev->dev.parent = &pdev->dev; + platform_set_drvdata(i2s->pdev, i2s); ret = platform_device_add(i2s->pdev); if (ret < 0) -- 1.8.3.2

1 0

[alsa-devel] How do I get this patch?
by Christian Winger Nyman 01 Aug '13

01 Aug '13

I have the VB-99, so this patch would be fantastic. But I don't understand, will it be implemented in the Linux 3.10 kernel or what would be the easiest way for me to get it? I run Ubuntu Studio 13.04. Thanks for any guidance. http://comments.gmane.org/gmane.linux.alsa.devel/109660

2 1

[alsa-devel] [PATCH v2] ASoC: Add PCM1681 codec driver.
by Marek Belisko 01 Aug '13

01 Aug '13

PCM1681 can be controlled via I2C, SPI or in bootstrap mode (no control mode). This code add support only for I2C mode. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- Changes from v1: - fix de-emphasis handling (disable when invalid freq is used) - constify deemph freq - digital_mute refactoring - remove empty functions (probe and remove) - call set_deemph in hw_params - remove CONFIG_I2C - small refactoring - remove PCM1681 from controls .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1681.c | 313 ++++++++++++++++++++ 4 files changed, 334 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 sound/soc/codecs/pcm1681.c diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt new file mode 100644 index 0000000..4df1718 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt @@ -0,0 +1,15 @@ +Texas Instruments PCM1681 8-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,pcm1681". + - reg: The i2c address. Should contain <0x4c>. + +Examples: + + i2c_bus { + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d038a58..8ace02a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -56,6 +56,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1681 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -297,6 +298,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1681 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fab4086..d8fa18e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -43,6 +43,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -173,6 +174,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 0000000..27da41b --- /dev/null +++ b/sound/soc/codecs/pcm1681.c @@ -0,0 +1,313 @@ +/* + * PCM1681 ASoC codec driver + * + * Copyright (c) StreamUnlimited GmbH 2013 + * Marek Belisko <marek.belisko(a)streamunlimited.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; 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. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1681_SOFT_MUTE_ALL 0xff +#define PCM1681_DEEMPH_RATE_MASK 0x18 +#define PCM1681_DEEMPH_MASK 0x01 + +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ + +static const struct reg_default pcm1681_reg_defaults[] = { + { 0x01, 0xff }, + { 0x02, 0xff }, + { 0x03, 0xff }, + { 0x04, 0xff }, + { 0x05, 0xff }, + { 0x06, 0xff }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x06 }, + { 0x0A, 0x00 }, + { 0x0B, 0xff }, + { 0x0C, 0x0f }, + { 0x0D, 0x00 }, + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, +}; + +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x00) || (reg == 0x0f)); +} + +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) +{ + return pcm1681_accessible_reg(dev, reg) && + (reg != PCM1681_ZERO_DETECT_STATUS); +} + +struct pcm1681_private { + struct regmap *regmap; + unsigned int format; + /* Current deemphasis status */ + unsigned int deemph; + /* Current rate for deemphasis control */ + unsigned int rate; +}; + +static const int pcm1681_deemph[] = { 44100, 48000, 32000 }; + +static int pcm1681_set_deemph(struct snd_soc_codec *codec) +{ + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int i = 0, val = -1, enable = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) + if (pcm1681_deemph[i] == priv->rate) + val = i; + + if (val != -1) { + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_RATE_MASK, val); + enable = 1; + } else + enable = 0; + + /* enable/disable deemphasis functionality */ + return regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_MASK, enable); +} + +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return pcm1681_set_deemph(codec); +} + +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The PCM1681 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val; + + if (mute) + val = PCM1681_SOFT_MUTE_ALL; + else + val = 0; + + return regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); +} + +static int pcm1681_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) + val = 0x00; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x03; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); + if (ret < 0) + return ret; + + return pcm1681_set_deemph(codec); +} + +static const struct snd_soc_dai_ops pcm1681_dai_ops = { + .set_fmt = pcm1681_set_dai_fmt, + .hw_params = pcm1681_hw_params, + .digital_mute = pcm1681_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); + +static const struct snd_kcontrol_new pcm1681_controls[] = { + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 7/8 Playback Volume", + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + pcm1681_get_deemph, pcm1681_put_deemph), +}; + +struct snd_soc_dai_driver pcm1681_dai = { + .name = "pcm1681-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = PCM1681_PCM_RATES, + .formats = PCM1681_PCM_FORMATS, + }, + .ops = &pcm1681_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1681_dt_ids[] = { + { .compatible = "ti,pcm1681", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); +#endif + +static const struct regmap_config pcm1681_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1, + .reg_defaults = pcm1681_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), + .writeable_reg = pcm1681_writeable_reg, + .readable_reg = pcm1681_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { + .controls = pcm1681_controls, + .num_controls = ARRAY_SIZE(pcm1681_controls), +}; + +static const struct i2c_device_id pcm1681_i2c_id[] = { + {"pcm1681", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); + +static int pcm1681_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + int ret; + struct pcm1681_private *priv; + + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(client, priv); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681, + &pcm1681_dai, 1); +} + +static int pcm1681_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver pcm1681_i2c_driver = { + .driver = { + .name = "pcm1681", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1681_dt_ids), + }, + .id_table = pcm1681_i2c_id, + .probe = pcm1681_i2c_probe, + .remove = pcm1681_i2c_remove, +}; + +module_i2c_driver(pcm1681_i2c_driver); + +MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Marek Belisko <marek.belisko(a)streamunlimited.com>"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

2 1

[alsa-devel] ASoC fixes for v3.11
by Mark Brown 01 Aug '13

01 Aug '13

The following changes since commit 5ae90d8e467e625e447000cb4335c4db973b1095: Linux 3.11-rc3 (2013-07-28 20:53:33 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc3 for you to fetch changes up to 3fef7f795fff7ccc58d55a28315ca73305515884: Merge remote-tracking branch 'asoc/fix/wm0010' into asoc-linus (2013-07-31 21:07:23 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.11 A fix to make sure userspace knows when control writes have caused a change in value, fixing some UIs, plus a few few driver fixes mainly cleaning up issues from recent refactorings on less mainstream platforms. ---------------------------------------------------------------- Dimitris Papastamos (1): ASoC: wm0010: Fix resource leak Lars-Peter Clausen (2): ASoC: dapm: Fix return value of snd_soc_dapm_put_{volsw,enum_virt}() ASoC: bf5xx-ac97: Fix compile error with SND_BF5XX_HAVE_COLD_RESET Mark Brown (4): ASoC: bfin-ac97: Fix prototype error following AC'97 refactoring Merge remote-tracking branch 'asoc/fix/blackfin' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm0010' into asoc-linus Ralf Baechle (1): ASoC: au1x: Fix build sound/soc/au1x/ac97c.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 5 +++-- sound/soc/blackfin/bf5xx-ac97.h | 1 - sound/soc/codecs/wm0010.c | 24 ++++++++++++------------ sound/soc/soc-dapm.c | 5 ++--- 5 files changed, 18 insertions(+), 19 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: wm0010: Use DMA-safe memory for SPI transfers
by Dimitris Papastamos 01 Aug '13

01 Aug '13

We should be allocating our buffers for the SPI transfers from the DMA zone. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 10adc41..d5ebcb0 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -420,7 +420,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) xfer->codec = codec; list_add_tail(&xfer->list, &xfer_list); - out = kzalloc(len, GFP_KERNEL); + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); @@ -429,7 +429,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) } xfer->t.rx_buf = out; - img = kzalloc(len, GFP_KERNEL); + img = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); @@ -523,14 +523,14 @@ static int wm0010_stage2_load(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ - img = kzalloc(fw->size, GFP_KERNEL); + img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort2; } - out = kzalloc(fw->size, GFP_KERNEL); + out = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate output buffer\n"); ret = -ENOMEM; @@ -670,14 +670,14 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = -ENOMEM; len = pll_rec.length + 8; - out = kzalloc(len, GFP_KERNEL); + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); goto abort; } - img_swap = kzalloc(len, GFP_KERNEL); + img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img_swap) { dev_err(codec->dev, "Failed to allocate image buffer\n"); -- 1.8.3.4

2 1

[alsa-devel] [PATCH 0/4] ALSA: Implement core jack support for kcontrol
by Felipe F. Tonello 01 Aug '13

01 Aug '13

From: "Felipe F. Tonello" <eu(a)felipetonello.com> This patch series enable KControl support for core Jack and update current codecs. This will will enable moderns user-space daemons, such as PulseAudio(>=2.0), to have jack detection support via newer ALSA KControl interface. Patch 1: Actually implementation using KControl. Patch 2: It also ensures HDA drivers to use properly this new ALSA Jack implementation. Patches 3-4: This patch makes sure that all users of ALSA Jack uses this new standard way to report detections to user-space. Felipe F. Tonello (4): ALSA: Added jack detection kcontrol support ALSA: HDA: Updating jack implementation according new ALSA Jacks ALSA: SoC: Updating jack implementation according new ALSA Jacks ALSA: oxygen: Updating jack implementation according new ALSA Jacks include/sound/jack.h | 6 ++++-- include/sound/soc.h | 2 +- sound/core/Kconfig | 1 + sound/core/ctljack.c | 3 ++- sound/core/jack.c | 29 +++++++++++++++++++++++++++-- sound/pci/hda/Kconfig | 8 -------- sound/pci/hda/hda_codec.h | 2 -- sound/pci/hda/hda_jack.c | 38 +++++++++++++++++--------------------- sound/pci/hda/hda_jack.h | 4 +--- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 4 ++-- sound/soc/mid-x86/mfld_machine.c | 6 +++--- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 4 ++-- sound/soc/omap/omap-twl4030.c | 4 ++-- sound/soc/omap/rx51.c | 6 +++--- sound/soc/pxa/hx4700.c | 4 ++-- sound/soc/pxa/palm27x.c | 4 ++-- sound/soc/pxa/ttc-dkb.c | 8 ++++---- sound/soc/pxa/z2.c | 4 ++-- sound/soc/samsung/goni_wm8994.c | 4 ++-- sound/soc/samsung/h1940_uda1380.c | 4 ++-- sound/soc/samsung/littlemill.c | 10 +++++----- sound/soc/samsung/lowland.c | 6 +++--- sound/soc/samsung/rx1950_uda1380.c | 4 ++-- sound/soc/samsung/smartq_wm8987.c | 4 ++-- sound/soc/samsung/speyside.c | 6 +++--- sound/soc/samsung/tobermory.c | 4 ++-- sound/soc/soc-jack.c | 4 ++-- sound/soc/tegra/tegra_alc5632.c | 4 ++-- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 8 ++++---- 32 files changed, 107 insertions(+), 94 deletions(-) -- 1.8.1.4

6 30

[alsa-devel] [patch] ASoC: dapm: using freed pointer in dapm_kcontrol_add_widget()
by Dan Carpenter 01 Aug '13

01 Aug '13

There is a typo here so we end up using the old freed pointer instead of the newly allocated one. (If the "n" is zero then the code works, obviously). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Only needed in linux-next diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9abb3b2..d74c356 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -225,13 +225,13 @@ static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol, new_data = krealloc(data, sizeof(*data) + sizeof(widget) * n, GFP_KERNEL); - if (!data) + if (!new_data) return -ENOMEM; - data->wlist.widgets[n - 1] = widget; - data->wlist.num_widgets = n; + new_data->wlist.widgets[n - 1] = widget; + new_data->wlist.num_widgets = n; - kcontrol->private_data = data; + kcontrol->private_data = new_data; return 0; }

4 7

[alsa-devel] New dapm panic on beaver with next-20130730
by Olof Johansson 31 Jul '13

31 Jul '13

Hi Mark, Lars-Peter, I noticed the below panic on beaver (Tegra30). Seems like seaboard (Tegra20) is fine. This is with next-20130730, tegra_defconfig. I noticed a bunch of recent dapm changes in the asoc tree on the 29th, and I was able to bisect it down to: commit 5106b92f80a2cd37c52cffed80b4f5acfb77ccfd Author: Lars-Peter Clausen <lars(a)metafoo.de> Date: Mon Jul 29 17:14:00 2013 +0200 ASoC: dapm: Keep a list of paths per kcontrol Currently we store for each path which control (if any at all) is associated with that control. But we are only ever interested in the reverse relationship, i.e. we want to know all the paths a certain control is associated with. This is currently implemented by always iterating over all paths. This patch updates the code to keep a list for each control which contains all the paths that are associated with that control. This improves the run time of e.g. soc_dapm_mixer_update_power() and soc_dapm_mux_update_power() from O(n) (with n being the number of paths for the card) to O(1). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)linaro.org> Panic output is below. I need to figure out why I'm not actually getting a backtrace in panics any more as well so this is of somewhat limited use, unfortuantely. Running a brute force function lookup on all words on the stack gives a rough idea of call path though: c03b1bb4 dapm_clear_walk_output c03b3450 dapm_generic_check_power c03b56b0 dapm_power_widgets c03b608c snd_soc_dapm_new_widgets [c069f8f8 kallsyms_token_index] c03ae9fc soc_post_component_init c03b0dd0 soc_probe_link_dais c069abb0 kallsyms_token_index [c029ac24 devm_kzalloc_release] c03b1870 snd_soc_register_card c03c3964 tegra_rt5640_probe c0707b74 tegra_rt5640_driver_init c02999e0 platform_drv_probe c02999c8 platform_drv_probe c029878c driver_probe_device [ 1.845620] Unable to handle kernel paging request at virtual address 6b6b6bd3 [ 1.852855] pgd = c0004000 [ 1.855552] [6b6b6bd3] *pgd=00000000 [ 1.859125] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 1.864422] Modules linked in: [ 1.867473] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-rc3-next-20130730 #42 [ 1.875023] task: ef064a40 ti: ef074000 task.ti: ef074000 [ 1.880421] PC is at dapm_clear_walk_output+0x8/0x68 [ 1.885371] LR is at dapm_clear_walk_output+0x54/0x68 [ 1.890408] pc : [<c03bb77c>] lr : [<c03bb7c8>] psr: 20000113 [ 1.890408] sp : ef075ce8 ip : 00000000 fp : 00000010 [ 1.901858] r10: ef075d2c r9 : c076dd00 r8 : c076dbf8 [ 1.907066] r7 : ef075d34 r6 : eeb490b8 r5 : 6b6b6bd3 r4 : ef39c4b0 [ 1.913574] r3 : 00000069 r2 : 00000192 r1 : 6b6b6bd3 r0 : eeb490b8 [ 1.920084] Flags: nzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel [ 1.927372] Control: 10c5387d Table: 8000404a DAC: 00000015 [ 1.933101] Process swapper/0 (pid: 1, stack limit = 0xef074238) [ 1.939090] Stack: (0xef075ce8 to 0xef076000) [ 1.943434] 5ce0: ef39c4b0 ef3ad728 eeb490b8 c03bb7c8 ef3ad6c0 00000000 [ 1.951591] 5d00: 00000000 c03bd074 c076dcf8 ef3ad6c0 ef075d3c c03bf310 00000000 c076dcf8 [ 1.959747] 5d20: 00000000 eec537f0 c077ca0c ef075d2c ef075d2c ef075d34 ef075d34 ef075d3c [ 1.967904] 5d40: ef075d3c c01c298c eec66734 ef3ad584 c076dcd4 000000f0 ef3ad5a0 c076dce8 [ 1.976060] 5d60: c0587888 00000005 00000010 c03bfcd8 c0587888 00000000 c076dbf8 c076dc34 [ 1.984216] 5d80: 00000000 eeb56010 eeb49000 c076dbf8 00000000 c06ab0f8 c076dda8 00000000 [ 1.992374] 5da0: 00000002 c03b8618 00000000 00000000 00000000 eeb56010 00000000 ef21e3c0 [ 2.000531] 5dc0: c076dbf8 c076dda8 ef21e0c0 00000000 00000002 c03ba9e4 c076dc08 000000d0 [ 2.008688] 5de0: ef000500 c076a5a8 c076dd08 00000000 ef21e184 c076dcf8 c076a5ec eeb49000 [ 2.016844] 5e00: c076dce0 c076a5a8 ef3b1e8c eeb49034 c06a63ac c076dc18 c02a2a24 c076dbf8 [ 2.025001] 5e20: 000005d4 ef369a90 c17dd4f4 00000000 000000b3 ef074028 00000000 c03bb484 [ 2.033157] 5e40: c17dba28 c076db24 ef18ae10 c03cd59c c07bffd8 ef18ae10 00000000 c076db38 [ 2.041315] 5e60: c0712a54 c02a17e0 c02a17c8 c02a058c 00000000 ef18ae10 c076db38 ef18ae44 [ 2.049471] 5e80: 00000000 c02a0778 00000000 c076db38 c02a06ec c029eac8 ef06d0e0 ef181d74 [ 2.057627] 5ea0: c076db38 ef3af840 c07526d8 c029fd44 c06ab0d4 c076db38 00000006 c076db38 [ 2.065783] 5ec0: 00000006 c0777a40 c0777a40 c02a0d54 c07277c0 00000006 c0777a40 c0777a40 [ 2.073941] 5ee0: c0712a54 c00089f8 ef025900 c0659cf4 ef1160c0 c053b498 00000000 00000001 [ 2.082098] 5f00: 0000027c c073bd78 c17dd77b c0550080 000000b3 c003d884 00000000 c071d9dc [ 2.090254] 5f20: c06937c8 c06ed458 00000006 00000006 c003d198 c003d1f0 00000000 c07277c0 [ 2.098411] 5f40: 00000006 c0777a40 c0777a40 c06f027c 000000b3 c071d9dc c071d9d0 c06f0970 [ 2.106569] 5f60: 00000006 00000006 c06f027c ffffbfc7 ff7ffffd fffff7ff 9fffffff 00000000 [ 2.114726] 5f80: c0526f58 00000000 00000000 00000000 00000000 00000000 00000000 c0526f60 [ 2.122884] 5fa0: 00000000 00000000 c0526f58 c000e578 00000000 00000000 00000000 00000000 [ 2.131040] 5fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 2.139198] 5fe0: 00000000 00000000 00000000 00000000 00000013 00000000 bf7fcdff eefffeff [ 2.147359] Code: 1affffe1 eafffff5 e92d4070 e1a05001 (e5914000) [ 2.153588] ---[ end trace a5214c5b7e3cefc9 ]--- [ 2.158220] Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b

4 5

[alsa-devel] [PATCH v2] ASoC: omap-abe-twl6040: Remove support for pdata (legacy boot)
by Peter Ujfalusi 31 Jul '13

31 Jul '13

Just recently OMAP4 legacy boot support has been removed. No reason to keep the code used by the legacy boot (pdata based) since neither OMAP4 or OMAP5 can boot in this mode. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Changes since v1: - Remove the header file which is no longer in use. Regards, Peter include/linux/platform_data/omap-abe-twl6040.h | 49 --------- sound/soc/omap/omap-abe-twl6040.c | 133 ++++++++----------------- 2 files changed, 41 insertions(+), 141 deletions(-) delete mode 100644 include/linux/platform_data/omap-abe-twl6040.h diff --git a/include/linux/platform_data/omap-abe-twl6040.h b/include/linux/platform_data/omap-abe-twl6040.h deleted file mode 100644 index 5d298ac..0000000 --- a/include/linux/platform_data/omap-abe-twl6040.h +++ /dev/null @@ -1,49 +0,0 @@ -/** - * omap-abe-twl6040.h - ASoC machine driver OMAP4+ devices, header. - * - * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com - * All rights reserved. - * - * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.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. - * - * 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., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef _OMAP_ABE_TWL6040_H_ -#define _OMAP_ABE_TWL6040_H_ - -/* To select if only one channel is connected in a stereo port */ -#define ABE_TWL6040_LEFT (1 << 0) -#define ABE_TWL6040_RIGHT (1 << 1) - -struct omap_abe_twl6040_data { - char *card_name; - /* Feature flags for connected audio pins */ - u8 has_hs; - u8 has_hf; - bool has_ep; - u8 has_aux; - u8 has_vibra; - bool has_dmic; - bool has_hsmic; - bool has_mainmic; - bool has_submic; - u8 has_afm; - /* Other features */ - bool jack_detection; /* board can detect jack events */ - int mclk_freq; /* MCLK frequency speed for twl6040 */ -}; - -#endif /* _OMAP_ABE_TWL6040_H_ */ diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 70cd5c7..ebb1390 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -23,7 +23,6 @@ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/mfd/twl6040.h> -#include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> #include <linux/of.h> @@ -166,19 +165,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFMR", NULL, "Line In"}, }; -static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm, - int connected, char *pin) -{ - if (!connected) - snd_soc_dapm_disable_pin(dapm, pin); -} - static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; - struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; int ret = 0; @@ -203,24 +193,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - /* - * NULL pdata means we booted with DT. In this case the routing is - * provided and the card is fully routed, no need to mark pins. - */ - if (!pdata) - return ret; - - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - return ret; } @@ -274,13 +246,18 @@ static struct snd_soc_card omap_abe_card = { static int omap_abe_probe(struct platform_device *pdev) { - struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; + struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; int ret = 0; + if (!node) { + dev_err(&pdev->dev, "of node is missing.\n"); + return -ENODEV; + } + card->dev = &pdev->dev; priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); @@ -289,78 +266,50 @@ static int omap_abe_probe(struct platform_device *pdev) priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (node) { - struct device_node *dai_node; - - if (snd_soc_of_parse_card_name(card, "ti,model")) { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } - ret = snd_soc_of_parse_audio_routing(card, - "ti,audio-routing"); - if (ret) { - dev_err(&pdev->dev, - "Error while parsing DAPM routing\n"); - return ret; - } + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "Error while parsing DAPM routing\n"); + return ret; + } - dai_node = of_parse_phandle(node, "ti,mcpdm", 0); - if (!dai_node) { - dev_err(&pdev->dev, "McPDM node is not provided\n"); - return -EINVAL; - } - abe_twl6040_dai_links[0].cpu_dai_name = NULL; - abe_twl6040_dai_links[0].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_of_node = dai_node; - dai_node = of_parse_phandle(node, "ti,dmic", 0); - if (dai_node) { - num_links = 2; - abe_twl6040_dai_links[1].cpu_dai_name = NULL; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_of_node = dai_node; - priv->dmic_codec_dev = platform_device_register_simple( + priv->dmic_codec_dev = platform_device_register_simple( "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, - "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } - } else { - num_links = 1; - } - - priv->jack_detection = of_property_read_bool(node, - "ti,jack-detection"); - of_property_read_u32(node, "ti,mclk-freq", - &priv->mclk_freq); - if (!priv->mclk_freq) { - dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); } - - omap_abe_card.fully_routed = 1; - } else if (pdata) { - if (pdata->card_name) { - card->name = pdata->card_name; - } else { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } - - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; + num_links = 1; + } + + priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); + of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; } + card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); -- 1.8.3.2

2 1

[alsa-devel] [PATCH 0/3] ASoC: Add Freescale i.MX S/PDIF controller driver
by Nicolin Chen 31 Jul '13

31 Jul '13

* This series of patches add i.MX S/PDIF drivers, including CPU DAI, machine. * It also add two missing sample rates support for spdif dummy codec drivers Nicolin Chen (3): ASoC: codec: spdif: Add S20_3LE and S24_LE support for dummy codec drivers ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 63 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/codecs/spdif_receiver.c | 2 + sound/soc/codecs/spdif_transmitter.c | 5 +- sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1276 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 233 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 9 files changed, 1758 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 7

Re: [alsa-devel] [PATCH -next] ASoC: rcar: fix return value check in rsnd_gen1_probe()
by Mark Brown 31 Jul '13

31 Jul '13

On Tue, Jul 30, 2013 at 07:51:37AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > In case of error, the function devm_ioremap_resource() returns ERR_PTR() > and never returns NULL. The NULL test in the return value check should be > replaced with IS_ERR(), and also remove the dev_err call to avoid redundant > error message. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: fsl: Fix module build
by Fabio Estevam 31 Jul '13

31 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Building imx_v6_v7_defconfig with all audio drivers as modules results in the folowing build error: ERROR: "imx_pcm_fiq_init" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_dma_init" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_fiq_exit" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_dma_exit" [sound/soc/fsl/snd-soc-imx-ssi.ko] undefined! ERROR: "imx_pcm_dma_init" [sound/soc/fsl/snd-soc-fsl-ssi.ko] undefined! ERROR: "imx_pcm_dma_exit" [sound/soc/fsl/snd-soc-fsl-ssi.ko] undefined! Fix this by allowing SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA to be also built as modules and by using 'IS_ENABLED' to cover the module case. Reported-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 4 ++-- sound/soc/fsl/imx-pcm.h | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3a79d01..c26449b 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -109,11 +109,11 @@ config SND_SOC_IMX_SSI tristate config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SOC_IMX_AUDMUX diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 9136625..5d5b733 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -38,7 +38,7 @@ struct imx_pcm_fiq_params { struct snd_dmaengine_dai_dma_data *dma_params_tx; }; -#ifdef CONFIG_SND_SOC_IMX_PCM_DMA +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); void imx_pcm_dma_exit(struct platform_device *pdev); #else @@ -52,7 +52,7 @@ static inline void imx_pcm_dma_exit(struct platform_device *pdev) } #endif -#ifdef CONFIG_SND_SOC_IMX_PCM_FIQ +#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) int imx_pcm_fiq_init(struct platform_device *pdev, struct imx_pcm_fiq_params *params); void imx_pcm_fiq_exit(struct platform_device *pdev); -- 1.8.1.2

3 4

[alsa-devel] [PATCH v2] ASoC: wm0010: Fix resource leak
by Dimitris Papastamos 31 Jul '13

31 Jul '13

If kzalloc() fails for `img' then we are going to leak the memory for `out'. We are freeing the memory of all the tx/rx transfers but the tx/rx buf pointers will be NULL if we drop out earlier. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f5e8356..10adc41 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -410,6 +410,16 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) rec->command, rec->length); len = rec->length + 8; + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + ret = -ENOMEM; + goto abort; + } + + xfer->codec = codec; + list_add_tail(&xfer->list, &xfer_list); + out = kzalloc(len, GFP_KERNEL); if (!out) { dev_err(codec->dev, @@ -417,6 +427,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.rx_buf = out; img = kzalloc(len, GFP_KERNEL); if (!img) { @@ -425,24 +436,13 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.tx_buf = img; byte_swap_64((u64 *)&rec->command, img, len); - xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); - if (!xfer) { - dev_err(codec->dev, "Failed to allocate xfer\n"); - ret = -ENOMEM; - goto abort1; - } - - xfer->codec = codec; - list_add_tail(&xfer->list, &xfer_list); - spi_message_init(&xfer->m); xfer->m.complete = wm0010_boot_xfer_complete; xfer->m.context = xfer; - xfer->t.tx_buf = img; - xfer->t.rx_buf = out; xfer->t.len = len; xfer->t.bits_per_word = 8; -- 1.8.3.4

2 1

[alsa-devel] [PATCH] ASoC: core: No need to use strncmp()
by Dimitris Papastamos 31 Jul '13

31 Jul '13

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d68a486..9f60894 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2309,7 +2309,7 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, return NULL; list_for_each_entry(kctl, &card->controls, list) - if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) + if (!strcmp(kctl->id.name, name)) return kctl; return NULL; } @@ -3056,7 +3056,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, return -EINVAL; list_for_each_entry(kctl, &card->controls, list) { - if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { + if (!strcmp(kctl->id.name, name)) { found = 1; break; } -- 1.8.3.4

2 5

[alsa-devel] [PATCH] ASoC: wm0010: Fix resource leak
by Dimitris Papastamos 31 Jul '13

31 Jul '13

This fixes "https://bugzilla.kernel.org/show_bug.cgi?id=60650". Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f5e8356..285026c 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -410,6 +410,13 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) rec->command, rec->length); len = rec->length + 8; + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL | GFP_DMA); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + ret = -ENOMEM; + goto abort1; + } + out = kzalloc(len, GFP_KERNEL); if (!out) { dev_err(codec->dev, @@ -417,6 +424,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.rx_buf = out; img = kzalloc(len, GFP_KERNEL); if (!img) { @@ -425,24 +433,16 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.tx_buf = img; byte_swap_64((u64 *)&rec->command, img, len); - xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); - if (!xfer) { - dev_err(codec->dev, "Failed to allocate xfer\n"); - ret = -ENOMEM; - goto abort1; - } - xfer->codec = codec; list_add_tail(&xfer->list, &xfer_list); spi_message_init(&xfer->m); xfer->m.complete = wm0010_boot_xfer_complete; xfer->m.context = xfer; - xfer->t.tx_buf = img; - xfer->t.rx_buf = out; xfer->t.len = len; xfer->t.bits_per_word = 8; -- 1.8.3.4

2 3

[alsa-devel] [PATCH] SOUND: au1x: Fix build
by Ralf Baechle 31 Jul '13

31 Jul '13

d8b51c11ff5a70244753ba60abfd47088cf4dcd4 [ASoC: ac97c: Use module_platform_driver()] broke the build: CC sound/soc/au1x/ac97c.o /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__initcall_" and "&" does not give a valid preprocessing token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__exitcall_" and "&" does not give a valid preprocessing token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:334:31: warning: ‘au1xac97c_driver’ defined but not used [-Wunused-variable] make[5]: *** [sound/soc/au1x/ac97c.o] Error 1 make[4]: *** [sound/soc/au1x] Error 2 make[3]: *** [sound/soc] Error 2 Signed-off-by: Ralf Baechle <ralf(a)linux-mips.org> diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694..c8a2de1 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver");

2 1

[alsa-devel] [PATCH 1/1] conf/cards: add Audio Advantage Micro II
by Przemek Rudy 31 Jul '13

31 Jul '13

With related kernel chnages add extended suport for: - AES bits controls (via device suffix AESn= or tools like iecset) - SPDIF switch control, turning on/off the optical transmitter (via alsamixer tool) Signed-off-by: Przemek Rudy <prudy1(a)o2.pl> diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 0d93175..8a6d9ca 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -101,6 +101,41 @@ USB-Audio."Audiophile USB (tm)".pcm.iec958 { } } +# For this card we can (and must to get IEC61937) set AES bits +USB-Audio."MicroII".pcm.iec958 "cards.USB-Audio.Audio Advantage MicroII.pcm.iec958" +USB-Audio."Audio Advantage MicroII".pcm.iec958 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + + type hooks + slave.pcm { + type hw + card $CARD + } + + hooks.0 { + type ctl_elems + hook_args [ + { + interface PCM + name "IEC958 Playback Default" + preserve true + optional true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + { + name "IEC958 Playback Switch" + lock true + preserve true + value 1 + } + ] + } +} ################################################################################ -- 1.8.3.1

2 1

Re: [alsa-devel] [PATCH - Preliminary support for Xonar Xense (resubmitAdded support for Xonar Xense. Only speaker out and mic in working.
by Clemens Ladisch 30 Jul '13

30 Jul '13

Ian Dawes wrote: > Clemens Ladisch <clemens(a)ladisch.de> wrote: >> Ian Dawes wrote: >>> Clemens Ladisch <clemens(a)ladisch.de> wrote: >>>> Ian Dawes wrote: >>>>> >>>>> Signed-off-by: Ian Dawes <madeallup.gen(a)gmail.com> >>>> >>>> The commit message was squashed into the subject line. There's >>>> probably an empty line missing between them. >>> >>> I'm not sure what this means. >> >> Did you intend for the subject to be this long? > > Probably not. I added "resubmitted" to the subject And the "Added support ..." was supposed to be in the subject, too? > How should I rectify this? Ensure there is an empty line after the subject. Regards, Clemens

2 2

Re: [alsa-devel] [PATCH 7/9] ASoC: tegra: move to generic DMA DT binding
by Stephen Warren 30 Jul '13

30 Jul '13

On 07/30/2013 04:12 AM, Richard Zhao wrote: > On Sat, Jul 27, 2013 at 03:58:21AM +0800, Stephen Warren wrote: >> On 07/23/2013 10:10 PM, Richard Zhao wrote: >>> - add tegra_dma_filter_data to specify dma info >>> DMA DT binding needs the device that raise dma request and dma name >>> to request a dma channel. tegra30_i2s is a special case. It should be ahub >>> device and it also has dma name that cannot handled by ASoC dmaengine code. >>> So we pass the info using filter data in snd_dmaengine_dai_dma_data. >>> - change i2s/ac97 drivers to use generic DT binding >>> - tegra30_i2s: alloc ahub tx/rx fifo at driver probe time >> >>> diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c >> >> I don't understand why the FIFO alloc/free had to move from >> startup()/shutdown() to probe()/remove(). > > The call path is: > [<c0520e38>] (dump_stack+0x78/0xbc) from [<c03bb854>] (tegra_pcm_request_chan+0x10/0x34) > [<c03bb854>] (tegra_pcm_request_chan+0x10/0x34) from [<c03b6c2c>] (dmaengine_pcm_new+0xe0/0x134) > [<c03b6c2c>] (dmaengine_pcm_new+0xe0/0x134) from [<c03b5824>] (soc_new_pcm+0x2f0/0x3cc) > [<c03b5824>] (soc_new_pcm+0x2f0/0x3cc) from [<c03aa1a0>] (soc_probe_link_dais+0x358/0x39c) > [<c03aa1a0>] (soc_probe_link_dais+0x358/0x39c) from [<c03ab334>] (snd_soc_instantiate_card+0x35c/0x7a0) > [<c03ab334>] (snd_soc_instantiate_card+0x35c/0x7a0) from [<c03ab9dc>] (snd_soc_register_card+0x264/0x330) > [<c03ab9dc>] (snd_soc_register_card+0x264/0x330) from [<c03be350>] (tegra_rt5640_probe+0xf4/0x17c) > [<c03be350>] (tegra_rt5640_probe+0xf4/0x17c) from [<c0294290>] (platform_drv_probe+0x18/0x1c) > > So I need to request dma channel at probe time. The AHUB driver should look up and cache the DMA channel information during its probe() call, and defer probe() there if necessary. The FIFO alloc/free calls should simply use the cached information for the allocated channel. That way, the changes are isolated in the AHUB driver and shouldn't require any changes in the I2S driver. The DT parsing and DMA request lookup will then happen well before the snd_soc_instantiate_card() backtrace you mention above, since it will prevent the card from being instantiated until all the ASoC components are fully probed. >> This will prevent later >> changes from making the connectivity between the AHUB FIFOs and I2S >> modules more dynamic, which is kind of the whole point of the AHUB HW >> module. > > That's because it's i2s driver to register pcm device rather not ahub. > Ideally dma channel is only bound to ahub. Long-term, this should change. However, I believe if you make the changes I mention above, this won't be an issue. >>> diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h >> >>> +#define TEGRA_DMA_NAME_MAX_LEN 10 >>> + >>> +/* >>> + * Generic DMA DT binding needs the device that raise dma request and dma name >>> + * to request a dma channel. tegra30_i2s is a special case. It should be ahub >>> + * device and it also has dma name that cannot handled by ASoC dmaengine code. >>> + * So we pass the info using filter data in snd_dmaengine_dai_dma_data. >>> + */ >>> +struct tegra_dma_filter_data { >>> + struct device *dma_dev; >>> + char dma_name[TEGRA_DMA_NAME_MAX_LEN]; >>> +}; >> >> I agree with Lars-Peter's comment that it'd be better to enhance the >> ASoC dmaengine support to allow the channel name to be specified so we >> don't need to much custom code. > > Not only dma-name but also dma client device for this case. > >> And that should use "char *" not a >> fixed-length array. > > Ok, then there' two solutions. > > struct tegra30_i2s { > ... > struct snd_dmaengine_dai_dma_data capture_dma_data; > char dma_rx_name[10]; > struct tegra_dma_filter_data filter_data_rx; > ... > struct snd_dmaengine_dai_dma_data playback_dma_data; > char dma_tx_name[10]; > struct tegra_dma_filter_data filter_data_tx; > }; > > probe: i2s->filter_data_rx.dma_name = i2s.dma_rx_name > > or > > probe: i2s->filter_data_rx.dma_name = kasprintf(GFP_KERNEL, "channel%d" ...); > remove: kfree(i2s->filter_data_rx.dma.name) > > Which one do you like? Either is fine, although the first option has less dynamic allocation, so is probably better. The AHUB driver should be constructing the channel names though.

1 0

[alsa-devel] [PATCH 1/2] ASoC: bf5xx-ac97: Fix compile error
by Lars-Peter Clausen 30 Jul '13

30 Jul '13

If CONFIG_SND_BF5XX_HAVE_COLD_RESET is enabled building the blackfin ac97 driver fails with the following compile error: sound/soc/blackfin/bf5xx-ac97.c: In function ‘asoc_bfin_ac97_probe’: sound/soc/blackfin/bf5xx-ac97.c:297: error: expected ‘;’ before ‘{’ token sound/soc/blackfin/bf5xx-ac97.c:302: error: label ‘gpio_err’ used but not defined The issue was introduced in commit 6dab2fd7 ("ASoC: bf5xx-ac97: Convert to devm_gpio_request_one()"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ac97.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index efb1dae..e82eb37 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -294,11 +294,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) /* Request PB3 as reset pin */ ret = devm_gpio_request_one(&pdev->dev, CONFIG_SND_BF5XX_RESET_GPIO_NUM, - GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") { + GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET"); + if (ret) { dev_err(&pdev->dev, "Failed to request GPIO_%d for reset: %d\n", CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); - goto gpio_err; + return ret; } #endif -- 1.8.0

2 5

[alsa-devel] [PATCH v6 0/8] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 30 Jul '13

30 Jul '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) As the at91 chips are on their way to the common clock framework, I thought it may be better to retrieve the codec MCLK via this framework. So it needs Boris Brezillon patch set "ARM: at91: move to common clk framework" I've separated this patch because I don't know what is better: * applying the patchset with a hard coded clock, and once at91 has moved to common clock framework, update the code and change the DT binding. * wait for the common clock framework. Outstanding issue: If this machine driver will support more SoCs maybe we should name it differently. Maybe atmel_wm8731.c ? Since it could be merged with sam9g20_wm8731.c when the g20 is DT only (There will be some more work though). It has been tested on at91sam9g35ek, and should work on g15, g25, x25 and x35 also. patches applies on next-20130729 + "ARM: at91: move to common clk framework" Changes from v5: * Update the wm8731 DT binding documentation with available audio endpoints. * Remove Uwe's signed-off since it seems to be "an integration signed-off" * Remove rate constrains if freq==0 in wm8731. * Allocate dynamically card, dai and private structures. * Add a separate patch to set the oscillator value via DT. NB: this last patch needs Boris Brezillon patch set "ARM: at91: move to common clk framework" Changes from v4: * Move the rates constraints into the codec * Hard code the DAI format since it cannot be changed anyway (and remove the DT bindings accordingly) * Add some consistency in the functions naming (sam9x5ek/at91sam9x5ek/sam9x5) * Retrieve the SSC id from the SSC controller phandle * Clarify the atmel-ssc.txt documentation * Develop audio-routing documentation * Tidy up the includes in the machine driver Changes from v3: * Remove the use of macros in device tree documentation. Changes from v2: * Change atmel,sam9x5-audio-wm8731 to atmel,sam9x5-wm8731-audio for consistency with sam9g20 machine driver. * Use the snd_pcm_hw_constraint() API as suggested by Mark and Lars-Peter. * Remove the unnecessary snd_soc_dapm_sync() call as Lars-Peter suggested. * Add documentation for machine driver DT binding. * Update atmel-ssc DT binding documentation. * Add GPL license instead of just the "GPL" word (hope that's ok !) Changes from v1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): ASoC: atmel: machine driver for at91sam9x5-wm8731 boards Richard Genoud (7): ASoC: wm8731: add rates constraints Documentation: DT: update atmel SSC with DMA binding ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: enable SSC ARM: AT91: DTS: sam9x5ek: add sound configuration ASoC: sam9x5: get codec MCLK via device tree .../devicetree/bindings/misc/atmel-ssc.txt | 23 ++- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 39 ++++ Documentation/devicetree/bindings/sound/wm8731.txt | 9 + arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 32 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 218 ++++++++++++++++++++ sound/soc/codecs/wm8731.c | 60 +++++- 9 files changed, 392 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: ASoC: uda134x: fix codec driver
by Russell King 30 Jul '13

30 Jul '13

For some reason, the DAC/ADCs are not being powered up when I try and use the UDA1341 driver; this used to work. Looking back in the git history, I don't see anything obvious which would cause this regression. However, from dumping the register writes, it seems that the codec is powered down, and nothing calls set_bias_level to wake the codec up. Moreover, this driver hasn't had DAPM support added to it, which prevents platform drivers from taking advantage of DAPMs facilities. So, let's add DAPM support to the driver. As we move the power control for the DAC/ADC into DAPM, we no longer need it in set_bias_level() - this function just becomes a way to manipulate the power control and sync the register cache with the hardware at the appropriate point. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- I noticed this while checking the Assabet audio, which uses the UDA1341. I've also added the corresponding support to my Assabet audio driver. If this looks like the correct thing to do to fix this, please apply. sound/soc/codecs/uda134x.c | 88 ++++++++++++++++++++++++++----------------- 1 files changed, 53 insertions(+), 35 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 6d0aa44..c94d4c1 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -325,7 +325,6 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; struct uda134x_platform_data *pd = codec->control_data; int i; u8 *cache = codec->reg_cache; @@ -334,23 +333,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - /* ADC, DAC on */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg | 0x03); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_PREPARE: /* power on */ @@ -362,23 +344,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - /* ADC, DAC power off */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03)); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_OFF: /* power off */ @@ -450,6 +415,37 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; +/* UDA1341 has the DAC/ADC power down in STATUS1 */ +static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0), +}; + +/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */ +static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0), +}; + +/* Common DAPM widgets */ +static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("VINL1"), + SND_SOC_DAPM_INPUT("VINR1"), + SND_SOC_DAPM_INPUT("VINL2"), + SND_SOC_DAPM_INPUT("VINR2"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route uda134x_dapm_routes[] = { + { "ADC", NULL, "VINL1" }, + { "ADC", NULL, "VINR1" }, + { "ADC", NULL, "VINL2" }, + { "ADC", NULL, "VINR2" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, +}; + static const struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, @@ -485,6 +481,8 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; struct uda134x_platform_data *pd = codec->card->dev->platform_data; + const struct snd_soc_dapm_widget *widgets; + unsigned num_widgets; int ret; @@ -526,6 +524,22 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) else uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pd->model == UDA134X_UDA1341) { + widgets = uda1341_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1341_dapm_widgets); + } else { + widgets = uda1340_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1340_dapm_widgets); + } + + ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets); + if (ret) { + printk(KERN_ERR "%s failed to register dapm controls: %d", + __func__, ret); + kfree(uda134x); + return ret; + } + switch (pd->model) { case UDA134X_UDA1340: case UDA134X_UDA1344: @@ -599,6 +613,10 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = { .read = uda134x_read_reg_cache, .write = uda134x_write, .set_bias_level = uda134x_set_bias_level, + .dapm_widgets = uda134x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets), + .dapm_routes = uda134x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes), }; static int uda134x_codec_probe(struct platform_device *pdev) -- 1.7.4.4

2 1

[alsa-devel] [PATCH 1/3] ALSA: hda - Clearing jackpoll_interval avoid pending work
by Wang Xingchao 30 Jul '13

30 Jul '13

Clearing jackpoll_interval before calling cancel_delayed_work_sync(), otherwise the work will be triggered again and cause impact in hda_jackpoll_work(). The next patch will poll jack once even with jackpoll_interval=0. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e2481ba..0bc20ef 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec) return; if (spec->hp_work_active) { snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1); + codec->jackpoll_interval = 0; cancel_delayed_work_sync(&codec->jackpoll_work); spec->hp_work_active = false; - codec->jackpoll_interval = 0; } } -- 1.8.3.2

1 2

[alsa-devel] [PATCH - microII 1/1] ALSA: usb-audio: Add Audio Advantage Micro II
by Przemek Rudy 30 Jul '13

30 Jul '13

Signed-off-by: Przemek Rudy <prudy1(a)o2.pl> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ebe9144..d42a584 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -9,6 +9,8 @@ * Alan Cox (alan(a)lxorguk.ukuu.org.uk) * Thomas Sailer (sailer(a)ife.ee.ethz.ch) * + * Audio Advantage Micro II support added by: + * Przemek Rudy (prudy1(a)o2.pl) * * 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 @@ -30,6 +32,7 @@ #include <linux/usb.h> #include <linux/usb/audio.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/control.h> #include <sound/hwdep.h> @@ -1315,6 +1318,211 @@ static struct std_mono_table ebox44_table[] = { {} }; +/* Audio Advantage Micro II findings: + * + * Mapping spdif AES bits to vendor register.bit: + * AES0: [0 0 0 0 2.3 2.2 2.1 2.0] - default 0x00 + * AES1: [3.3 3.2.3.1.3.0 2.7 2.6 2.5 2.4] - default: 0x01 + * AES2: [0 0 0 0 0 0 0 0] + * AES3: [0 0 0 0 0 0 x 0] - 'x' bit is set basing on standard usb request + * (UAC_EP_CS_ATTR_SAMPLE_RATE) for Audio Devices + * + * power on values: + * r2: 0x10 + * r3: 0x20 (b7 is zeroed just before playback (except IEC61937) and set + * just after it to 0xa0, presumably it disables/mutes some analog + * parts when there is no audio.) + * r9: 0x28 + * + * Optical transmitter on/off: + * vendor register.bit: 9.1 + * 0 - on (0x28 register value) + * 1 - off (0x2a register value) + * + */ +static int snd_microii_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned int ep; + unsigned char data[3]; + int rate; + + ucontrol->value.iec958.status[0] = kcontrol->private_value & 0xff; + ucontrol->value.iec958.status[1] = (kcontrol->private_value >> 8) & 0xff; + ucontrol->value.iec958.status[2] = 0x00; + + /* use known values for that card: interface#1 altsetting#1 */ + iface = usb_ifnum_to_if(mixer->chip->dev, 1); + alts = &iface->altsetting[1]; + ep = get_endpoint(alts, 0)->bEndpointAddress; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_rcvctrlpipe(mixer->chip->dev, 0), + UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, + ep, + data, + sizeof(data)); + if (err < 0) + goto end; + + rate = data[0] | (data[1] << 8) | (data[2] << 16); + ucontrol->value.iec958.status[3] = (rate == 48000) ? + IEC958_AES3_CON_FS_48000 : IEC958_AES3_CON_FS_44100; + + err = 0; +end: + return err; +} + +static int snd_microii_spdif_default_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg; + unsigned long priv_backup = kcontrol->private_value; + + reg = ((ucontrol->value.iec958.status[1] & 0x0f) << 4) | + (ucontrol->value.iec958.status[0] & 0x0f); + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 2, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xfffff0f0; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0x0f) << 8; + kcontrol->private_value |= (ucontrol->value.iec958.status[0] & 0x0f); + + reg = (ucontrol->value.iec958.status[0] & IEC958_AES0_NONAUDIO) ? + 0xa0 : 0x20; + reg |= (ucontrol->value.iec958.status[1] >> 4) & 0x0f; + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 3, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xffff0fff; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0xf0) << 8; + + /* The frequency bits in AES3 cannot be set via register access. */ + + /* Silently ignore any bits from the request that cannot be set. */ + + err = (priv_backup != kcontrol->private_value); +end: + return err; +} + +static int snd_microii_spdif_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.iec958.status[0] = 0x0f; + ucontrol->value.iec958.status[1] = 0xff; + ucontrol->value.iec958.status[2] = 0x00; + ucontrol->value.iec958.status[3] = 0x00; + + return 0; +} + +static int snd_microii_spdif_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = !(kcontrol->private_value & 0x02); + + return 0; +} + +static int snd_microii_spdif_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg = ucontrol->value.integer.value[0] ? 0x28 : 0x2a; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 9, + NULL, + 0); + + if (!err) { + err = (reg != (kcontrol->private_value & 0x0ff)); + if (err) + kcontrol->private_value = reg; + } + + return err; +} + +static struct snd_kcontrol_new snd_microii_mixer_spdif[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_default_get, + .put = snd_microii_spdif_default_put, + .private_value = 0x00000100UL,/* reset value */ + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_mask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), + .info = snd_ctl_boolean_mono_info, + .get = snd_microii_spdif_switch_get, + .put = snd_microii_spdif_switch_put, + .private_value = 0x00000028UL,/* reset value */ + } +}; + +static int snd_microii_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_microii_mixer_spdif); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_microii_mixer_spdif[i], mixer)); + if (err < 0) + return err; + } + + return err; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1353,6 +1561,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_xonar_u1_controls_create(mixer); break; + case USB_ID(0x0d8c, 0x0103): /* Audio Advantage Micro II */ + err = snd_microii_controls_create(mixer); + break; + case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ err = snd_nativeinstruments_create_mixer(mixer, snd_nativeinstruments_ta6_mixers, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8b75bcf..2be7410 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3434,4 +3434,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* + * The original product_name is "USB Sound Device", however this name + * is also used by the CM106 based cards, so make it unique. + */ + USB_DEVICE(0x0d8c, 0x0103), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .product_name = "Audio Advantage MicroII", + .ifnum = QUIRK_NO_INTERFACE + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- 1.8.1.4

3 8

[alsa-devel] [PATCH - Preliminary support for Xonar Xense (resubmitAdded support for Xonar Xense. Only speaker out and mic in working.
by Ian Dawes 29 Jul '13

29 Jul '13

Signed-off-by: Ian Dawes <madeallup.gen(a)gmail.com> diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda..df99ec9 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8428) }, { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index c8c7f2c..2567810 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -499,6 +499,17 @@ static void xonar_stx_init(struct oxygen *chip) xonar_st_init_common(chip); } +static void xonar_xense_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_st_init(chip); +} + static void xonar_d2_cleanup(struct oxygen *chip) { xonar_disable_output(chip); @@ -1138,6 +1149,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x8428: + chip->model = model_xonar_st; + chip->model.shortname = "Xonar Xense"; + chip->model.chip = "AV100"; + chip->model.init = xonar_xense_init; + chip->model.dac_channels_pcm = 8; + chip->model.dac_channels_mixer = 8; + break; default: return -EINVAL; } -- 1.8.1.2

2 1

Re: [alsa-devel] snd_seq_system_info_get_cur_ports function
by Jose B 29 Jul '13

29 Jul '13

Dear Devs, I've been working on a wrapper to oversimplify using the alsa midi sequencer, and for it I was writing a function to quickly check if any new ports were registered on the sequencer after the application using the wrapper starts up. I noticed there is no function for getting the current port count but there is one for getting the current number of sequencer clients : snd_seq_system_info_get_cur_clients(const snd_seq_system_info_t *info) Shouldnt there be a corresponding function for getting the current port count as seen below? Or is it just missing from the documentation? snd_seq_system_info_get_cur_ports(const snd_seq_system_info_t *info) -- It doesn't show in code::blocks' autocomplete so i'm guessing it wasn't written yet. Is anyone willing to check this out please? It would be really handy to have this func. Thanks, and happy dev-ing Jose'

2 1

[alsa-devel] [PATCH] ASoC: bfin-ac97: Fix prototype error following AC'97 refactoring
by Mark Brown 29 Jul '13

29 Jul '13

From: Mark Brown <broonie(a)linaro.org> As part of the multiplatform refactoring for AC'97 the AC'97 bus ops were staticised meaning that the prototype (which was never needed) conflicts with the declaration causing build failures. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/blackfin/bf5xx-ac97.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 15c635e..0c3e22d 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -9,7 +9,6 @@ #ifndef _BF5XX_AC97_H #define _BF5XX_AC97_H -extern struct snd_ac97_bus_ops bf5xx_ac97_ops; extern struct snd_ac97 *ac97; /* Frame format in memory, only support stereo currently */ struct ac97_frame { -- 1.8.3.2

2 1

[alsa-devel] [PATCH 0/2] tea575x: Move from sound to media
by Ondrej Zary 29 Jul '13

29 Jul '13

Hello, TEA575x is neither a sound device nor an i2c device. Let's finally move it from sound/i2c/other to drivers/media/radio. Tested with snd-es1968, snd-fm801 and radio-sf16fmr2. -- Ondrej Zary

3 5

[alsa-devel] [PATCH 1/2] ASoC: core: Add snd_soc_card_get_kcontrol()
by Dimitris Papastamos 29 Jul '13

29 Jul '13

This is useful for drivers who want to grab a pointer to snd_kcontrol outside of the kcontrol callbacks. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 16 ++++++++++++++++ 2 files changed, 18 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7..b33d1de 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -475,6 +475,8 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, const char *long_name, const char *prefix); +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name); int snd_soc_add_codec_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_platform_controls(struct snd_soc_platform *platform, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d82ee38..d68a486 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2299,6 +2299,22 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev, return 0; } +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name) +{ + struct snd_card *card = soc_card->snd_card; + struct snd_kcontrol *kctl; + + if (unlikely(!name)) + return NULL; + + list_for_each_entry(kctl, &card->controls, list) + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) + return kctl; + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); + /** * snd_soc_add_codec_controls - add an array of controls to a codec. * Convenience function to add a list of controls. Many codecs were -- 1.8.3.4

2 2

[alsa-devel] [PATCH] ALSA: hda - Remove analog mic pin override from STAC9228 dell-bios quirk
by Takashi Iwai 29 Jul '13

29 Jul '13

The current fixup for dell-bios model with STAC9228 codec contains the override of pin 0x0c for analog mic. But this is actually just adding a bogus pin and confuses the parser. Better to remove it for the auto-mic switching. Meanwhile, for a possible regression, keep the old configuration as model=dell-bios-amic, so that people can test it again quickly. Tested on Dell 1420n laptop. Reported-and-tested-by: Eric Shattow <lucent(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_sigmatel.c | 14 ++++++++++++-- 2 files changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 809d72b..a46ddb8 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -244,6 +244,7 @@ STAC9227/9228/9229/927x 5stack-no-fp D965 5stack without front panel dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup + dell-bios-amic Fixes with Dell BIOS setup including analog mic volknob Fixes with volume-knob widget 0x24 auto BIOS setup (default) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e2f8359..8f6c357 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -158,6 +158,7 @@ enum { STAC_D965_VERBS, STAC_DELL_3ST, STAC_DELL_BIOS, + STAC_DELL_BIOS_AMIC, STAC_DELL_BIOS_SPDIF, STAC_927X_DELL_DMIC, STAC_927X_VOLKNOB, @@ -3228,8 +3229,6 @@ static const struct hda_fixup stac927x_fixups[] = { [STAC_DELL_BIOS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - /* configure the analog microphone on some laptops */ - { 0x0c, 0x90a79130 }, /* correct the front output jack as a hp out */ { 0x0f, 0x0227011f }, /* correct the front input jack as a mic */ @@ -3239,6 +3238,16 @@ static const struct hda_fixup stac927x_fixups[] = { .chained = true, .chain_id = STAC_927X_DELL_DMIC, }, + [STAC_DELL_BIOS_AMIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* configure the analog microphone on some laptops */ + { 0x0c, 0x90a79130 }, + {} + }, + .chained = true, + .chain_id = STAC_DELL_BIOS, + }, [STAC_DELL_BIOS_SPDIF] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3267,6 +3276,7 @@ static const struct hda_model_fixup stac927x_models[] = { { .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" }, { .id = STAC_DELL_3ST, .name = "dell-3stack" }, { .id = STAC_DELL_BIOS, .name = "dell-bios" }, + { .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" }, { .id = STAC_927X_VOLKNOB, .name = "volknob" }, {} }; -- 1.8.3.1

1 0

[alsa-devel] [patch] ALSA: usx2y: remove an unneeded check
by Dan Carpenter 29 Jul '13

29 Jul '13

The test here is always true because S[i].urb is an array not a pointer. Also it's bogus because the intent was to test: if (S->urb[i]) { instead of: if (S[i].urb) { Anway, usb_kill_urb() and usb_free_urb() accept NULL pointers so we can just remove this. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 1f9bbd5..5a51b18 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -305,11 +305,9 @@ static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S) { int i; for (i = 0; i < URBS_AsyncSeq; ++i) { - if (S[i].urb) { - usb_kill_urb(S->urb[i]); - usb_free_urb(S->urb[i]); - S->urb[i] = NULL; - } + usb_kill_urb(S->urb[i]); + usb_free_urb(S->urb[i]); + S->urb[i] = NULL; } kfree(S->buffer); }

2 1

[alsa-devel] [PATCH] ALSA: compress: fix the return value for SNDRV_COMPRESS_VERSION
by Vinod Koul 29 Jul '13

29 Jul '13

the return value of SNDRV_COMPRESS_VERSION always return default -ENOTTY as the return value was never updated for this call assign return value from put_user() Reported-by: Haynes <hgeorge(a)codeaurora.org> CC: stable(a)vger.kernel.org Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892..9896954 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): -- 1.7.0.4

2 1

[alsa-devel] [PATCH - Preliminary Xonar Xense support 1/1] Added support for Xonar Xense. Only speaker out and mic in working.
by Gen 29 Jul '13

29 Jul '13

Signed-off-by: Gen <madeallup.gen(a)gmail.com> diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda..df99ec9 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8428) }, { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index c8c7f2c..2567810 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -499,6 +499,17 @@ static void xonar_stx_init(struct oxygen *chip) xonar_st_init_common(chip); } +static void xonar_xense_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_st_init(chip); +} + static void xonar_d2_cleanup(struct oxygen *chip) { xonar_disable_output(chip); @@ -1138,6 +1149,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x8428: + chip->model = model_xonar_st; + chip->model.shortname = "Xonar Xense"; + chip->model.chip = "AV100"; + chip->model.init = xonar_xense_init; + chip->model.dac_channels_pcm = 8; + chip->model.dac_channels_mixer = 8; + break; default: return -EINVAL; } -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Selectively call snd_hda_shutup_pins()
by Takashi Iwai 29 Jul '13

29 Jul '13

Instead of calling snd_hda_shutup_pins() unconditionally, allow it be called in spec->shutup callback. In this way, we can avoid calling this function if it causes a problem like we see in the next patch following this. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..dbd59df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -282,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec) { alc_auto_setup_eapd(codec, false); msleep(200); + snd_hda_shutup_pins(codec); } /* generic EAPD initialization */ @@ -826,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec) if (spec && spec->shutup) spec->shutup(codec); - snd_hda_shutup_pins(codec); + else + snd_hda_shutup_pins(codec); } #define alc_free snd_hda_gen_free @@ -2573,15 +2575,13 @@ static void alc269_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (spec->codec_variant != ALC269_TYPE_ALC269VB) - return; - if (spec->codec_variant == ALC269_TYPE_ALC269VB) alc269vb_toggle_power_output(codec, 0); if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } + snd_hda_shutup_pins(codec); } static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, -- 1.8.3.1

1 1

[alsa-devel] [PATCH] ALSA: compress: fix the return value for SNDRV_COMPRESS_VERSION
by Vinod Koul 29 Jul '13

29 Jul '13

Reported-by: Haynes <hgeorge(a)codeaurora.org> CC: stable(a)vger.kernel.org Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892..9896954 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): -- 1.7.0.4

2 2

[alsa-devel] Broken headphone and speaker amplifier supplies for the rt5640 driver
by Lars-Peter Clausen 29 Jul '13

29 Jul '13

Hi, The rt5640 driver sets for both the headphone and speaker amplifier widget the shift to SND_SOC_NOPM, which is -1. This doesn't make much sense and breaks compilation with some changes I've made. I guess for the heaphone amplifier the right fix is to set the shift to RT5640_PWR_HA_BIT. For the speaker amplifier I'd assume that the right bit is RT5640_PWR_CLS_D_BIT. But the speaker amplifier widget also as an event callback in which it already seems to set/clear the bit. I think the right fix is to remove the register change from the event callback and set the shift to RT5640_PWR_CLS_D_BIT. But I neither have the hardware nor the datasheet, so this is all just an educated guess. - Lars

4 6

Re: [alsa-devel] Broken headphone and speaker amplifier suppliesforthe rt5640 driver
by Bard Liao 29 Jul '13

29 Jul '13

> -----Original Message----- > From: alsa-devel-bounces(a)alsa-project.org > [mailto:alsa-devel-bounces@alsa-project.org] On Behalf Of Stephen Warren > Sent: Saturday, July 27, 2013 12:58 AM > To: Lars-Peter Clausen > Cc: Bard Liao; Linux-ALSA > Subject: Re: [alsa-devel] Broken headphone and speaker amplifier suppliesfor > the rt5640 driver > > On 07/26/2013 09:06 AM, Lars-Peter Clausen wrote: > > Hi, > > > > The rt5640 driver sets for both the headphone and speaker amplifier > > widget the shift to SND_SOC_NOPM, which is -1. This doesn't make much > > sense and breaks compilation with some changes I've made. > > > > I guess for the heaphone amplifier the right fix is to set the shift > > to RT5640_PWR_HA_BIT ... > > If I make those changes (see below for the exact patch I tried), then both > headphones and speakers do appear to still work fine. Bard will have to > comment on whether the changes are actually correct though:-) The control bits for headphone amplifier and speaker amplifier are both correct. But rt5640 need a sequence for headphone depop. So, I would like to do something like this. - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), I will send a patch ASAP. Thanks. > > > diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c > > index ce585e3..e3ab0b2 100644 > > --- a/sound/soc/codecs/rt5640.c > > +++ b/sound/soc/codecs/rt5640.c > > @@ -899,8 +899,6 @@ static int spk_event(struct snd_soc_dapm_widget > > *w, > > > > switch (event) { > > case SND_SOC_DAPM_POST_PMU: > > - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, > > - 0x0001, 0x0001); > > regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, > > 0xf000, 0xf000); > > break; > > @@ -908,8 +906,6 @@ static int spk_event(struct snd_soc_dapm_widget > *w, > > case SND_SOC_DAPM_PRE_PMD: > > regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, > > 0xf000, 0x0000); > > - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, > > - 0x0001, 0x0000); > > break; > > > > default: > > @@ -1159,13 +1155,13 @@ static const struct snd_soc_dapm_widget > rt5640_dapm_widgets[] = { > > SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", > RT5640_PWR_ANLG1, > > RT5640_PWR_MA_BIT, 0, NULL, 0), > > SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, > > - SND_SOC_NOPM, 0, NULL, 0), > > + RT5640_PWR_HA_BIT, 0, NULL, 0), > > SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, > > RT5640_PWR_HP_L_BIT, 0, NULL, 0), > > SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, > > RT5640_PWR_HP_R_BIT, 0, NULL, 0), > > SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, > > - SND_SOC_NOPM, 0, spk_event, > > + RT5640_PWR_CLS_D_BIT, 0, spk_event, > > SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), > > /* Output Lines */ > > SND_SOC_DAPM_OUTPUT("SPOLP"), > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] [PATCH 0/6] R-Car sound driver support
by Kuninori Morimoto 29 Jul '13

29 Jul '13

Hi Mark These are basic Renesas R-Car sound driver support. Renesas R-Car series sound circuit consists of SSI and its peripheral. But this peripheral circuit is different between R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2). This driver aims at supporting both Gen1/Gen2. The basic driver style must be satisfactory. But it was tested on Gen1 board only at this point. Gen2 will be supported soon. And, it supports PIO transfer only at this point, DMA transfer will be supported soon. These are based on mark/for-next branch Kuninori Morimoto (6): ASoC: add Renesas R-Car core feature ASoC: add Renesas R-Car module feature ASoC: add Renesas R-Car Generation feature ASoC: add Renesas R-Car SCU feature ASoC: add Renesas R-Car ADG feature ASoC: add Renesas R-Car SSI feature include/sound/rcar_snd.h | 75 +++++ sound/soc/sh/Kconfig | 7 + sound/soc/sh/Makefile | 3 + sound/soc/sh/rcar/Makefile | 2 + sound/soc/sh/rcar/adg.c | 234 +++++++++++++++ sound/soc/sh/rcar/core.c | 699 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 306 +++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 253 ++++++++++++++++ sound/soc/sh/rcar/scu.c | 124 ++++++++ sound/soc/sh/rcar/ssi.c | 581 ++++++++++++++++++++++++++++++++++++ 10 files changed, 2284 insertions(+) Best regards --- Kuninori Morimoto

2 35

[alsa-devel] [PATCH 1/2] ASoC: lm4857: Use table based setup for DAPM and controls
by Lars-Peter Clausen 28 Jul '13

28 Jul '13

Let the ASoC core take care of registering the DAPM widget and routes as well as the controls. This makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/lm4857.c | 26 +++++++------------------- 1 file changed, 7 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 9f9f595..5ea2ed0 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -174,28 +174,9 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { static int lm4857_probe(struct snd_soc_codec *codec) { struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; codec->control_data = lm4857->i2c; - ret = snd_soc_add_codec_controls(codec, lm4857_controls, - ARRAY_SIZE(lm4857_controls)); - if (ret) - return ret; - - ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets, - ARRAY_SIZE(lm4857_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, lm4857_routes, - ARRAY_SIZE(lm4857_routes)); - if (ret) - return ret; - - snd_soc_dapm_new_widgets(dapm); - return 0; } @@ -207,6 +188,13 @@ static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { .reg_word_size = sizeof(uint8_t), .reg_cache_default = lm4857_default_regs, .set_bias_level = lm4857_set_bias_level, + + .controls = lm4857_controls, + .num_controls = ARRAY_SIZE(lm4857_controls), + .dapm_widgets = lm4857_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm4857_dapm_widgets), + .dapm_routes = lm4857_routes, + .num_dapm_routes = ARRAY_SIZE(lm4857_routes), }; static int lm4857_i2c_probe(struct i2c_client *i2c, -- 1.8.0

2 2

[alsa-devel] [PATCH] ASoC: compress: use soc_xxx handlers for metadata
by Vinod Koul 28 Jul '13

28 Jul '13

the compress metadata handlers were wrongly named sst_xxx Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-compress.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 06a8000..d220150 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -334,7 +334,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_set_metadata(struct snd_compr_stream *cstream, +static int soc_compr_set_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -347,7 +347,7 @@ static int sst_compr_set_metadata(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_get_metadata(struct snd_compr_stream *cstream, +static int soc_compr_get_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -364,8 +364,8 @@ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, - .set_metadata = sst_compr_set_metadata, - .get_metadata = sst_compr_get_metadata, + .set_metadata = soc_compr_set_metadata, + .get_metadata = soc_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, -- 1.7.0.4

2 1

[alsa-devel] RFC: support for 12 & 24Khz
by Vinod Koul 28 Jul '13

28 Jul '13

Hey Takashi, For compressed audio we also need to support the PCM rates of 12 and 24KHz. Looking at pcm.h these are not defined so we can simply add them at the end. But am worried about wider impact of adding these rates. Can you let me know if more is required to be done or below is fine diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 84b10f9..e418d8d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -126,6 +126,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ #define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +#define SNDRV_PCM_RATE_12000 <1<<13> /* 12000Hz */ +#define SNDRV_PCM_RATE_24000 <1<<14> /* 24000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..42600b0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1779,12 +1779,13 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_24000 != 1 << 14 #error "Change this table" #endif static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; + 48000, 64000, 88200, 96000, 176400, 192000, + 12000, 24000 }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .count = ARRAY_SIZE(rates), ~Vinod --

5 11

[alsa-devel] Garbage at end of playback: ecasound or alsa broken?
by Knut Petersen 28 Jul '13

28 Jul '13

Hi everybody! ====================================================================== Summary: A very short period (about 45ms) garbage is played at the end of every playback. ====================================================================== Hardware ======== mobo: AOpen i915GMm-hfs cpu: Pentium-M ram: 2GB intel hda, realtek codec rme digi96/pad onboard intel hda, realtek alc880 Software ======= opensuse 12.3 linux kernel 3.10.1, 3.11-rc1 ecasound, git ab6a9c3be5440b05f2c9eb240ee0d42578379f8d Steps to reproduce the problem: ========================= 1: generate test file =============== Start audacity, generate a 44.1 khz stereo track, one second audio, start 400 Hz, end 4000Hz, start amplitude 0.5, end amplitude 1.0 export to testin.wav as 16bit wav 2: playback test file and record it ========================= Setup soundcards and cabling, execute "ecasound -t 2 -f:16,2,44100 -a:1 -i testin.wav -o alsahw,0,0 -a:2 -i alsahw,0,0 -o testout.wav" 3:Expected result: ============== testout.wav should hold a proper recording of testin.wav, padded with silence at the end. 4: my result ========= testout.wav 1 - starts with about 90 Samples of silence (not nice, how can I avoid that?) 2 - continues with 1 second of properly recorded audio (expected) 3 - continues with about 1982 samples of audio (0,045s) (unexpected, severly broken) The garbage audio is part of the original audio, starting at about 26xxHz, ending at about 28xx Hz 4 - continues with silence to the end of the file (expected) Discussion ========= The same problem exists if I record rme96 to rme96, hda to rme96 or hda to hda. So it´s probably not a problem of the rme96 or hda alsa drivers. Is this problem known? Can anybody confirm the problem with different hardware? Should I blame alsa or ecasound? Any ideas? cu, Knut

2 1

[alsa-devel] [PATCH 1/3] ASoC: smdk_wm8994: Make driver name more unique
by Mark Brown 27 Jul '13

27 Jul '13

From: Mark Brown <broonie(a)linaro.org> Avoid collisions with other SMDK audio by using the CODEC name. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/smdk_wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 581ea4a..05c479c 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -200,7 +200,7 @@ MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); static struct platform_driver smdk_audio_driver = { .driver = { - .name = "smdk-audio", + .name = "smdk-audio-wm8894", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), }, @@ -212,4 +212,4 @@ module_platform_driver(smdk_audio_driver); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:smdk-audio"); +MODULE_ALIAS("platform:smdk-audio-wm8994"); -- 1.8.3.2

1 2

Re: [alsa-devel] [PATCH 35/50] media: usb: cx231xx: spin_lock in complete() cleanup
by Hans Verkuil 27 Jul '13

27 Jul '13

On 07/11/2013 11:05 AM, Ming Lei wrote: > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> > Cc: Hans Verkuil <hans.verkuil(a)cisco.com> > Cc: linux-media(a)vger.kernel.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > drivers/media/usb/cx231xx/cx231xx-audio.c | 6 ++++++ > drivers/media/usb/cx231xx/cx231xx-core.c | 10 ++++++---- > drivers/media/usb/cx231xx/cx231xx-vbi.c | 5 +++-- > 3 files changed, 15 insertions(+), 6 deletions(-) > > diff --git a/drivers/media/usb/cx231xx/cx231xx-audio.c b/drivers/media/usb/cx231xx/cx231xx-audio.c > index 81a1d97..58c1b5c 100644 > --- a/drivers/media/usb/cx231xx/cx231xx-audio.c > +++ b/drivers/media/usb/cx231xx/cx231xx-audio.c > @@ -136,6 +136,7 @@ static void cx231xx_audio_isocirq(struct urb *urb) > stride = runtime->frame_bits >> 3; > > for (i = 0; i < urb->number_of_packets; i++) { > + unsigned long flags; > int length = urb->iso_frame_desc[i].actual_length / > stride; > cp = (unsigned char *)urb->transfer_buffer + > @@ -158,6 +159,7 @@ static void cx231xx_audio_isocirq(struct urb *urb) > length * stride); > } > > + local_irq_save(flags); > snd_pcm_stream_lock(substream); Can't you use snd_pcm_stream_lock_irqsave here? Ditto for the other media drivers where this happens: em28xx and tlg2300. I've reviewed the media driver changes and they look OK to me, so if my comment above is fixed, then I can merge them for 3.12. Or are these changes required for 3.11? Regards, Hans > > dev->adev.hwptr_done_capture += length; > @@ -174,6 +176,7 @@ static void cx231xx_audio_isocirq(struct urb *urb) > period_elapsed = 1; > } > snd_pcm_stream_unlock(substream); > + local_irq_restore(flags); > } > if (period_elapsed) > snd_pcm_period_elapsed(substream); > @@ -224,6 +227,7 @@ static void cx231xx_audio_bulkirq(struct urb *urb) > stride = runtime->frame_bits >> 3; > > if (1) { > + unsigned long flags; > int length = urb->actual_length / > stride; > cp = (unsigned char *)urb->transfer_buffer; > @@ -242,6 +246,7 @@ static void cx231xx_audio_bulkirq(struct urb *urb) > length * stride); > } > > + local_irq_save(flags); > snd_pcm_stream_lock(substream); > > dev->adev.hwptr_done_capture += length; > @@ -258,6 +263,7 @@ static void cx231xx_audio_bulkirq(struct urb *urb) > period_elapsed = 1; > } > snd_pcm_stream_unlock(substream); > + local_irq_restore(flags); > } > if (period_elapsed) > snd_pcm_period_elapsed(substream); > diff --git a/drivers/media/usb/cx231xx/cx231xx-core.c b/drivers/media/usb/cx231xx/cx231xx-core.c > index 4ba3ce0..593b397 100644 > --- a/drivers/media/usb/cx231xx/cx231xx-core.c > +++ b/drivers/media/usb/cx231xx/cx231xx-core.c > @@ -798,6 +798,7 @@ static void cx231xx_isoc_irq_callback(struct urb *urb) > container_of(dma_q, struct cx231xx_video_mode, vidq); > struct cx231xx *dev = container_of(vmode, struct cx231xx, video_mode); > int i; > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -813,9 +814,9 @@ static void cx231xx_isoc_irq_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->video_mode.slock); > + spin_lock_irqsave(&dev->video_mode.slock, flags); > dev->video_mode.isoc_ctl.isoc_copy(dev, urb); > - spin_unlock(&dev->video_mode.slock); > + spin_unlock_irqrestore(&dev->video_mode.slock, flags); > > /* Reset urb buffers */ > for (i = 0; i < urb->number_of_packets; i++) { > @@ -842,6 +843,7 @@ static void cx231xx_bulk_irq_callback(struct urb *urb) > struct cx231xx_video_mode *vmode = > container_of(dma_q, struct cx231xx_video_mode, vidq); > struct cx231xx *dev = container_of(vmode, struct cx231xx, video_mode); > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -857,9 +859,9 @@ static void cx231xx_bulk_irq_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->video_mode.slock); > + spin_lock_irqsave(&dev->video_mode.slock, flags); > dev->video_mode.bulk_ctl.bulk_copy(dev, urb); > - spin_unlock(&dev->video_mode.slock); > + spin_unlock_irqrestore(&dev->video_mode.slock, flags); > > /* Reset urb buffers */ > urb->status = usb_submit_urb(urb, GFP_ATOMIC); > diff --git a/drivers/media/usb/cx231xx/cx231xx-vbi.c b/drivers/media/usb/cx231xx/cx231xx-vbi.c > index c027942..38e78f8 100644 > --- a/drivers/media/usb/cx231xx/cx231xx-vbi.c > +++ b/drivers/media/usb/cx231xx/cx231xx-vbi.c > @@ -306,6 +306,7 @@ static void cx231xx_irq_vbi_callback(struct urb *urb) > struct cx231xx_video_mode *vmode = > container_of(dma_q, struct cx231xx_video_mode, vidq); > struct cx231xx *dev = container_of(vmode, struct cx231xx, vbi_mode); > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -322,9 +323,9 @@ static void cx231xx_irq_vbi_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->vbi_mode.slock); > + spin_lock_irqsave(&dev->vbi_mode.slock, flags); > dev->vbi_mode.bulk_ctl.bulk_copy(dev, urb); > - spin_unlock(&dev->vbi_mode.slock); > + spin_unlock_irqrestore(&dev->vbi_mode.slock, flags); > > /* Reset status */ > urb->status = 0; >

2 1

[alsa-devel] [PATCH] ASoc: kirkwood: free the external clock when same as the internal one
by Jean-Francois Moine 27 Jul '13

27 Jul '13

Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/{kirkwood-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index ba72039..bbadf53 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c~ +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -495,6 +495,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { + devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); } else { dev_info(&pdev->dev, "found external clock\n"); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

Re: [alsa-devel] Decibel Levels for Storm Medusa 5.1 USB Headset
by Raymond Yau 27 Jul '13

27 Jul '13

> >>> I hope this Information helps. I have the Storm Medusa 5.1 USB Headset > >>> which reports wrong decibel levels to pulseaudio. My Alsa-Info are > >>> located at > >>> > >>> > > http://www.alsa-project.org/db/?f=a882f3db6769c5dfce83c266946eff52d1d55349 > >>> The Headset is card2 in the above dump. > >>> > >>> My decibel levels-fix > >>> > >>> [DecibelFix Speaker] > >>> db-values = 1:-30 37:3 > >> How did you define these values? > > http://mailman.alsa-project.org/pipermail/alsa-devel/2007-November/003867.h… > > > > If your headset have windows driver, need to find out which USB chip and > > look at the datasheet of the chip > > > > e.g. cm106l ’s datasheet contain a table of dac volume cortol > Ok looks like cm6206 is only in newer versions of the headset in older > (like mine) there is cm106l , maybe there are other chips as well but i > don't know. http://mailman.alsa-project.org/pipermail/alsa-devel/2007-November/003877.h… Seem same USB audio id idVendor 0x0d8c C-Media Electronics, Inc. idProduct 0x0006 Do you find out the audio chip from product spec, user manual or windows driver ? Master volume control by default; per-channel volume control by C-Media driver Refer to the function block diagram , dac with volume controls 0 to -45dB There is a table of dac volume control in linear scale

1 0

[alsa-devel] [PATCH 0/9] ARM: tegra: move to generic DMA DT binding
by Richard Zhao 26 Jul '13

26 Jul '13

The patch series aim to move apbdma to generic DT binding. Changes: - It add dmas/dma-names properties while leave nvidia,dma-request-selector to be compatible. - update apbdma driver and dma client drivers - ASoC tegra adds a new struct to pass device and dma-name. - at last, remove legacy nvidia,dma-request-selector Richard Zhao (9): ARM: dts: add generic DMA DT binding for tegra apbdma dma: tegra20-apbdma: move to generic device tree bindings spi: tegra114: move to generic dma DT binding spi: tegra20-slink: move to generic dma DT binding spi: tegra20-sflash: move to generic dma DT binding serial: tegra: move to generic dma DT binding ASoC: tegra: move to generic DMA DT binding ARM: dts: tegra: remove legacy nvidia,dma-request-selector properties dma: tegra20-apbdma: remove legacy nvidia,dma-request-selector support .../devicetree/bindings/dma/tegra20-apbdma.txt | 1 + .../bindings/serial/nvidia,tegra20-hsuart.txt | 8 ++-- .../bindings/sound/nvidia,tegra20-ac97.txt | 8 ++-- .../bindings/sound/nvidia,tegra20-i2s.txt | 8 ++-- .../bindings/sound/nvidia,tegra30-ahub.txt | 12 +++--- .../bindings/spi/nvidia,tegra114-spi.txt | 10 +++-- .../bindings/spi/nvidia,tegra20-sflash.txt | 10 +++-- .../bindings/spi/nvidia,tegra20-slink.txt | 10 +++-- arch/arm/boot/dts/tegra114.dtsi | 41 +++++++++++++------- arch/arm/boot/dts/tegra20.dtsi | 40 +++++++++++++------- arch/arm/boot/dts/tegra30.dtsi | 37 ++++++++++++------ drivers/dma/tegra20-apb-dma.c | 38 ++++++++++++++++++- drivers/spi/spi-tegra114.c | 16 +++----- drivers/spi/spi-tegra20-slink.c | 16 +++----- drivers/tty/serial/serial-tegra.c | 16 +------- sound/soc/tegra/tegra20_ac97.c | 17 +++------ sound/soc/tegra/tegra20_ac97.h | 2 + sound/soc/tegra/tegra20_i2s.c | 26 ++++--------- sound/soc/tegra/tegra20_i2s.h | 2 + sound/soc/tegra/tegra30_ahub.c | 25 +++++------- sound/soc/tegra/tegra30_ahub.h | 11 +++--- sound/soc/tegra/tegra30_i2s.c | 44 +++++++++++++++------- sound/soc/tegra/tegra30_i2s.h | 2 + sound/soc/tegra/tegra_pcm.c | 14 +++++++ sound/soc/tegra/tegra_pcm.h | 13 +++++++ 25 files changed, 260 insertions(+), 167 deletions(-) -- 1.8.1.5

3 12

[alsa-devel] [patch] ASoC: adau1701: type bug with ADAU1707_CLKDIV_UNSET
by Dan Carpenter 26 Jul '13

26 Jul '13

ADAU1707_CLKDIV_UNSET is always compared against an unsigned int and not an unsigned long. The current tests are always false. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 44d8a95..2c10252 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -91,7 +91,7 @@ #define ADAU1701_OSCIPOW_OPD 0x04 #define ADAU1701_DACSET_DACINIT 1 -#define ADAU1707_CLKDIV_UNSET (-1UL) +#define ADAU1707_CLKDIV_UNSET (-1U) #define ADAU1701_FIRMWARE "adau1701.bin"

3 2

[alsa-devel] [PATCH] ASoC: wm8994: Add clock bindings to the device tree
by Mark Brown 26 Jul '13

26 Jul '13

From: Mark Brown <broonie(a)linaro.org> Due to the variable availability of the clock API there is no code in the driver to use these at present, for the time being the machine drivers will need to handle requesting clocks. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Documentation/devicetree/bindings/sound/wm8994.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt index f2f3e80..e045e90 100644 --- a/Documentation/devicetree/bindings/sound/wm8994.txt +++ b/Documentation/devicetree/bindings/sound/wm8994.txt @@ -32,6 +32,10 @@ Optional properties: The second cell is the flags, encoded as the trigger masks from Documentation/devicetree/bindings/interrupts.txt + - clocks : A list of up to two phandle and clock specifier pairs + - clock-names : A list of clock names sorted in the same order as clocks. + Valid clock names are "MCLK1" and "MCLK2". + - wlf,gpio-cfg : A list of GPIO configuration register values. If absent, no configuration of these registers is performed. If any value is over 0xffff then the register will be left as default. If present 11 -- 1.8.3.2

3 5

[alsa-devel] [RFC PATCH 0/2] tea575x: Move from sound to media
by Ondrej Zary 26 Jul '13

26 Jul '13

Hello, TEA575x is neither a sound device nor an i2c device. Let's finally move it from sound/i2c/other to drivers/media/radio. Tested with snd-es1968, snd-fm801 and radio-sf16fmr2. I guess the Kconfig dependencies are not correct. -- Ondrej Zary

2 1

[alsa-devel] [PATCH 4/4] ASoc: kirkwood: add DT support
by Jean-Francois Moine 26 Jul '13

26 Jul '13

The kirkwood audio driver is used without DT in the Kirkwood machine. This patch adds a DT compatible definition for use in other Marvell machines as the Armada 88AP510 (Dove). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- .../devicetree/bindings/sound/kirkwood-i2s.txt | 29 ++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 34 +++++++++++++++------- 2 files changed, 52 insertions(+), 11 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/kirkwood-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt new file mode 100644 index 0000000..015a769 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) I2S controller + +Required properties: + +- compatible: "marvell,mvebu-i2s" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "external" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,mvebu-i2s"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3517784..fbe5150 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -12,7 +12,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/platform_device.h> #include <linux/io.h> #include <linux/slab.h> #include <linux/mbus.h> @@ -22,6 +21,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" #define DRV_NAME "kirkwood-i2s" @@ -497,16 +498,18 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); - return -EINVAL; - } - - priv->burst = data->burst; + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); + return -EINVAL; + } - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, "internal"); if (IS_ERR(priv->clk)) { - dev_err(&pdev->dev, "no clock\n"); + dev_err(&pdev->dev, "no internal clock\n"); return PTR_ERR(priv->clk); } @@ -514,7 +517,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (err < 0) return err; - priv->extclk = clk_get(&pdev->dev, "extclk"); + priv->extclk = clk_get(&pdev->dev, "external"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { clk_put(priv->extclk); @@ -531,7 +534,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -556,12 +559,21 @@ fail: return err; } +#ifdef CONFIG_OF +static struct of_device_id kirkwood_i2s_of_match[] = { + { .compatible = "marvell,mvebu-i2s" }, + { } +}; +MODULE_DEVICE_TABLE(of, kirkwood_i2s_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(kirkwood_i2s_of_match), }, }; -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 6

[alsa-devel] [PATCH 3/4] ASoc: kirkwood: merge kirkwood-i2c and kirkwood-dma
by Jean-Francois Moine 26 Jul '13

26 Jul '13

To avoid the declaration of a 'kirkwood-pcm-audio' device in the DT, this patch merges the kirkwood-i2c and kirkwood-dma drivers into one module. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- arch/arm/mach-kirkwood/common.c | 6 ------ sound/soc/kirkwood/Kconfig | 5 ----- sound/soc/kirkwood/Makefile | 4 +--- sound/soc/kirkwood/kirkwood-dma.c | 30 +----------------------------- sound/soc/kirkwood/kirkwood-i2s.c | 5 +++++ sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- sound/soc/kirkwood/kirkwood.h | 1 + 8 files changed, 10 insertions(+), 45 deletions(-) diff --git a/arch/arm/mach-kirkwood/common.c b/arch/arm/mach-kirkwood/common.c index e9238b5..fd02f1b 100644 --- a/arch/arm/mach-kirkwood/common.c +++ b/arch/arm/mach-kirkwood/common.c @@ -587,15 +587,9 @@ static struct platform_device kirkwood_i2s_device = { }, }; -static struct platform_device kirkwood_pcm_device = { - .name = "kirkwood-pcm-audio", - .id = -1, -}; - void __init kirkwood_audio_init(void) { platform_device_register(&kirkwood_i2s_device); - platform_device_register(&kirkwood_pcm_device); } /***************************************************************************** diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index c62d715..b6917a7 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -6,14 +6,10 @@ config SND_KIRKWOOD_SOC the Kirkwood I2S interface. You will also need to select the audio interfaces to support below. -config SND_KIRKWOOD_SOC_I2S - tristate - config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) depends on I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help Say Y if you want to add support for SoC audio on @@ -22,7 +18,6 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help Say Y if you want to add support for SoC audio on diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 3e62ae9..cf22ad3 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -1,8 +1,6 @@ -snd-soc-kirkwood-objs := kirkwood-dma.o -snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o +snd-soc-kirkwood-objs := kirkwood-i2s.o kirkwood-dma.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o snd-soc-openrd-objs := kirkwood-openrd.o snd-soc-t5325-objs := kirkwood-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cef5277..9f4c595 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -364,36 +364,8 @@ static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm) } } -static struct snd_soc_platform_driver kirkwood_soc_platform = { +struct snd_soc_platform_driver kirkwood_soc_platform = { .ops = &kirkwood_dma_ops, .pcm_new = kirkwood_dma_new, .pcm_free = kirkwood_dma_free_dma_buffers, }; - -static int kirkwood_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform); -} - -static int kirkwood_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver kirkwood_pcm_driver = { - .driver = { - .name = "kirkwood-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = kirkwood_soc_platform_probe, - .remove = kirkwood_soc_platform_remove, -}; - -module_platform_driver(kirkwood_pcm_driver); - -MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); -MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:kirkwood-pcm-audio"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d3e50af..3517784 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -459,6 +459,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) { struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_component(&pdev->dev); if (!IS_ERR(priv->extclk)) { @@ -544,6 +545,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto fail; } + err = snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform);; + if (err) { + dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); + goto fail; return 0; fail: diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index b979c71..0fa3082 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -55,7 +55,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { .name = "CS42L51", .stream_name = "CS42L51 HiFi", .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .platform_name = "kirkwood-i2s", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 1d0ed6f..f48ebda 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -71,7 +71,7 @@ static struct snd_soc_dai_link t5325_dai[] = { .name = "ALC5621", .stream_name = "ALC5621 HiFi", .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .platform_name = "kirkwood-i2s", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 4d92637..f3de119 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -133,4 +133,5 @@ struct kirkwood_dma_data { int burst; }; +extern struct snd_soc_platform_driver kirkwood_soc_platform; #endif -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH 2/4] ASoc: kirkwood: simplify probe error
by Jean-Francois Moine 26 Jul '13

26 Jul '13

The function kirkwood_i2s_dev_remove() may be used when probe fails. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 44 +++++++++++++++++++-------------------- 1 file changed, 21 insertions(+), 23 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..d3e50af 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -455,6 +455,21 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { .name = DRV_NAME, }; +static int kirkwood_i2s_dev_remove(struct platform_device *pdev) +{ + struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_component(&pdev->dev); + + if (!IS_ERR(priv->extclk)) { + clk_disable_unprepare(priv->extclk); + clk_put(priv->extclk); + } + clk_disable_unprepare(priv->clk); + + return 0; +} + static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; @@ -525,34 +540,17 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, soc_dai, 1); - if (!err) - return 0; - dev_err(&pdev->dev, "snd_soc_register_component failed\n"); - - if (!IS_ERR(priv->extclk)) { - clk_disable_unprepare(priv->extclk); - clk_put(priv->extclk); + if (err) { + dev_err(&pdev->dev, "snd_soc_register_component failed\n"); + goto fail; } - clk_disable_unprepare(priv->clk); + return 0; +fail: + kirkwood_i2s_dev_remove(pdev); return err; } -static int kirkwood_i2s_dev_remove(struct platform_device *pdev) -{ - struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); - - snd_soc_unregister_component(&pdev->dev); - - if (!IS_ERR(priv->extclk)) { - clk_disable_unprepare(priv->extclk); - clk_put(priv->extclk); - } - clk_disable_unprepare(priv->clk); - - return 0; -} - static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH 1/4] ASoc: kirkwood: change the pcm/dma stream data pointer
by Jean-Francois Moine 26 Jul '13

26 Jul '13

In the kirkwood pcm/dma driver, the private stream related data pointer was hold in the platform drvdata. As this pointer may be used by other parts of the audio subsystem, this patch uses the substream runtime private data pointer instead. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-dma.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index a9f1453..cef5277 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -127,10 +127,9 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) int err; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_platform *platform = soc_runtime->platform; struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; struct kirkwood_dma_data *priv; - struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform); + struct kirkwood_dma_priv *prdata = runtime->private_data; const struct mbus_dram_target_info *dram; unsigned long addr; @@ -171,7 +170,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) return -EBUSY; } - snd_soc_platform_set_drvdata(platform, prdata); + runtime->private_data = prdata; /* * Enable Error interrupts. We're only ack'ing them but @@ -199,8 +198,7 @@ static int kirkwood_dma_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; - struct snd_soc_platform *platform = soc_runtime->platform; - struct kirkwood_dma_priv *prdata = snd_soc_platform_get_drvdata(platform); + struct kirkwood_dma_priv *prdata = substream->runtime->private_data; struct kirkwood_dma_data *priv; priv = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -217,7 +215,7 @@ static int kirkwood_dma_close(struct snd_pcm_substream *substream) writel(0, priv->io + KIRKWOOD_ERR_MASK); free_irq(priv->irq, prdata); kfree(prdata); - snd_soc_platform_set_drvdata(platform, NULL); + substream->runtime->private_data = NULL; } return 0; -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH 1/3] ALSA: hda - Clearing jackpoll_interval avoid pending work
by Wang Xingchao 26 Jul '13

26 Jul '13

Clearing jackpoll_interval before calling cancel_delayed_work_sync(), otherwise the work will be triggered again and cause impact in hda_jackpoll_work(). The next patch will poll jack once even with jackpoll_interval=0. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e2481ba..0bc20ef 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec) return; if (spec->hp_work_active) { snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1); + codec->jackpoll_interval = 0; cancel_delayed_work_sync(&codec->jackpoll_work); spec->hp_work_active = false; - codec->jackpoll_interval = 0; } } -- 1.8.3.2

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[alsa-devel] Decibel Levels for Storm Medusa 5.1 USB Headset
by Alexander Blinne 26 Jul '13

26 Jul '13

Hello everyone, I hope this Information helps. I have the Storm Medusa 5.1 USB Headset which reports wrong decibel levels to pulseaudio. My Alsa-Info are located at http://www.alsa-project.org/db/?f=a882f3db6769c5dfce83c266946eff52d1d55349 The Headset is card2 in the above dump. My decibel levels-fix [DecibelFix Speaker] db-values = 1:-30 37:3 works fine. Level 0 is mute, hence -infinity dB. Greetings

3 3

[alsa-devel] [PATCH] ASoC: fsl: Set sdma peripheral type directly
by Nicolin Chen 25 Jul '13

25 Jul '13

Let CPU DAI drivers set SDMA periperal type directly to support more dma types(SPDIF, ESAI) other than only two for SSI. This will easily allow some non-SSI drivers to use the imx-pcm-dma as well. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- @Timur Compile-passed with mpc85xx_defconfig by using gcc-4.6.3-nolibc_powerpc-linux. --- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/imx-pcm.h | 7 ++----- sound/soc/fsl/imx-ssi.c | 4 ++-- 3 files changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 11469fe..4d78df7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -775,9 +775,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) "fsl,spba-bus"); imx_pcm_dma_params_init_data(&ssi_private->filter_data_tx, - dma_events[0], shared); + dma_events[0], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx, - dma_events[1], shared); + dma_events[1], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI); } /* Initialize the the device_attribute structure */ diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index fd56cad..9136625 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -22,14 +22,11 @@ static inline void imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, - int dma, bool shared) + int dma, enum sdma_peripheral_type peripheral_type) { dma_data->dma_request = dma; dma_data->priority = DMA_PRIO_HIGH; - if (shared) - dma_data->peripheral_type = IMX_DMATYPE_SSI_SP; - else - dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->peripheral_type = peripheral_type; } struct imx_pcm_fiq_params { diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f029e27..f58bcd8 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -571,13 +571,13 @@ static int imx_ssi_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); if (res) { imx_pcm_dma_params_init_data(&ssi->filter_data_tx, res->start, - false); + IMX_DMATYPE_SSI); } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); if (res) { imx_pcm_dma_params_init_data(&ssi->filter_data_rx, res->start, - false); + IMX_DMATYPE_SSI); } platform_set_drvdata(pdev, ssi); -- 1.7.1

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[alsa-devel] [PATCH 4/4 v2] ASoc: kirkwood: add DT support
by Jean-Francois Moine 25 Jul '13

25 Jul '13

The kirkwood audio driver is used without DT in the Kirkwood machine. This patch adds a DT compatible definition for use in other Marvell machines as the Armada 88AP510 (Dove). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2 fix compilation error --- Documentation/devicetree/bindings/sound/kirkwood-i2s.txt | 29 ++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 35 +++++++++++++++------- 2 files changed, 53 insertions(+), 11 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/kirkwood-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt new file mode 100644 index 0000000..015a769 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/kirkwood-i2s.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) I2S controller + +Required properties: + +- compatible: "marvell,mvebu-i2s" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "external" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,mvebu-i2s"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/home/jef/kernel/sound/soc/kirkwood/kirkwood-i2s.c.new index a1037e2..26c2f74 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/home/jef/kernel/sound/soc/kirkwood/kirkwood-i2s.c.new @@ -12,7 +12,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/platform_device.h> #include <linux/io.h> #include <linux/slab.h> #include <linux/mbus.h> @@ -22,6 +21,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" #define DRV_NAME "kirkwood-i2s" @@ -477,6 +478,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np = pdev->dev.of_node; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -497,16 +499,18 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); - return -EINVAL; - } - - priv->burst = data->burst; + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); + return -EINVAL; + } - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, "internal"); if (IS_ERR(priv->clk)) { - dev_err(&pdev->dev, "no clock\n"); + dev_err(&pdev->dev, "no internal clock\n"); return PTR_ERR(priv->clk); } @@ -514,7 +518,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (err < 0) return err; - priv->extclk = clk_get(&pdev->dev, "extclk"); + priv->extclk = clk_get(&pdev->dev, "external"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { clk_put(priv->extclk); @@ -531,7 +535,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -557,12 +561,21 @@ fail: return err; } +#ifdef CONFIG_OF +static struct of_device_id kirkwood_i2s_of_match[] = { + { .compatible = "marvell,mvebu-i2s" }, + { } +}; +MODULE_DEVICE_TABLE(of, kirkwood_i2s_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(kirkwood_i2s_of_match), }, }; -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH 3/4 v2] ASoc: kirkwood: merge kirkwood-i2c and kirkwood-dma
by Jean-Francois Moine 25 Jul '13

25 Jul '13

To avoid the declaration of a 'kirkwood-pcm-audio' device in the DT, this patch merges the kirkwood-i2c and kirkwood-dma drivers into one module. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2 - fix compilation error --- arch/arm/mach-kirkwood/common.c | 6 ------ sound/soc/kirkwood/Kconfig | 5 ----- sound/soc/kirkwood/Makefile | 4 +--- sound/soc/kirkwood/kirkwood-dma.c | 30 +----------------------------- sound/soc/kirkwood/kirkwood-i2s.c | 6 ++++++ sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- sound/soc/kirkwood/kirkwood.h | 1 + 8 files changed, 11 insertions(+), 45 deletions(-) diff --git a/arch/arm/mach-kirkwood/common.c b/arch/arm/mach-kirkwood/common.c index e9238b5..fd02f1b 100644 --- a/arch/arm/mach-kirkwood/common.c +++ b/arch/arm/mach-kirkwood/common.c @@ -587,15 +587,9 @@ static struct platform_device kirkwood_i2s_device = { }, }; -static struct platform_device kirkwood_pcm_device = { - .name = "kirkwood-pcm-audio", - .id = -1, -}; - void __init kirkwood_audio_init(void) { platform_device_register(&kirkwood_i2s_device); - platform_device_register(&kirkwood_pcm_device); } /***************************************************************************** diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index c62d715..b6917a7 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -6,14 +6,10 @@ config SND_KIRKWOOD_SOC the Kirkwood I2S interface. You will also need to select the audio interfaces to support below. -config SND_KIRKWOOD_SOC_I2S - tristate - config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) depends on I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help Say Y if you want to add support for SoC audio on @@ -22,7 +18,6 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C - select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help Say Y if you want to add support for SoC audio on diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 3e62ae9..cf22ad3 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -1,8 +1,6 @@ -snd-soc-kirkwood-objs := kirkwood-dma.o -snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o +snd-soc-kirkwood-objs := kirkwood-i2s.o kirkwood-dma.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o snd-soc-openrd-objs := kirkwood-openrd.o snd-soc-t5325-objs := kirkwood-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cef5277..9f4c595 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -364,36 +364,8 @@ static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm) } } -static struct snd_soc_platform_driver kirkwood_soc_platform = { +struct snd_soc_platform_driver kirkwood_soc_platform = { .ops = &kirkwood_dma_ops, .pcm_new = kirkwood_dma_new, .pcm_free = kirkwood_dma_free_dma_buffers, }; - -static int kirkwood_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform); -} - -static int kirkwood_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver kirkwood_pcm_driver = { - .driver = { - .name = "kirkwood-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = kirkwood_soc_platform_probe, - .remove = kirkwood_soc_platform_remove, -}; - -module_platform_driver(kirkwood_pcm_driver); - -MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); -MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:kirkwood-pcm-audio"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d3e50af..3517784 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -459,6 +459,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) { struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_component(&pdev->dev); if (!IS_ERR(priv->extclk)) { @@ -544,6 +545,11 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto fail; } + err = snd_soc_register_platform(&pdev->dev, &kirkwood_soc_platform);; + if (err) { + dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); + goto fail; + } return 0; fail: diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index b979c71..0fa3082 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -55,7 +55,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { .name = "CS42L51", .stream_name = "CS42L51 HiFi", .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .platform_name = "kirkwood-i2s", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 1d0ed6f..f48ebda 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -71,7 +71,7 @@ static struct snd_soc_dai_link t5325_dai[] = { .name = "ALC5621", .stream_name = "ALC5621 HiFi", .cpu_dai_name = "kirkwood-i2s", - .platform_name = "kirkwood-pcm-audio", + .platform_name = "kirkwood-i2s", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 4d92637..f3de119 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -133,4 +133,5 @@ struct kirkwood_dma_data { int burst; }; +extern struct snd_soc_platform_driver kirkwood_soc_platform; #endif -- 1.8.3.2 -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda - jack poll once if jackpoll_interval==0
by Wang Xingchao 25 Jul '13

25 Jul '13

jackpoll_interval used to poll jack event periodically, if it's 0, give the caller one chance to probe jack status. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8a005f0..74c11bc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1216,11 +1216,12 @@ static void hda_jackpoll_work(struct work_struct *work) { struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - if (!codec->jackpoll_interval) - return; snd_hda_jack_set_dirty_all(codec); snd_hda_jack_poll_all(codec); + if (!codec->jackpoll_interval) + return; + queue_delayed_work(codec->bus->workq, &codec->jackpoll_work, codec->jackpoll_interval); } -- 1.8.3.2

4 16

[alsa-devel] [PATCH 0/6] ASoC: add CSR SiRFSoC sound drivers
by Barry Song 25 Jul '13

25 Jul '13

This patchset adds CSR SiRFSoC sound drivers including: 1. the platform DMA driver which will be shard by all DAI 2. the I2S CPU DAI driver 3. the USP-based PCM CPU DAI driver 4. CPU DAI and Codec driver for internal chip codec 5. the mach driver for EVB board using internal codec Barry Song (1): arm: prima2: defconfig: enable sound components Rongjun Ying (5): ASoC: sirf: add sirf platform driver which provides DMA ASoC: sirf: add I2S CPU DAI driver ASoC: usp-pcm: add CPU DAI driver for PCM simulated from USP ASoC: sirf-soc-inner: add drivers for both CPU and Codec DAIs ASoC: sirf-inner: add mach driver for SiRFSoC internal codec arch/arm/configs/prima2_defconfig | 10 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sirf/Kconfig | 18 ++ sound/soc/sirf/Makefile | 11 + sound/soc/sirf/sirf-audio.h | 266 ++++++++++++++++ sound/soc/sirf/sirf-i2s.c | 411 ++++++++++++++++++++++++ sound/soc/sirf/sirf-inner.c | 267 ++++++++++++++++ sound/soc/sirf/sirf-pcm.c | 221 +++++++++++++ sound/soc/sirf/sirf-pcm.h | 17 + sound/soc/sirf/sirf-soc-inner.c | 653 ++++++++++++++++++++++++++++++++++++++ sound/soc/sirf/sirf-usp.c | 481 ++++++++++++++++++++++++++++ sound/soc/sirf/sirf-usp.h | 276 ++++++++++++++++ 13 files changed, 2630 insertions(+), 3 deletions(-) create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio.h create mode 100644 sound/soc/sirf/sirf-i2s.c create mode 100644 sound/soc/sirf/sirf-inner.c create mode 100644 sound/soc/sirf/sirf-pcm.c create mode 100644 sound/soc/sirf/sirf-pcm.h create mode 100644 sound/soc/sirf/sirf-soc-inner.c create mode 100644 sound/soc/sirf/sirf-usp.c create mode 100644 sound/soc/sirf/sirf-usp.h -- 1.8.2.3

3 18

[alsa-devel] [PATCH] ASoC: rt5640: add headset detection function
by bardliao＠realtek.com 25 Jul '13

25 Jul '13

From: Bard Liao <bardliao(a)realtek.com> Add headset detection function to rt5640 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 61 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 3 +++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index ce585e3..dc0d063 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1911,6 +1911,67 @@ static int rt5640_resume(struct snd_soc_codec *codec) #define rt5640_resume NULL #endif +/** + * rt5640_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +int rt5640_headset_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int jack_insert) +{ + int jack_type = 0; + int sclk_src = 0; + int reg63, reg64; + + if (jack_insert) { + reg63 = snd_soc_read(codec, RT5640_PWR_ANLG1); + reg64 = snd_soc_read(codec, RT5640_PWR_ANLG2); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + snd_soc_write(codec, RT5640_PWR_ANLG1, 0xa814); + sclk_src = snd_soc_read(codec, RT5640_GLB_CLK) & + RT5640_SCLK_SRC_MASK; + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, + 0x2 << RT5640_SCLK_SRC_SFT); + } + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LDO2, RT5640_PWR_LDO2); + snd_soc_update_bits(codec, RT5640_PWR_ANLG2, + RT5640_PWR_MB1, RT5640_PWR_MB1); + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK | RT5640_MIC1_OVTH_MASK | + RT5640_PWR_CLK25M_MASK | RT5640_PWR_MB_MASK, + RT5640_MIC1_OVCD_EN | RT5640_MIC1_OVTH_600UA | + RT5640_PWR_MB_PU | RT5640_PWR_CLK25M_PU); + snd_soc_update_bits(codec, RT5640_DUMMY1, 0x1, 0x1); + msleep(100); + if (snd_soc_read(codec, RT5640_IRQ_CTRL2) & 0x8) + jack_type = SND_JACK_HEADPHONE; + else + jack_type = SND_JACK_HEADSET; + snd_soc_update_bits(codec, RT5640_IRQ_CTRL2, + RT5640_MB1_OC_CLR, 0); + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_SCLK_SRC_MASK, sclk_src); + snd_soc_write(codec, RT5640_PWR_ANLG1, reg63); + snd_soc_write(codec, RT5640_PWR_ANLG2, reg64); + snd_soc_jack_report(jack, jack_type, SND_JACK_HEADSET); + } else { + snd_soc_update_bits(codec, RT5640_MICBIAS, + RT5640_MIC1_OVCD_MASK, + RT5640_MIC1_OVCD_DIS); + snd_soc_jack_report(jack, 0, jack->jack->type); + } + + return jack_type; +} +EXPORT_SYMBOL_GPL(rt5640_headset_detect); + #define RT5640_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5640_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..263615b 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2089,4 +2089,7 @@ struct rt5640_priv { int dmic_en; }; +int rt5640_headset_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int jack_insert); + #endif -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] [PATCH 0/4] ASoc: kirkwood: extend the kirkwood audio subsystem for DT usage
by Jean-Francois Moine 25 Jul '13

25 Jul '13

This patch series adds DT support to the kirkwood audio subsystem. It cancels the previous patch requests: - ARM: kirkwood: enable S/PDIF - ARM: kirkwood: extend the kirkwood i2s driver for DT usage - ARM: kirkwood: extend the kirkwood pcm/dma driver for DT usage It also contains the merge of kirkwood-i2c and kirkwood-dma into one module, and, the removal of the kirkwood-pcm-audio device. This involves changes in the openrd and t5325 drivers, as in the common Kirkwood machine initialization. These last changes have not been tested. -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] PCM extra attributes
by Vinod Koul 25 Jul '13

25 Jul '13

Hey Takashi, We were looking to find the substream index in the userspace for the device opened (in case we support multiple substreams and mix them). Since we also wanted to have some specific alsa controls for each substream the problem of not knowing substream in usermode make stuff interesting! I was pointed to an old RFC of yours [1] and it fits my needs. I can get the substream index using this and program the alsa controls. Now reading the archive looks like it was note merged. Would be okay to do so for above case. I can resurrect the patchset if you would like... Thanks ~Vinod [1]: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018651.html --

3 3

[alsa-devel] [PATCH 0/4 V7] Power-well API implementation for Haswell
by Wang Xingchao 25 Jul '13

25 Jul '13

Hi all, This is V7 and here're some changes notes: change from V6-->V7: - rename variable - use HAS_POWER_WELL instead of IS_HASWELL - put structure inside drm_i915_private - use WARN_ON for global pointer check change from V5-->V6: - Remove duplication code in new introduced probe work - move duplication code in azx_probe_continue - remove unused #ifdef - replace request_module with symbol_request - replace spin_lock_irq with spin_lock_irqsave in gfx side - other typo fixes (review by Takashi) change from V4-->V5: - fix reference count bug - new patch on general runtime pm support for audio pci device - new patch to avoid request_module() deadlock change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Takashi Iwai (1): ALSA: hda - Move azx_first_init() into azx_probe_continue() Wang Xingchao (3): ALSA: hda - Fix runtime PM check ALSA: hda - Add power-welll support for haswell HDA i915/drm: Add private api for power well usage drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 ++++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 81 ++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 ++++++++++++++++ sound/pci/hda/Kconfig | 10 +++++ sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_i915.c | 75 ++++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++++++ sound/pci/hda/hda_intel.c | 87 ++++++++++++++++++++++++++++++-------- 10 files changed, 324 insertions(+), 24 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

7 40

[alsa-devel] unsubscribe my ID
by Reena Chauhan Tomar 25 Jul '13

25 Jul '13

Kindly unsubscribe this ID <reena.nith(a)gmail.com> from your mailing list. -- Reena Tomar Scientist-C Application & Baseband Platform, SBSG ,DEAL,Raipur Road Dehradun-248001

1 0

[alsa-devel] usb-audio: capture/playback break (implicit feedback related?)
by Eldad Zack 24 Jul '13

24 Jul '13

Hi, I'm still trying to figure out the issue with the capture and playback breaking with implicit feedback. I've disabled setting the sync ep for my device and did the following: start capture start playback stop playback Now, after a while, capture stops. I sniffed the usb traffic, and saw this: When I stop the traffic, the driver sets the alternate setting to 0 on the playback interface (2 for my device). So far so good. The first iso inbound packet has data in it, but starting from the second packet, it's all zero (in length). After a while, the device stops sending packets completely. The driver still tries, and evetually sets the alternate setting to 0 on the capture interface (3 for my device). Again, there is no implicit feedback involved here. I also ruled out playback interface being put into altsetting 1 (without us knowing it) when the capture was put into altsetting 1 by doing a GET_INTERFACE in set_format(). I am not sure if this an M-Audio issue, just my specific device (Fast Track C400) or maybe it appears in other devices. Anyone has an idea why this happens? I'd appreciate if anyone can test this on other devices. Clemens, since you recently added support for Roland devices that use implicit feedback, would you please test it and see if you get the same results, if you have time? If it helps, I can send the capture. Cheers, Eldad P.S. GET_INTERFACE adapted from drivers/usb/misc/usbtest.c: static int snd_usb_get_interface(struct usb_device *dev, struct usb_interface *iface) { int ret; char buf; ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), USB_REQ_GET_INTERFACE, USB_DIR_IN|USB_RECIP_INTERFACE, 0, iface->altsetting[0].desc.bInterfaceNumber, &buf, sizeof(buf), USB_CTRL_GET_TIMEOUT); if (ret == sizeof(buf)) return buf; return -ERANGE; }

1 0

[alsa-devel] [PATCH] fix error returned by compress_wait()
by Vinod Koul 24 Jul '13

24 Jul '13

From: Eric Laurent <elaurent(a)google.com> oops() expects positive error codes. Signed-off-by: Eric Laurent <elaurent(a)google.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/compress.c b/compress.c index 734fbbf..0e71c28 100644 --- a/compress.c +++ b/compress.c @@ -623,7 +623,7 @@ int compress_wait(struct compress *compress, int timeout_ms) return 0; } if (fds.revents & POLLERR) { - return oops(compress, -EIO, "poll returned error!"); + return oops(compress, EIO, "poll returned error!"); } return ret; } -- 1.7.0.4

1 0

[alsa-devel] [PATCH 00/27] devm cleanup, part #1, take #3
by Wolfram Sang 24 Jul '13

24 Jul '13

Here is another bit of cleaning up the devm usage. It is again removing the resource check with devm_ioremap_resource, because a) new drivers came in and b) coccinelle had a bug and missed to find a couple of occasions. Unlike last time, I think it is better if these patches go in via the subsystem trees to reduce merge conflicts. And there is one driver which I fixed manually because the original code needed some bigger update. All is based on v3.11-rc2 and the branch can be found at: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git devm_no_resource_check Other things which happened: I wanted to get rid of devm_request_and_ioremap, luckily other people are already working on it. Hooray! I already sent a patch series picking another low hanging fruit, i.e. drivers can skip devm_pinctrl handling if they are using only the default pin setup. And not devm related, there is still my proposal to rename INIT_COMPLETION to reinit_completion, that probably needs some more persistence... Yay, so much to clean up \o/ Regards, Wolfram Wolfram Sang (27): arch/mips/lantiq/xway: don't check resource with devm_ioremap_resource drivers/amba: don't check resource with devm_ioremap_resource drivers/cpuidle: don't check resource with devm_ioremap_resource drivers/dma: don't check resource with devm_ioremap_resource drivers/gpu/host1x/drm: don't check resource with devm_ioremap_resource drivers/i2c/busses: don't check resource with devm_ioremap_resource drivers/input/serio: don't check resource with devm_ioremap_resource drivers/iommu: don't check resource with devm_ioremap_resource drivers/media/platform: don't check resource with devm_ioremap_resource drivers/memory: don't check resource with devm_ioremap_resource drivers/mtd/nand: don't check resource with devm_ioremap_resource drivers/net/ethernet/stmicro/stmmac: don't check resource with devm_ioremap_resource drivers/pci/host: don't check resource with devm_ioremap_resource drivers/pinctrl: don't check resource with devm_ioremap_resource drivers/pwm: don't check resource with devm_ioremap_resource drivers/scsi/ufs: don't check resource with devm_ioremap_resource drivers/spi: don't check resource with devm_ioremap_resource drivers/staging/imx-drm: don't check resource with devm_ioremap_resource drivers/usb/phy: don't check resource with devm_ioremap_resource drivers/watchdog: don't check resource with devm_ioremap_resource sound/soc/au1x: don't check resource with devm_ioremap_resource sound/soc/cirrus: don't check resource with devm_ioremap_resource sound/soc/nuc900: don't check resource with devm_ioremap_resource sound/soc/pxa: don't check resource with devm_ioremap_resource sound/soc/tegra: don't check resource with devm_ioremap_resource sound/soc/txx9: don't check resource with devm_ioremap_resource thermal: ti-bandgap: cleanup resource allocation arch/mips/lantiq/xway/dma.c | 4 ---- drivers/amba/tegra-ahb.c | 2 -- drivers/cpuidle/cpuidle-kirkwood.c | 3 --- drivers/dma/mmp_pdma.c | 3 --- drivers/dma/mmp_tdma.c | 3 --- drivers/gpu/host1x/drm/hdmi.c | 3 --- drivers/i2c/busses/i2c-stu300.c | 3 --- drivers/input/serio/olpc_apsp.c | 3 --- drivers/iommu/tegra-smmu.c | 2 -- drivers/media/platform/coda.c | 5 ----- drivers/memory/tegra20-mc.c | 2 -- drivers/memory/tegra30-mc.c | 2 -- drivers/mtd/nand/mxc_nand.c | 5 ----- .../net/ethernet/stmicro/stmmac/stmmac_platform.c | 3 --- drivers/pci/host/pcie-designware.c | 12 ------------ drivers/pinctrl/pinctrl-imx.c | 3 --- drivers/pinctrl/pinctrl-rockchip.c | 5 ----- drivers/pinctrl/pinctrl-u300.c | 3 --- drivers/pwm/pwm-lpc32xx.c | 3 --- drivers/pwm/pwm-renesas-tpu.c | 5 ----- drivers/scsi/ufs/ufshcd-pltfrm.c | 6 ------ drivers/spi/spi-bcm2835.c | 6 ------ drivers/staging/imx-drm/imx-tve.c | 5 ----- drivers/thermal/ti-soc-thermal/ti-bandgap.c | 20 ++++---------------- drivers/usb/phy/phy-rcar-usb.c | 5 ----- drivers/watchdog/nuc900_wdt.c | 5 ----- drivers/watchdog/ts72xx_wdt.c | 10 ---------- sound/soc/au1x/psc-ac97.c | 3 --- sound/soc/cirrus/ep93xx-ac97.c | 3 --- sound/soc/cirrus/ep93xx-i2s.c | 3 --- sound/soc/nuc900/nuc900-ac97.c | 3 --- sound/soc/pxa/mmp-sspa.c | 3 --- sound/soc/tegra/tegra20_ac97.c | 7 ------- sound/soc/txx9/txx9aclc-ac97.c | 3 --- 34 files changed, 4 insertions(+), 152 deletions(-) -- 1.7.10.4

2 12

[alsa-devel] [PATCH 1/5] ASoC: dapm: Fix return value of snd_soc_dapm_put_{volsw, enum_virt}()
by Lars-Peter Clausen 24 Jul '13

24 Jul '13

The ALSA core expect the put callback of a control to return 1 if the value of the control changed and 0 if it did not. Both snd_soc_dapm_put_volsw() and snd_soc_dapm_put_enum_virt() currently always returns 0. For both functions we already have a 'change' variable which either contains 1 or 0 depending on whether the value has changed or not, so just return that. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index deb1b3d..4097a55 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2703,7 +2703,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, } mutex_unlock(&card->dapm_mutex); - return 0; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2831,7 +2831,6 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; - int ret = 0; int wi; if (ucontrol->value.enumerated.item[0] >= e->max) @@ -2851,7 +2850,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, } mutex_unlock(&card->dapm_mutex); - return ret; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); -- 1.8.0

2 5

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix EAPD GPIO control for Sigmatel codecs
by Takashi Iwai 24 Jul '13

24 Jul '13

The EAPD GPIO is dynamically turned on/off for some machines with Sigmatel codecs, but this didn't work as expected, and it resulted in spontaneous lost of speaker outputs per HP plugging or power-saving. This patch fixes the bug by simply including spec->eapd_mask into spec->gpio_mask and spec->gpio_data bits. Reported-and-tested-by: Eric Shattow <lucent(a)gmail.com> Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e2f8359..766e567 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -417,9 +417,11 @@ static void stac_update_outputs(struct hda_codec *codec) val &= ~spec->eapd_mask; else val |= spec->eapd_mask; - if (spec->gpio_data != val) + if (spec->gpio_data != val) { + spec->gpio_data = val; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, val); + } } } @@ -3612,20 +3614,18 @@ static int stac_parse_auto_config(struct hda_codec *codec) static int stac_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int gpio; int i; /* override some hints */ stac_store_hints(codec); /* set up GPIO */ - gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. * otherwise, unsol event will turn it on/off dynamically */ if (!spec->eapd_switch) - gpio |= spec->eapd_mask; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); + spec->gpio_data |= spec->eapd_mask; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); snd_hda_gen_init(codec); @@ -3915,6 +3915,7 @@ static void stac_setup_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + spec->gpio_mask |= spec->eapd_mask; if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; -- 1.8.3.1

2 5

[alsa-devel] [PATCHv2] ALSA: usb-audio: hiface: return correct XRUN indication
by Eldad Zack 24 Jul '13

24 Jul '13

Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of hiface, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Cc: Antonio Ospite <ospite(a)studenti.unina.it> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- * Corrected commit message sound/usb/hiface/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 6430ed2..c21a3df 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -503,7 +503,7 @@ static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) snd_pcm_uframes_t dma_offset; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); dma_offset = sub->dma_off; -- 1.8.1.5

2 1

[alsa-devel] [GIT PULL] ASoC updates for v3.11
by Mark Brown 24 Jul '13

24 Jul '13

The following changes since commit 3b2f64d00c46e1e4e9bd0bb9bb12619adac27a4b: Linux 3.11-rc2 (2013-07-21 12:05:29 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc2 for you to fetch changes up to c5ae4dca11523dff15cfb1763b64a71b805976aa: Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus (2013-07-23 19:17:11 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.11 A few small updates again, the sgtl5000 one fixes some newly triggered issues due to some probe ordering changes which were introduced in the last merge window. ---------------------------------------------------------------- Boris BREZILLON (1): ASoC: atmel-ssc: remove clk_disable_unprepare call from critical section Chih-Chung Chang (1): ASoC: max98088 - fix element type of the register cache. Dan Carpenter (1): ASoC: core: double free in snd_soc_add_platform() H Hartley Sweeten (1): ASoC: ep93xx: fix build of ep93xx-ac97.c Mark Brown (6): Merge remote-tracking branch 'asoc/fix/atmel' into asoc-linus Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/ep93xx' into asoc-linus Merge remote-tracking branch 'asoc/fix/max98088' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus Richard Zhao (1): ASoC: tegra: correct playback_dma_data setup Shawn Guo (1): ASoC: sgtl5000: defer the probe if clock is not found drivers/misc/atmel-ssc.c | 11 ++++++++--- sound/soc/cirrus/ep93xx-ac97.c | 4 ++-- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/sgtl5000.c | 3 +++ sound/soc/soc-core.c | 4 +--- sound/soc/tegra/tegra20_ac97.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 4 ++-- 7 files changed, 20 insertions(+), 14 deletions(-)

2 1

[alsa-devel] [PATCH 1/4] ASOC: tegra: move AC97 clock defines to the controller node
by Lucas Stach 24 Jul '13

24 Jul '13

Different from other Tegra sound controllers drivers, the AC97 controller driver uses the tegra asoc utils directly to request the needed clocks, as they are needed at AC97 init time. Move the DT clock defines to the right place. Signed-off-by: Lucas Stach <dev(a)lynxeye.de> --- CC: <alsa-devel(a)alsa-project.org> CC: <broonie(a)kernel.org> --- arch/arm/boot/dts/tegra20-colibri-512.dtsi | 5 ----- arch/arm/boot/dts/tegra20.dtsi | 6 +++++- sound/soc/tegra/tegra20_ac97.c | 2 +- 3 files changed, 6 insertions(+), 7 deletions(-) diff --git a/arch/arm/boot/dts/tegra20-colibri-512.dtsi b/arch/arm/boot/dts/tegra20-colibri-512.dtsi index 2fcb3f2..fbb52e0 100644 --- a/arch/arm/boot/dts/tegra20-colibri-512.dtsi +++ b/arch/arm/boot/dts/tegra20-colibri-512.dtsi @@ -491,11 +491,6 @@ "Mic", "MIC1"; nvidia,ac97-controller = <&ac97>; - - clocks = <&tegra_car TEGRA20_CLK_PLL_A>, - <&tegra_car TEGRA20_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA20_CLK_CDEV1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; }; regulators { diff --git a/arch/arm/boot/dts/tegra20.dtsi b/arch/arm/boot/dts/tegra20.dtsi index 9653fd8..ad13f57 100644 --- a/arch/arm/boot/dts/tegra20.dtsi +++ b/arch/arm/boot/dts/tegra20.dtsi @@ -223,7 +223,11 @@ reg = <0x70002000 0x200>; interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; nvidia,dma-request-selector = <&apbdma 12>; - clocks = <&tegra_car TEGRA20_CLK_AC97>; + clocks = <&tegra_car TEGRA20_CLK_PLL_A>, + <&tegra_car TEGRA20_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA20_CLK_CDEV1>, + <&tegra_car TEGRA20_CLK_AC97>; + clock-names = "pll_a", "pll_a_out0", "mclk", "ac97"; status = "disabled"; }; diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 6c48662..6bbffd1 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -326,7 +326,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, ac97); - ac97->clk_ac97 = devm_clk_get(&pdev->dev, NULL); + ac97->clk_ac97 = devm_clk_get(&pdev->dev, "ac97"); if (IS_ERR(ac97->clk_ac97)) { dev_err(&pdev->dev, "Can't retrieve ac97 clock\n"); ret = PTR_ERR(ac97->clk_ac97); -- 1.8.3.1

3 7

Re: [alsa-devel] [PATCH] ALSA: SoC: Added jack detection kcontrol support
by Mark Brown 24 Jul '13

24 Jul '13

On Fri, Jul 19, 2013 at 05:59:22PM -0700, Felipe F. Tonello wrote: > From: "Felipe F. Tonello" <ftonello(a)cercacor.com> > > It adds soc jack support for alsa kcontrol. > > This support is necessary since the new kcontrol is used by user-space > daemons, such as PulseAudio(>=2.0), to do jack detection. > > Signed-off-by: Felipe F. Tonello <ftonello(a)cercacor.com> > --- > include/sound/soc.h | 1 + > sound/soc/Kconfig | 1 + > sound/soc/soc-jack.c | 22 ++++++++++++++++++++++ > 3 files changed, 24 insertions(+) No, this makes no sense at all as an ASoC specific feature - it should be a standard feature of ALSA jacks. The whole point of having ALSA core infrastructure for jacks is to avoid having to go round doing make work stuff like this. The fact that the kcontrols were introduced as a HDA-specific feature at all is of course a problem here...

3 5

[alsa-devel] question about alsa-lib commit 8c9e4114 (Add struct timeval and timespec definition when _POSIX_C_SOURCE is not defined)
by John Spencer 24 Jul '13

24 Jul '13

the commit http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=8c9e4114a23aad96… was added in 2006 with a FIXME hint that it could cause a build error. this is exactly what happens when using the header with musl libc to build software depending on alsa-lib (for example "sweep"). may i ask what for this hack was added ? i suspect it's for supporting some ancient version of glibc that didnt define those structs. in that case something like #if defined(__GLIBC__) && (__GLIBC__ < 2) would probably make more sense. i could even imagine that the target for the hack is no longer supported anyway, so i think it could simply be removed. the problematic part is that it's the job of the *application* to request the POSIX namespace, not the job of the libc. since sweep's build environment does not set _POSIX_SOURCE, the code gets activated and causes a conflict. --JS p.s. thanks for applying the signal.h patch!

1 0

[alsa-devel] [PATCH] ASoC: Remove unused dapm_get_snd_card() and dapm_get_soc_card()
by Lars-Peter Clausen 24 Jul '13

24 Jul '13

These two functions were added two years ago in commit 4805608 ("ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.") but have remained unused so far. Considering that the dapm context actually has a direct pointer to the card the functions also seem to be unnecessary. E.g. the expressions 'dapm_get_soc_card(dapm)' and 'dapm->card' yield the same result. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 30 ------------------------------ 1 file changed, 30 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 346ac91..deb1b3d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -174,36 +174,6 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } -/* get snd_card from DAPM context */ -static inline struct snd_card *dapm_get_snd_card( - struct snd_soc_dapm_context *dapm) -{ - if (dapm->codec) - return dapm->codec->card->snd_card; - else if (dapm->platform) - return dapm->platform->card->snd_card; - else - BUG(); - - /* unreachable */ - return NULL; -} - -/* get soc_card from DAPM context */ -static inline struct snd_soc_card *dapm_get_soc_card( - struct snd_soc_dapm_context *dapm) -{ - if (dapm->codec) - return dapm->codec->card; - else if (dapm->platform) - return dapm->platform->card; - else - BUG(); - - /* unreachable */ - return NULL; -} - static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; -- 1.8.0

4 8

[alsa-devel] [PATCH] ASoC: Samsung: Modify the I2S driver to support I2S on Exynos5420
by Padmavathi Venna 23 Jul '13

23 Jul '13

Exynos5420 added support for I2S TDM mode. For this, there are some register changes in the I2S controller. This patch adds the relevant register changes to support I2S in normal mode. This patch adds a quirk for TDM mode and if TDM mode is present all the relevent changes will be applied. Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> [abrestic: style cleanup and documentation] Signed-off-by: Andrew Bresticker <abrestic(a)chromium.org> Reviewed-on: https://gerrit-int.chromium.org/37840 Reviewed-by: Simon Glass <sjg(a)google.com> --- .../devicetree/bindings/sound/samsung-i2s.txt | 2 + include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 51 ++++++--- sound/soc/samsung/i2s.c | 117 ++++++++++++++++---- 4 files changed, 132 insertions(+), 39 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index 025e66b..b8593d5 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -28,6 +28,8 @@ Optional SoC Specific Properties: enabled or disabled based on need. - samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA, then this flag is enabled. +- samsung,supports-tdm: If the I2S controller supports TDM, then this flag + must be enabled. - samsung,idma-addr: Internal DMA register base address of the audio sub system(used in secondary sound source). - pinctrl-0: Should specify pin control groups used for this controller. diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 8827259..9efc04d 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -36,6 +36,7 @@ struct samsung_i2s { */ #define QUIRK_NO_MUXPSR (1 << 2) #define QUIRK_NEED_RSTCLR (1 << 3) +#define QUIRK_SUPPORTS_TDM (1 << 4) /* Quirks of the I2S controller */ u32 quirks; dma_addr_t idma_addr; diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h index c0e6d9a..821a502 100644 --- a/sound/soc/samsung/i2s-regs.h +++ b/sound/soc/samsung/i2s-regs.h @@ -31,6 +31,10 @@ #define I2SLVL1ADDR 0x34 #define I2SLVL2ADDR 0x38 #define I2SLVL3ADDR 0x3c +#define I2SSTR1 0x40 +#define I2SVER 0x44 +#define I2SFIC2 0x48 +#define I2STDM 0x4c #define CON_RSTCLR (1 << 31) #define CON_FRXOFSTATUS (1 << 26) @@ -95,24 +99,39 @@ #define MOD_RXONLY (1 << 8) #define MOD_TXRX (2 << 8) #define MOD_MASK (3 << 8) -#define MOD_LR_LLOW (0 << 7) -#define MOD_LR_RLOW (1 << 7) -#define MOD_SDF_IIS (0 << 5) -#define MOD_SDF_MSB (1 << 5) -#define MOD_SDF_LSB (2 << 5) -#define MOD_SDF_MASK (3 << 5) -#define MOD_RCLK_256FS (0 << 3) -#define MOD_RCLK_512FS (1 << 3) -#define MOD_RCLK_384FS (2 << 3) -#define MOD_RCLK_768FS (3 << 3) -#define MOD_RCLK_MASK (3 << 3) -#define MOD_BCLK_32FS (0 << 1) -#define MOD_BCLK_48FS (1 << 1) -#define MOD_BCLK_16FS (2 << 1) -#define MOD_BCLK_24FS (3 << 1) -#define MOD_BCLK_MASK (3 << 1) +#define MOD_LRP_SHIFT 7 +#define MOD_LR_LLOW 0 +#define MOD_LR_RLOW 1 +#define MOD_SDF_SHIFT 5 +#define MOD_SDF_IIS 0 +#define MOD_SDF_MSB 1 +#define MOD_SDF_LSB 2 +#define MOD_SDF_MASK 3 +#define MOD_RCLK_SHIFT 3 +#define MOD_RCLK_256FS 0 +#define MOD_RCLK_512FS 1 +#define MOD_RCLK_384FS 2 +#define MOD_RCLK_768FS 3 +#define MOD_RCLK_MASK 3 +#define MOD_BCLK_SHIFT 1 +#define MOD_BCLK_32FS 0 +#define MOD_BCLK_48FS 1 +#define MOD_BCLK_16FS 2 +#define MOD_BCLK_24FS 3 +#define MOD_BCLK_MASK 3 #define MOD_8BIT (1 << 0) +#define EXYNOS5420_MOD_LRP_SHIFT 15 +#define EXYNOS5420_MOD_SDF_SHIFT 6 +#define EXYNOS5420_MOD_RCLK_SHIFT 4 +#define EXYNOS5420_MOD_BCLK_SHIFT 0 +#define EXYNOS5420_MOD_BCLK_64FS 4 +#define EXYNOS5420_MOD_BCLK_96FS 5 +#define EXYNOS5420_MOD_BCLK_128FS 6 +#define EXYNOS5420_MOD_BCLK_192FS 7 +#define EXYNOS5420_MOD_BCLK_256FS 8 +#define EXYNOS5420_MOD_BCLK_MASK 0xf + #define MOD_CDCLKCON (1 << 12) #define PSR_PSREN (1 << 15) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 3fcf8d7..398f8db 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -198,7 +198,13 @@ static inline bool is_manager(struct i2s_dai *i2s) /* Read RCLK of I2S (in multiples of LRCLK) */ static inline unsigned get_rfs(struct i2s_dai *i2s) { - u32 rfs = (readl(i2s->addr + I2SMOD) >> 3) & 0x3; + u32 rfs; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) + rfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_RCLK_SHIFT; + else + rfs = readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT; + rfs &= MOD_RCLK_MASK; switch (rfs) { case 3: return 768; @@ -212,21 +218,26 @@ static inline unsigned get_rfs(struct i2s_dai *i2s) static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) { u32 mod = readl(i2s->addr + I2SMOD); + int rfs_shift; - mod &= ~MOD_RCLK_MASK; + if (i2s->quirks & QUIRK_SUPPORTS_TDM) + rfs_shift = EXYNOS5420_MOD_RCLK_SHIFT; + else + rfs_shift = MOD_RCLK_SHIFT; + mod &= ~(MOD_RCLK_MASK << rfs_shift); switch (rfs) { case 768: - mod |= MOD_RCLK_768FS; + mod |= (MOD_RCLK_768FS << rfs_shift); break; case 512: - mod |= MOD_RCLK_512FS; + mod |= (MOD_RCLK_512FS << rfs_shift); break; case 384: - mod |= MOD_RCLK_384FS; + mod |= (MOD_RCLK_384FS << rfs_shift); break; default: - mod |= MOD_RCLK_256FS; + mod |= (MOD_RCLK_256FS << rfs_shift); break; } @@ -236,9 +247,22 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) /* Read Bit-Clock of I2S (in multiples of LRCLK) */ static inline unsigned get_bfs(struct i2s_dai *i2s) { - u32 bfs = (readl(i2s->addr + I2SMOD) >> 1) & 0x3; + u32 bfs; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + bfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_BCLK_SHIFT; + bfs &= EXYNOS5420_MOD_BCLK_MASK; + } else { + bfs = readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT; + bfs &= MOD_BCLK_MASK; + } switch (bfs) { + case 8: return 256; + case 7: return 192; + case 6: return 128; + case 5: return 96; + case 4: return 64; case 3: return 24; case 2: return 16; case 1: return 48; @@ -250,21 +274,50 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { u32 mod = readl(i2s->addr + I2SMOD); + int bfs_shift; + int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM; - mod &= ~MOD_BCLK_MASK; + if (tdm) { + bfs_shift = EXYNOS5420_MOD_BCLK_SHIFT; + mod &= ~(EXYNOS5420_MOD_BCLK_MASK << bfs_shift); + } else { + bfs_shift = MOD_BCLK_SHIFT; + mod &= ~(MOD_BCLK_MASK << bfs_shift); + } + + /* Non-TDM I2S controllers do not support BCLK > 48 * FS */ + if (!tdm && bfs > 48) { + dev_err(&i2s->pdev->dev, "Unsupported BCLK divider\n"); + return; + } switch (bfs) { case 48: - mod |= MOD_BCLK_48FS; + mod |= (MOD_BCLK_48FS << bfs_shift); break; case 32: - mod |= MOD_BCLK_32FS; + mod |= (MOD_BCLK_32FS << bfs_shift); break; case 24: - mod |= MOD_BCLK_24FS; + mod |= (MOD_BCLK_24FS << bfs_shift); break; case 16: - mod |= MOD_BCLK_16FS; + mod |= (MOD_BCLK_16FS << bfs_shift); + break; + case 64: + mod |= (EXYNOS5420_MOD_BCLK_64FS << bfs_shift); + break; + case 96: + mod |= (EXYNOS5420_MOD_BCLK_96FS << bfs_shift); + break; + case 128: + mod |= (EXYNOS5420_MOD_BCLK_128FS << bfs_shift); + break; + case 192: + mod |= (EXYNOS5420_MOD_BCLK_192FS << bfs_shift); + break; + case 256: + mod |= (EXYNOS5420_MOD_BCLK_256FS << bfs_shift); break; default: dev_err(&i2s->pdev->dev, "Wrong BCLK Divider!\n"); @@ -492,19 +545,31 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, struct i2s_dai *i2s = to_info(dai); u32 mod = readl(i2s->addr + I2SMOD); u32 tmp = 0; + int lrp_shift, sdf_shift, sdf_mask, lrp_rlow; + + if (i2s->quirks & QUIRK_SUPPORTS_TDM) { + lrp_shift = EXYNOS5420_MOD_LRP_SHIFT; + sdf_shift = EXYNOS5420_MOD_SDF_SHIFT; + } else { + lrp_shift = MOD_LRP_SHIFT; + sdf_shift = MOD_SDF_SHIFT; + } + + sdf_mask = MOD_SDF_MASK << sdf_shift; + lrp_rlow = MOD_LR_RLOW << lrp_shift; /* Format is priority */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - tmp |= MOD_LR_RLOW; - tmp |= MOD_SDF_MSB; + tmp |= lrp_rlow; + tmp |= (MOD_SDF_MSB << sdf_shift); break; case SND_SOC_DAIFMT_LEFT_J: - tmp |= MOD_LR_RLOW; - tmp |= MOD_SDF_LSB; + tmp |= lrp_rlow; + tmp |= (MOD_SDF_LSB << sdf_shift); break; case SND_SOC_DAIFMT_I2S: - tmp |= MOD_SDF_IIS; + tmp |= (MOD_SDF_IIS << sdf_shift); break; default: dev_err(&i2s->pdev->dev, "Format not supported\n"); @@ -519,10 +584,10 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_NB_IF: - if (tmp & MOD_LR_RLOW) - tmp &= ~MOD_LR_RLOW; + if (tmp & lrp_rlow) + tmp &= ~lrp_rlow; else - tmp |= MOD_LR_RLOW; + tmp |= lrp_rlow; break; default: dev_err(&i2s->pdev->dev, "Polarity not supported\n"); @@ -544,15 +609,18 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } + /* + * Don't change the I2S mode if any controller is active on this + * channel. + */ if (any_active(i2s) && - ((mod & (MOD_SDF_MASK | MOD_LR_RLOW - | MOD_SLAVE)) != tmp)) { + ((mod & (sdf_mask | lrp_rlow | MOD_SLAVE)) != tmp)) { dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; } - mod &= ~(MOD_SDF_MASK | MOD_LR_RLOW | MOD_SLAVE); + mod &= ~(sdf_mask | lrp_rlow | MOD_SLAVE); mod |= tmp; writel(mod, i2s->addr + I2SMOD); @@ -1170,6 +1238,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (of_find_property(np, "samsung,supports-rstclr", NULL)) quirks |= QUIRK_NEED_RSTCLR; + if (of_find_property(np, "samsung,supports-tdm", NULL)) + quirks |= QUIRK_SUPPORTS_TDM; + if (of_property_read_u32(np, "samsung,idma-addr", &idma_addr)) { if (quirks & QUIRK_SEC_DAI) { -- 1.7.4.4

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[alsa-devel] [PATCH 1/2] ASoC: Build adau1701 when SND_SOC_ALL_CODECS is selected
by Lars-Peter Clausen 23 Jul '13

23 Jul '13

The adau1701 driver was removed from SND_SOC_ALL_CODECS in commit commit ee8c7e9 ("ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency") due to the dependency on the SigmaDSP firmware loader which was only available on a limited set of platforms. This was fixed quite some time ago in commit 40216ce7 ("ASoC: Move SigmaDSP firmware loader to ASoC") though, so we can add the driver back again to SND_SOC_ALL_CODECS. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 63efc4e..d038a58 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI + select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C -- 1.8.0

2 2